2016-11-23 15:47 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 14.2.0 Released.

2016-11-22 18:52 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 14.2.0-rc2 Released.

2016-11-21 09:49 +0000 [fc5e21ec67]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Use $(LIB_RT) for link of libasteriskpj

	  libasteriskpj was hard coded to use -lrt but librt is linux specific
	  so we now use the LIB_RT variable which gets set by configure.

	  Change-Id: I41148884517e3031f7675a413d524c86e8614694
	  (cherry picked from commit 34a763415f77f4cb17edc728fb51347e979d2fb5)

2016-11-18 19:05 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 14.2.0-rc1 Released.

2016-11-18 09:46 +0000 [7af1aae57f]  Mark Michelson <mmichelson@digium.com>

	* Bump ARI version to 2.0.0

	  In order to not have version number overlap between different versions
	  of Asterisk, each new major version of Asterisk will mean we also bump
	  the ARI major version number.

	  This particular change does NOT introduce any known breaking changes to
	  ARI.

	  For discussion relating to this topice, see:
	  http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html

	  Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665

2016-11-17 10:52 +0000 [7a665c5c6e]  Mark Michelson <mmichelson@digium.com>

	* manager: update minor version

	  Based on bridge video AMI event changes, bump the minor version of AMI.

	  Change-Id: Idf84507354170400813cda780906c94c9f1b60b4

2016-11-16 20:24 +0000 [54d7e65014]  gtjoseph <gjoseph@digium.com>

	* build:  Various OpenBSD issues

	  OpenBSD's 'find' doesn't take the -delete argument so you have to pipe
	  through 'xargs rm -rf'.

	  'echo -e' doesn't like \t starting a line. It just prints 't' which
	  causes the libasteriskpj.exports file to be garbage.  They were just
	  cosmetic so they were removed.

	  librt doesn't exist so the link of libasteriskpj.so fails. It's not
	  actually needed for linux anyway so -lrt was removed from the link.

	  res_rtp_asterisk was failing to load because of an undefined
	  DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if
	  so DTLSv1_method is used instead.

	  ASTERISK-26608

	  Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c

2016-11-14 14:32 +0000 [0825528322]  gtjoseph <gjoseph@digium.com>

	* Revert "Revert "AGI: Only defer frames when in an interception routine.""

	  This reverts commit 7155020c96ecd9b843d9e4073c6c0854e73641e2.

	  Change-Id: I4e1681d2ca8bf36eda849f129bc0dcbc39ef5bc6

2016-11-14 14:31 +0000 [6b2efc116f]  gtjoseph <gjoseph@digium.com>

	* Revert "Revert "autoservice: Use frame deferral API""

	  This reverts commit 8b9996a90c5a81ac104a371ccb8cf89a33c65c75.

	  Change-Id: Ibb2fccfc37188dc54f3776c2d5b41cc425d5cd9c

2016-11-14 14:31 +0000 [f6acb765a8]  gtjoseph <gjoseph@digium.com>

	* Revert "Revert "channel: Use frame deferral API for safe sleep.""

	  This reverts commit 97679ee846893823cdbfd95d91cecc3c14048601.

	  Change-Id: Ib68f76e75882c0230378c4fed97964988e690f3c

2016-11-16 15:42 +0000 [d6783e8fd1]  Mark Michelson <mmichelson@digium.com>

	* res_format_attr_opus: Fix fmtp generation.

	  res_format_attr_opus assumed that the string being passed into it was
	  empty. It tried to determine if the only thing it had written was

	  a=fmtp:<num>

	  And if it had, it would reset the string. Its calculation was off when
	  working with chan_sip, though. chan_sip passes the entire built SDP
	  rather than an empty string. This resulted in always putting an empty
	  fmtp line in the SDP.

	  ASTERISK-26520 #close
	  Reported by scgm11

	  Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5

2016-11-15 16:23 +0000 [d89a1645fc]  Richard Mudgett <rmudgett@digium.com>

	* codec_opus: Fix warning when Opus negotiated but codec_opus not loaded.

	  When Opus is negotiated but not loaded, the log is spammed with messages
	  because the system does not know how to calculate the number of samples in
	  a frame.

	  * Suppress the warning by supplying a function that assumes 20ms of
	  samples in the frame.  For pass through support it doesn't really seem to
	  matter what number of samples is returned anyway.

	  ASTERISK-26605 #close

	  Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f

2016-11-14 14:36 +0000 [fb05acbb32]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.

	  Responding to authentication challenges leaks PJSIP memory pools.

	  The leak was introduced with a pjproject 2.5.5 API change.
	  https://trac.pjsip.org/repos/ticket/1929 changed the API usage of
	  pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to
	  clean up cached authentication allocations that get allocated with
	  pjsip_auth_clt_reinit_req().

	  ASTERISK-26516 #close

	  Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8

2016-11-15 12:01 +0000 [d6f9e2b54a]  gtjoseph <gjoseph@digium.com>

	* file.c/__ast_file_read_dirs:  Fix issues on filesystems without d_type

	  One of the code paths in __ast_file_read_dirs will only get executed if
	  the OS doesn't support dirent->d_type OR if the filesystem the
	  particular file is on doesn't support it.  So, while standard Linux
	  systems support the field, some filesystems like XFS do not.  In this
	  case, we need to call stat() to determine whether the directory entry
	  is a file or directory so we append the filename to the supplied
	  directory path and call stat.  We forgot to truncate path back to just
	  the directory afterwards though so we were passing a complete file name
	  to the callback in the dir_name parameter instead of just the directory
	  name.

	  The logic has been re-written to only create a full_path if we need to
	  call stat() or if we need to descend into another directory.

	  Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba

2016-11-15 08:07 +0000 [f1359c7dc8]  Joshua Colp <jcolp@digium.com>

	* manager: Bump AMI version number.

	  During the development of Asterisk 14 the behavior of
	  the Command AMI action was altered such that the result
	  was returned on lines with a prefix of "Output: ". While
	  this was documented in the UPGRADE.txt file it is also
	  reasonable that this should bump the AMI version number.

	  ASTERISK-26556

	  Change-Id: Idf1bf01608e53f7bfdf43ddb4d0683e53f74ee42

2016-11-14 15:57 +0000 [fb9b867d7d]  Matt Jordan <mjordan@digium.com>

	* pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS

	  The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how
	  many pairs of local/remote candidates will be made. If for some reason
	  we reach this upper bound, ICE will generally fail and no media will
	  flow between the browser and Asterisk.

	  This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of
	  pairs of candidates we'd theoretically allow, which is
	  PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied
	  PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame
	  Docker), this is far too low to allow WebRTC calls to succeed.

	  Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed
	  even when the system Asterisk was running on had quite a few virtual
	  interfaces.

	  Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55

2016-11-14 15:32 +0000 [46bedcbbad]  Matt Jordan <mjordan@digium.com>

	* apps/app_echo: Only relay a single video source change frame

	  In 9785e8d0, app_echo was updated to relay video source updates to the
	  channel for the purposes of displaying video in WebRTC tests.
	  Unfortunately, this can cause a Kafkaesque nightmare if two or more
	  Local channels are in a bridge together where their ends are in
	  app_echo. When this situation occurs, a video update sent into app_echo
	  will cause the video update to be relayed to the other Local channels,
	  causing another round of video updates, etc. In not much time at all,
	  the channel length queues will be overwhelmed, channel alert pipes will
	  fail, and all hell will break loose as Asterisk merrily continues to
	  throw more video update requests onto the channels.

	  This patch updates app_echo to *only* relay a single video update. Once
	  a video update has been made, all further video updates are dropped.
	  This meets the intended purpose of the original patch: if we get a video
	  update and we're in app_echo, go ahead and ask the sender to update
	  themselves. However, once we've got that video stream sync'd up, don't
	  keep spamming the world.

	  Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74

2016-11-08 10:11 +0000 [62cbcb2e54]  Matt Jordan <mjordan@digium.com>

	* res/ari/resource_bridges: Add the ability to manipulate the video source

	  In multi-party bridges, Asterisk currently supports two video modes:
	   * Follow the talker, in which the speaker with the most energy is shown
	     to all participants but the speaker, and the speaker sees the
	     previous video source
	   * Explicitly set video sources, in which all participants see a locked
	     video source

	  Prior to this patch, ARI had no ability to manipulate the video source.
	  This isn't important for two-party bridges, in which Asterisk merely
	  relays the video between the participants. However, in a multi-party
	  bridge, it can be advantageous to allow an external application to
	  manipulate the video source.

	  This patch provides two new routes to accomplish this:
	  (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
	      Sets a video source to an explicit channel
	  (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
	      Removes any explicit video source, and sets the video mode to talk
	      detection

	  ASTERISK-26595 #close

	  Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621

2016-11-14 14:03 +0000 [ac6051c302]  gtjoseph <gjoseph@digium.com>

	* channel:  Fix issues in hangup scenarios caused by frame deferral

	  ASTERISK-26343

	  Change-Id: I06dbf7366e26028251964143454a77d017bb61c8
	  (cherry picked from commit 0be46aaf6b8b9eb5b0160ec591cdc2c6e1802a6d)
	  (cherry picked from commit 7263a17ca046923c245296a3cee21df0e4bd499a)
	  (cherry picked from commit ebf8d94eb2f0d14b824ee6243789870325583624)

2016-11-14 14:31 +0000 [88111da235]  gtjoseph <gjoseph@digium.com>

	* Revert "Revert "Add API for channel frame deferral.""

	  This reverts commit b640f18a440e0b7438d836f04ca7146a4922e379.

	  Change-Id: I4ad8e7f66b3e6457ac07e7965822ae39ac08464d

2016-11-11 10:45 +0000 [eb5077fb26]  Mark Michelson

	* res_ari: Add support for channel variables in ARI events.

	  This works the same as for AMI manager variables. Set
	  "channelvars=foo,bar" in your ari.conf general section, and then the
	  channel variables "foo" and "bar" (along with their values), will
	  appear in every Stasis websocket channel event.

	  ASTERISK-26492 #close
	  patches:
	    ari_vars.diff submitted by Mark Michelson

	  Change-Id: I5609ba239259577c0948645df776d7f3bc864229

2016-11-14 12:16 +0000 [e2046bfaf3]  gtjoseph <gjoseph@digium.com>

	* cli:  Fix ast_el_read_char to work with libedit >= 3.1

	  Libedit 3.1 is not build with unicode on as a default and so the
	  prototype for the el_gets callback changed from expecting a char buffer
	  to accepting a wchar buffer.  If ast_el_read_char isn't changed,
	  the cli reads garbage from teh terminal.

	  Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and
	  updated ast_el_read_char to use the HAVE_ define to detemrine whether
	  to use char or wchar.

	  ASTERISK-26592 #close

	  Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a

2016-11-11 02:41 +0000 [ba09e7a23f]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* Fix closing rtp ports after call finished in chan_unistim.

	  Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
	  rtp instance destroy for chan_unistim. Also several fixes
	  for displayed text translation.

	  Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc

2016-11-10 10:57 +0000 [cc4a2c8c76]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp.

	  When optimistic SRTP was on it was possible for us to still
	  set up a call without an audio stream if an offer was received
	  with required SRTP.

	  This change makes it so this scenario will now fail with a 488
	  response.

	  ASTERISK-26575

	  Change-Id: I7d14187037681f48879bd20319ac79d0877318f3

2016-09-23 17:54 +0000 [d36695e0bb]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip.c: Rework endpt_send_request() req_wrapper code.

	  * Don't hold the req_wrapper lock too long in endpt_send_request().  We
	  could block the PJSIP monitor thread if the timeout timer expires.
	  sip_get_tpselector_from_endpoint() does a sorcery access that could take
	  awhile accessing a database.  pjsip_endpt_send_request() might take awhile
	  if selecting a transport.

	  * Shorten the time that the req_wrapper lock is held in the callback
	  functions.

	  * Simplify endpt_send_request() req_wrapper->timeout code.

	  * Removed some redundant req_wrapper->timeout_timer->id assignments.

	  Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9

2016-09-21 15:10 +0000 [add253cbd0]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Fix tdata leaks in off nominal paths.

	  Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b

2016-10-24 12:41 +0000 [49dd637047]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_registrar_expire.c: Remove extra linefeed in debug message.

	  Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94

2016-11-09 15:14 +0000 [119e1fd6cf]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Perform resolution when explicit IPv6 transport is used.

	  This change fixes the SIP resolver such that if an IPv6 transport
	  is explicitly used it will resolve NAPTR, SRV, and AAAA records.

	  You can explicitly use one by specifying it on an endpoint.

	  ASTERISK-26571

	  Change-Id: I2ed3ce81b43a6a8a937c0ebc1b8ed2da5ac2ef36

2016-11-10 08:33 +0000 [90389f6b7d]  Joshua Colp <jcolp@digium.com>

	* app_queue: Add mention of 'ABANDON' variable to CHANGES.

	  ASTERISK-26558

	  Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e

2016-11-10 07:38 +0000 [b640f18a44]  gtjoseph <gjoseph@digium.com>

	* Revert "Add API for channel frame deferral."

	  This reverts commit 4a8b1940b896f7e783a5429d0fffee8bed0e4afd.
	  Multiple testsuite failures were detected after the fact.

	  Change-Id: Ife9f86872d323adb9c068d2942339c7be9499257

2016-11-10 07:37 +0000 [7155020c96]  gtjoseph <gjoseph@digium.com>

	* Revert "AGI: Only defer frames when in an interception routine."

	  This reverts commit 8d8323b1427b1abe60b4b8777dccb3a2232315e8.
	  Multiple testsuite failures were detected after the fact.

	  Change-Id: I322684ed82ba32cfce226a94fe19b87a4ec02567

2016-11-10 07:37 +0000 [8b9996a90c]  gtjoseph <gjoseph@digium.com>

	* Revert "autoservice: Use frame deferral API"

	  This reverts commit 0288fba2f0c0d208b3e6d563f5f77badf15cd3a7.
	  Multiple testsuite failures were detected after the fact.

	  Change-Id: I9951207019a0f528787f735ccfed7c4b76335505

2016-11-10 07:37 +0000 [97679ee846]  gtjoseph <gjoseph@digium.com>

	* Revert "channel: Use frame deferral API for safe sleep."

	  This reverts commit 1db9e7886cc72355663123e906cf5f2dc4a2ef84.
	  Multiple testsuite failures were detected after the fact.

	  Change-Id: Ifd5817e90371e05c4011d5cc97da145c196d60fe

2016-11-09 18:18 +0000 [dbcb958254]  gtjoseph <gjoseph@digium.com>

	* build:  Fix default values for some SANITIZER options

	  2 of the sanitizers didn't have default values so in systems that
	  don't support sanitizers menuselect would spit out warnings.  They
	  were harmless but confusing.  They've now been set to "0".

	  Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58

2016-11-06 06:04 +0000 [abd41590d7]  Sebastian Gutierrez <sgutierrez@integraccs.com>

	* app_queue: new variable set when abandoned

	  sets the variable ABANDONED to TRUE if the call was not answered.

	  ASTERISK-26558

	  Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3

2016-11-08 10:48 +0000 [65a0b6bca2]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_session: Do not call session supplements when it's too late.

	  res_pjsip_sesssion was hooking into transaction and invite state
	  changes. One of the reasons for doing so was due to the
	  PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the
	  message sending process, and so we should call session supplements to
	  alter the outgoing message.

	  In reality, this event was meant to indicate that the message either
	  a) had already been sent, or
	  b) required a DNS lookup and would be sent when the DNS query
	  completed.

	  In case (a), this meant we were altering an already-sent
	  request/response for no reason. In case (b), this potentially meant we
	  could be trying to alter a request/response at the same time that the
	  DNS resolution completed. In this case, it meant we might be stomping on
	  memory being used by the thread actually sending the message. This
	  caused potential crashes and memory corruption.

	  This patch removes the calls to session supplements from the case where
	  the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to
	  alter the message at this point is too late, and it can cause nothing
	  but harm to try to do it. Because there were no longer any calls to the
	  handle_outgoing() function, it has been removed.

	  Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92

2016-11-03 16:46 +0000 [1db9e7886c]  Mark Michelson <mmichelson@digium.com>

	* channel: Use frame deferral API for safe sleep.

	  This is another case where manual frame deferral can be replaced with
	  centralized routines instead.

	  Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e
	  (cherry picked from commit d149c4b9e07eeb880d8428ad52c6fdb315cc15f5)

2016-11-03 16:46 +0000 [0288fba2f0]  Mark Michelson <mmichelson@digium.com>

	* autoservice: Use frame deferral API

	  Rather than use manual frame deferral, just let the channel API do it
	  for us.

	  ASTERISK-26343

	  Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49
	  (cherry picked from commit 8ba3e2fc27f9966b8c7ce75c1eca6208613a9315)

2016-11-03 16:42 +0000 [8d8323b142]  Mark Michelson <mmichelson@digium.com>

	* AGI: Only defer frames when in an interception routine.

	  AGI recently was modified to defer important frames. This was because
	  when AGI was used in a connected line interception routine, the
	  resulting connected line frame would end up getting discarded by the
	  AGI.

	  However, this caused bad behavior in other cases. Specifically, during a
	  transfer, if someone attempted to manually set the Caller ID on a
	  channel in an AGI, the deferred connected line frame would end up
	  overwriting what had been manually set in the AGI.

	  Since the initial issue was specific to interception routines, this
	  change removes the manual frame deferral from AGI and instead uses the
	  new frame deferral API in interception routines.

	  ASTERISK-26343 #close
	  Reported by Morton Tryfoss

	  Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208

2016-11-03 16:36 +0000 [4a8b1940b8]  Mark Michelson <mmichelson@digium.com>

	* Add API for channel frame deferral.

	  There are several places in Asterisk that have duplicated logic
	  for deferring important frames until later.

	  This commit adds a couple of API calls to facilitate this automatically.

	  ast_channel_start_defer_frames(): Future reads of deferrable frames on
	  this channel will be deferred until later.

	  ast_channel_stop_defer_frames(): Any frames that have been deferred get
	  requeued onto the channel.

	  ASTERISK-26343

	  Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641

2016-11-02 10:52 +0000 [cfede6a1fc]  Joshua Colp <jcolp@digium.com>

	* res_stasis: Don't unsubscribe from a NULL bridge.

	  A NULL bridge has special meaning in res_stasis for
	  unsubscribing. It means that a subscription to ALL
	  bridges should be removed. This should not be done
	  as part of the normal subscription management in
	  the res_stasis channel loop.

	  ASTERISK-26468

	  Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0

2016-11-03 07:42 +0000 [2849e596b8]  Alexander Anikin <may213@yandex.ru>

	* chan_ooh323: Fixes to work right with Cisco devices

	  Changed output packets queue processing algo to one read-one write
	  instead of all read-all send

	  Remove h.245 tunneling parameter from ReleaseComplete packet

	  ASTERISK-24400 #close
	  Reported by: Dmitry Melekhov
	  Tested by: Dmitry Melekhov

	  Change-Id: I0b31933b062a21011dbac9a82b8bcfe345f406f6

2016-11-03 13:10 +0000 [e57f994237]  Alexander Anikin <may213@yandex.ru>

	* chan_ooh323: reset rrq count on gk registration

	  reset registration attempts count on success registration on gatekeeper

	  Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336

2016-11-06 03:46 +0000 [0f627421b9]  Michael Kuron <m.kuron@gmx.de>

	* automon: restore mixing of the both channels after recording stops

	  This is a regression over Asterisk 11, introduced by
	  2dc8a060064f359a17f5ebcd515d85fe5203c019. Previously, recordings started via
	  the automon DTMF code would automatically be mixed together using sox because
	  app_monitor would be called with the m option. This commit restores this
	  behavior.

	  Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759

2016-11-04 15:42 +0000 [59d25b2081]  Matt Jordan <mjordan@digium.com>

	* res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems

	  Not surprisingly, using Respoke (and possibly other systems) it is
	  possible to blow past the 16k limit for a WebSocket packet size. This
	  patch bumps it up to 32k, which, at least for Respoke, is sufficient.
	  For now.

	  Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that
	  matter), this patch adds a LOW_MEMORY directive that sets the buffer to
	  8k for systems who have asked for their reduced memory availability to
	  be considered.

	  Change-Id: Id235902537091b58608196844dc4b045e383cd2e

2016-11-04 15:40 +0000 [d889fb375b]  Matt Jordan <mjordan@digium.com>

	* res_stasis: Set a video source mode on Stasis created bridges

	  When a bridge is created via ARI (through res_stasis), no video source
	  mode is set by default. As a result, any endpoint sending video media
	  won't ever see any video reflected back to it.

	  This patch defaults a bridge to a 'follow the talker' video mode.
	  Further work can be done to add routes that allow for the video mode to
	  be controlled through the /bridges resource.

	  Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866

2016-11-04 15:37 +0000 [62d60c1f58]  Matt Jordan <mjordan@digium.com>

	* main/bridge_channel: Fix channel reference leak on video source

	  When a channel is made the video source, the bridge holds a reference to
	  it. Whenever the video source changes, that reference is released.
	  However, a ref leak does occur if the channel leaves the bridge (such as
	  being hung up) while it is the video source, as the bridge never
	  releases the ref in such a case.

	  This patch adds a line to the bridge_channel_internal_join routine such
	  that, when a channel finishes its time in the bridge, it notifies the
	  bridge via ast_bridge_remove_video_src that if it is a video source its
	  reference should be released.

	  ASTERISK-26555 #close

	  Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a

2016-11-04 15:36 +0000 [be254aa8e3]  Matt Jordan <mjordan@digium.com>

	* main/bridge: Add some verbose logging for video source changes

	  It's actually quite useful to see the source of a video stream change.
	  This doesn't happen terribly often, even with talk detection - but when
	  it does, it's nice to know which channel is now providing your video
	  stream.

	  As a verbose 5 level message, it shouldn't be terribly spammy or costly
	  to have, and is 'lower level' then most other verbose messages that the
	  bridge system emits.

	  ASTERISK-26555

	  Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c

2016-11-04 15:33 +0000 [ff96e37509]  Matt Jordan <mjordan@digium.com>

	* bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source

	  WebRTC clients really, really want to know the SSRC of the media they're
	  getting. Changing the SSRC is generally not a good thing.

	  bridge_softmix, starting in Asterisk 12, started changing the SSRC of
	  parties as they joined or left the bridge. With most phones, this isn't
	  a problem: phones just play back the stream they're getting. With WebRTC
	  clients, however, the SSRC is tied to a media stream that may be
	  negotiated. When a new SSRC just shows up, the media can be dropped.

	  As it turns out, the SSRC change shouldn't even be necessary. From the
	  perspective of the client, it's still talking to Asterisk with the same
	  media stream: why indicate that the far party has suddenly changed to a
	  different source of media?

	  This patch opts to just remove the SSRC changes. With this patch, video
	  clients that join/leave a softmix bridge actually get the video stream
	  instead of freaking out.

	  ASTERISK-26555

	  Change-Id: I27fec098b32e7c8718b4b65f3fd5fa73527968bf

2016-10-28 15:11 +0000 [25895897f8]  Kevin Harwell <kharwell@digium.com>

	* stasis_recording/stored: remove calls to deprecated readdir_r function.

	  The readdir_r function has been deprecated and should no longer be used. This
	  patch removes the readdir_r dependency (replaced it with readdir) and also moves
	  the directory search code to a more centralized spot (file.c)

	  Also removed a strict dependency on the dirent structure's d_type field as it
	  is not portable. The code now checks to see if the value is available. If so,
	  it tries to use it, but defaults back to using the stats function if necessary.

	  Lastly, for most implementations of readdir it *should* be thread-safe to make
	  concurrent calls to it as long as different directory streams are specified.
	  glibc falls into this category. However, since it is possible that there exist
	  some implementations that are not safe, locking has been added for those other
	  than glibc.

	  ASTERISK-26412
	  ASTERISK-26509 #close

	  Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba

2016-11-04 10:57 +0000 [ac82b40bff]  Kevin Harwell <kharwell@digium.com>

	* Revert "chan_sip: Fix lastrtprx always updated"

	  This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc.

	  Unfortunately, the aforementioned commit caused a regression (incoming calls
	  would eventually disconnect). Thus it is being removed.

	  ASTERISK-26523 #close
	  ASTERISK-25270

	  Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d

2016-11-03 13:45 +0000 [e6b414a66a]  Alexander Anikin <may213@yandex.ru>

	* chan_ooh323: Fix infinite loop on read second part of H.225 packet

	  Fix logic on read second part of H.225 packet. There was infinite loop on
	  wrong connections due to read before poll.

	  Change-Id: I42b4bf75c46e4a5c5df5c5ca1f0bd74b8944e7ff

2016-11-03 11:55 +0000 [aa04c453b8]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Fix issue with libasteriskpj needing libresample

	  libresample is only needed by pjproject if we're building pjsua, which
	  we only do if TEST_FRAMEWORK is selected.  It's required by pjsua to
	  process audio which is needed by some testsuite tests.  Unfortunately,
	  pjproject relies on a newer version of libresample than the version
	  that ships by most distros so we need to compile the version that's
	  bundled with pjproject.  Since we only need it for pjsua, we DON'T want
	  it's symbols exposed when we actually build asterisk.

	  There was a problem however... TEST_FRAMEWORK is only known AFTER we've
	  already run ./configure on both asterisk and pjproject but pjproject's
	  ./configure needs to test it to know whether to set up to build
	  libresample or not.  The previous way of figuring this out was to
	  always tell ./configure "yes" but not actually build the library.  This
	  caused an issue where building libasteriskpj was being told to include
	  libresample but it wasn't actually there.

	  The solution is to still do a default pjproject configure during an
	  asterisk ./configure but if makeopts or menuselect.makeopts changes
	  subsequently, we now reconfigure pjproject, taking into account the
	  current state of TEST_FRAMEWORK.  Previously, if makeopts or
	  menuselect.makeopts changed, only a recompile of pjproject was done.

	  Change-Id: I9b5d84c61384a3ae07fe30e85c49698378cc4685

2016-11-01 19:48 +0000 [1012c28437]  Sebastian Gutierrez <sgutierrez@integraccs.com>

	* chan_sip: add missing account code

	  Added missing account to AMI event of sip show peers

	  ASTERISK-26176 #close

	  Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482

2016-11-02 09:15 +0000 [b2a078efc9]  Joshua Colp <jcolp@digium.com>

	* app_dial: Fix incorrect device state when channel is picked up.

	  Given the scenario where multiple channels are dialed using Dial()
	  but the caller is picked up using PickupChan() all outgoing channels
	  except the channel specified to PickupChan() would be marked
	  as ringing until the call had been hung up.

	  When using the PickupChan application the channel executing the
	  application is swapped into place of another channel. As part
	  of this process the channel is answered. The Dial application
	  has explicit logic which checks if the channel is answered,
	  cancels all other outgoing channels, and bridges. This logic is
	  different than the normal logic that is executed when an outgoing
	  channel is answered. This different logic failed to publish dial
	  events stating that the other outgoing channels had been canceled.
	  As a result references to the outgoing channels were held onto by
	  the dial masquerade process until the call had been ended and
	  the channels had gone away. This would result in the channels
	  appearing in the "core show channels" list despite not being present
	  anymore and would also result in incorrect device state.

	  This change makes it so that this logic also publishes
	  dial events stating that the other outgoing channels have been
	  canceled.

	  ASTERISK-26549

	  Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f

2016-11-02 08:55 +0000 [6445f21caa]  Alexander Traud <pabstraud@compuserve.com>

	* rtp_engine: Allow more than 32 dynamic payload types.

	  Since adding all remaining rates of Signed Linear (ASTERISK-24274) and SILK
	  (Gerrit 3136), only one RTP Payload Type is left in the dynamic range (96-127).
	  RFC 3551 section 3 allows to reassign other ranges. Consequently, when the
	  dynamic range is exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in
	  asterisk.conf. This enables the range 35-63 (or 0-63) giving room for another
	  29 (or 64) payload types.

	  ASTERISK-26311 #close

	  Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964

2016-11-01 13:13 +0000 [4fda9e7b0b]  Richard Mudgett <rmudgett@digium.com>

	* bundled pjproject: Fix DNS write to freed memory.

	  PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS
	  patch.

	  The patch below fixes a write to freed memory under cartain DNS lookup
	  conditions.

	  0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch

	  ASTERISK-26516
	  Reported by:  Richard Mudgett

	  Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5

2016-10-28 13:30 +0000 [d3e0d5d40b]  Matt Krokosz <mkrokosz@vonage.com>

	* res_pjsip_outbound_publish: Fix crash when publishing device state.

	  While publishing device state between multiple instances of Asterisk,
	  a crash will sporadically occur under high CPS which looks to be a
	  race condition operating on the publisher queue.

	  ASTERISK-26506

	  Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6

2016-11-01 04:18 +0000 [2897fc9ab0]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* netsock.c: fix includes for HURD

	  ASTERISK-25070

	  Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814

2016-11-01 04:00 +0000 [3748e336ac]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* define PATH_MAX for HURD

	  PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD
	  define it to a constant. It is indeed not safe to assume there won't be
	  longer paths and Asterisk generally does err safely on such cases.

	  So even for HURD we'll just pretend PATH_MAX is 4096.

	  ASTERISK-25070 #close

	  Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3

2016-10-31 17:35 +0000 [35f9d472ba]  Kevin Harwell <kharwell@digium.com>

	* codecs.conf.sample: Add sample and option descriptions for codec_opus

	  codecs.conf.sample was missing codec opus's configuration options, descriptions,
	  and examples. This patch adds the configuration options and examples to
	  codecs.conf.sample that can be used with codec_opus.

	  ASTERISK-26538 #close

	  Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b

2016-10-20 07:27 +0000 [1a3e699316]  Matt Jordan <mjordan@digium.com>

	* res/stasis: Add CLI commands for displaying/debugging ARI apps

	  This patch adds three new CLI commands:
	   - ari show apps: list the registered ARI applications
	   - ari show app: show detailed information about an ARI application
	   - ari set debug: dump events being sent to an ARI application

	  Note that while these CLI commands live in the res_stasis module, we use
	  the 'ari' family for these commands. This was done as most users of
	  Asterisk aren't aware of the semantic differences between ARI and
	  res_stasis, and some 'ari' CLI commands already exist.

	  ASTERISK-26488 #close

	  Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5

2016-11-01 08:32 +0000 [782dfa09a8]  Grachev Sergey <grachev@mcn.ru>

	* chan_sip: Incorrect display option Outbound reg. retry 403

	  If in sip.conf (general section) set option register_retry_403=no,
	  the command "sip show settings" return value:
	  Outbound reg. retry 403:0
	  If in sip.conf (general section) set option register_retry_403=yes,
	  the command "sip show settings" return value:
	  Outbound reg. retry 403:-1

	  * In static char "sip show settings" for "Outbound.reg. retry 403"
	  option use AST_CLI_YESNO

	  ASTERISK-26476 #close

	  Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9

2016-11-01 06:56 +0000 [69196a8db4]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Limit number of formats to defined maximum.

	  The res_pjsip_sdp_rtp module did not restrict the number of
	  formats added to a media stream in the SDP to the defined
	  limit. If allow=all was used with additional loaded codecs this
	  could result in the next media stream being overwritten some.

	  This change restricts the module to limit it to the defined
	  maximum and also increases the maximum in our bundled pjproject.

	  ASTERISK-26541 #close

	  Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8

2016-10-31 16:12 +0000 [23812d60b9]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Fix compile of pjsua so it handles audio

	  In order for pjsua and its python binding to actually negotiate
	  audio for the testsuite tests, it needs g711 and resample.  The
	  pj* libraries themselves do not.  Unfortunately, pjproject relies
	  on a brand new libresample that most distros don't ship so we need
	  to use the libresample already bundled with pjproject.  Only the pjsua
	  executable and the _pjsua.so python library are linked with it so it
	  shouldn't interfere with asterisk itself.

	  Also it was pointed out that apply_patches couldn't handle multiple
	  patches that depended on each other during the dry-run, so the
	  dry-run was removed.

	  Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098

2016-10-31 13:46 +0000 [d9f9691d31]  Etienne Lessard <elessard@proformatique.com>

	* manager: Add documentation for NewConnectedLine event.

	  The NewConnectedLine event has been added by commit fe7671f, but the
	  documentation was missing.

	  ASTERISK-26537 #close

	  Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6

2016-10-30 13:33 +0000 [b2bf6c4c22]  Corey Farrell <git@cfware.com>

	* vector: Prevent NULL argument to memcpy.

	  Headers declare that memcpy does not accept NULL argument for the first
	  two parameters.  Add a conditional block to prevent memcpy and ast_free
	  from running on vectors with NULL element array.

	  ASTERISK-26526 #close

	  Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71

2016-10-29 10:19 +0000 [39ba7aa91f]  Corey Farrell <git@cfware.com>

	* astobj2: Declare private variable data_size for AO2_DEBUG only.

	  Every ao2 object contains storage for a private variable data_size,
	  though the value is never read if AO2_DEBUG is disabled.  This change
	  makes the variable conditional, reducing memory usage.

	  ASTERISK-26524 #close

	  Change-Id: If859929e507676ebc58b0f84247a4231e11da07f

2016-10-28 14:55 +0000 [79ac79ab03]  Richard Mudgett <rmudgett@digium.com>

	* bundled pjproject: Crashes while resolving DNS names.

	  PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS
	  patch.

	  The patches below fix the DNS lookup race condition crash caused by
	  attempting to send the same message twice for the single DNS lookup.

	  0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch
	  0006-r5473-svn-backport-Fix-pending-query.patch

	  The patch below removes a cached DNS response from the hash table when
	  another thread is referencing the old entry.  The table still contained
	  the entry when it was destroyed which can result in inexplicable crashes.

	  0006-r5475-svn-backport-Remove-DNS-cache-entry.patch

	  ASTERISK-26344 #close
	  Reported by: Ian Gilmour

	  ASTERISK-26387 #close
	  Reported by: Harley Peters

	  Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4

2016-10-28 16:59 +0000 [d84eaa46fa]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Fix issue where "/version.mak" wasn't found

	  main/Makefile includes third-party/pjproject/build.mak but
	  doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak"
	  evaluates to "/version.mak".  Fix is to set PJDIR in main/Makefile
	  before the include.

	  Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604

2016-10-27 21:49 +0000 [b76afa5e4f]  Corey Farrell <git@cfware.com>

	* Fix shutdown crash caused by modules being left open.

	  It is only safe to run ast_register_cleanup callbacks when all modules
	  have been unloaded.  Previously these callbacks were run during graceful
	  shutdown, making it possible to crash during shutdown.

	  ASTERISK-26513 #close

	  Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21

2016-10-28 09:50 +0000 [c2a2643c69]  Rusty Newton <rnewton@digium.com>

	* SAC documentation: don't specify transports for endpoints and registrations

	  Removing explicit transport definition for endpoints and registrations. It
	  isn't necessary and isn't generally advised.

	  ASTERISK-26514 #close

	  Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb

2016-10-26 18:48 +0000 [162bb27cfb]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Remove usage of tar's --strip-components option

	  Older versions of tar don't support the --strip-components option so
	  instead of doing 'tar --strip-components=1 -C source', we now just
	  untar to the tarball's root directory (pjproject-<version>) and
	  rename that directory to 'source'.

	  Also fixed an issue where the pjproject source directory is a hard
	  coded absolute pathname.

	  ASTERISK-26510 #close
	  ASTERISK-22480 #close

	  Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0

2016-10-27 08:07 +0000 [82ef2bb69d]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls.

	  The res_pjsip_caller_id module wrongly assumed that a
	  saved From header would always exist on sessions. This
	  is true until an inbound call is received and a session
	  timer causes an UPDATE to be sent. In this case there will
	  be no saved From header and a crash will occur. This change
	  makes it fall back to the From header of the outgoing request
	  if no saved From header is present.

	  ASTERISK-26307 #close

	  Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa

2016-10-26 07:51 +0000 [ebc293e609]  Joshua Colp <jcolp@digium.com>

	* app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS.

	  When executing the MailboxExists dialplan application and
	  MAILBOX_EXISTS dialplan function the passed in temporary voice
	  mailbox was not cleared, causing it to try to free garbage.

	  ASTERISK-26503 #close

	  Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3

2016-10-23 07:38 +0000 [791d2319ce]  Joshua Colp <jcolp@digium.com>

	* pjsip: Fix a few media bugs with reinvites and asymmetric payloads.

	  When channel format changes occurred as a result of an RTP
	  re-negotiation the bridge was not informed this had happened.
	  As a result the bridge technology was not re-evaluated and the
	  channel may have been in a bridge technology that was incompatible
	  with its formats. The bridge is now unbridged and the technology
	  re-evaluated when this occurs.

	  The chan_pjsip module also allowed asymmetric codecs for sending
	  and receiving. This did not work with all devices and caused one
	  way audio problems. The default has been changed to NOT do this
	  but to match the sending codec to the receiving codec. For users
	  who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
	  which will return chan_pjsip to the previous behavior.

	  The codecs returned by the chan_pjsip module when queried by
	  the bridge_native_rtp module were also not reflective of the
	  actual negotiated codecs. The nativeformats are now returned as
	  they reflect the actual negotiated codecs.

	  ASTERISK-26423 #close

	  Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc

2016-10-26 06:32 +0000 [110c18f413]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Fix address family of explicit media_address.

	  When an explicit media_address is provided the address family
	  in the SDP needs to be set to reflect it.

	  ASTERISK-26309

	  Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79

2016-10-25 11:20 +0000 [f2d406ced8]  gtjoseph <gjoseph@digium.com>

	* test_astobj2_thrash:  Fix multithreaded issues

	  The test uses 4 threads to grow, count, lookup and shrink 15K objects
	  in a container.  If there's only 1 execution engine available, the test
	  will complete in <50ms.  If each threads gets its own execution engine,
	  the test may timeout after 60 seconds because the count thread does a
	  locked ao2_callback on the whole container in a tight loop with only
	  a sched_yield to give up time.  The lock contention makes the test
	  execution times wildly variable and mostly timeout.  2 execution
	  engines are OK, 3 results in about 33% failure rate and >=4 causes
	  a 80% failure rate.

	  To fix, the sched_yield was changed to a usleep(500).

	  Also, the number of buckets specified for the container was an even
	  number so that was changed to the next prime number greater than
	  (MAX_HASH_ENTRIES / 100).  That's 151 currently.

	  Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77

2016-10-18 09:04 +0000 [3e040685c7]  Alexei Gradinari <alex2grad@gmail.com>

	* chan_pjsip: segfault on already disconnected session

	  On heavy loaded system the TCP/TLS incoming calls could be
	  disconnected by pjproject while these calls are being
	  processed by asterisk.

	  This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref
	  to inform pjproject that an INVITE session is in use.

	  ASTERISK-26482 #close

	  Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33

2016-10-24 14:13 +0000 [cb342298cd]  Pascal Cadotte Michaud <pcadotte@proformatique.com>

	* typo: s/paranthesis/parenthesis/ in a comment

	  Change-Id: I7c1f4eb051177ee22cbe97e063d4a3effe29be30

2016-10-24 10:55 +0000 [0836b617e5]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Fixed various build issues

	  * CFLAGS is now properly set when using older gcc.
	  * All third-party pjproject targets have been removed.  This fixes
	    an issue with older libsrtp in some distros.
	  * Manually removing the source directory now causes a rebuild.
	  * EXTERNALS_CACHE_DIR is now properly checked.
	  * Whitespace fixes.

	  Change-Id: I98fec6847efc5602a9f41cb95096fd660a49fa60

2016-09-19 06:13 +0000 [47de0ee4f6]  Joshua Colp <jcolp@digium.com>

	* pjsip: Support dual stack automatically.

	  This change adds support for dual stack automatically. No
	  configuration is required and the IP address and version
	  in the SIP messages and SDP will be automatically changed
	  based on the transport over which the message is being
	  sent. RTP usage has also been changed to listen on both
	  IPv4 and IPv6 simultaneously to allow media to flow, and
	  to allow ICE support on both simultaneously. This also
	  allows failover between IPv6 and IPv4 to work as expected.

	  ASTERISK-26309 #close

	  Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d

2016-10-19 12:05 +0000 [ba145944a0]  Mark Michelson <mmichelson@digium.com>

	* ARI: Add duplicate channel ID checking for channel creation.

	  This is similar to what is done for origination, but for the 14 and up
	  channel creation method. When attempting to create a channel, if a
	  channel ID is specified and a channel already exists with that ID, then
	  a 409 is returned.

	  Change-Id: I77f9253278c6947939c418073b6b31065489187c

2016-10-17 14:18 +0000 [98378133c0]  Mark Michelson <mmichelson@digium.com>

	* ARI: Detect duplicate channel IDs

	  ARI and AMI allow for an explicit channel ID to be specified
	  when originating channels. Unfortunately, there is nothing in
	  place to prevent someone from using the same ID for multiple
	  channels. Further complicating things, adding ID validation to channel
	  allocation makes it impossible for ARI to discern why channel allocation
	  failed, resulting in a vague error code being returned.

	  The fix for this is to institute a new method for channel errors to be
	  discerned. The method mirrors errno, in that when an error occurs, the
	  caller can consult the channel errno value to determine what the error
	  was. This initial iteration of the feature only introduces "unknown" and
	  "channel ID exists" errors. However, it's possible to add more errors as
	  needed.

	  ARI uses this feature to determine why channel allocation failed and can
	  return a 409 error during origination to show that a channel with the
	  given ID already exists.

	  ASTERISK-26421

	  Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06

2016-10-19 17:53 +0000 [8e3aee2036]  snuffy <snuffy22@gmail.com>

	* Fix issue with CLI not returning to prompt after running "features show"

	  ASTERISK-26444 #close

	  Change-Id: I91d645b7e6e5dba35f8c410df2be77a8c0e3acb8

2016-10-04 18:24 +0000 [494bebeb6f]  Michael Walton <mike@farsouthnet.com>

	* res_rtp_asterisk: Add ice_blacklist option

	  Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the
	  form ice_blacklist = <subnet spec>, e.g. ice_blacklist =
	  192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay
	  discovery. This is useful for optimizing the ICE process where a system
	  has multiple host address ranges and/or physical interfaces and certain
	  of them are not expected to be used for RTP. Multiple ice_blacklist
	  configuration lines may be used. If left unconfigured, all discovered
	  host addresses are used, as per previous behavior.

	  Documention in rtp.conf.sample.

	  ASTERISK-26418 #close

	  Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9

2016-10-18 16:30 +0000 [0abc71dfd6]  Mark Michelson <mmichelson@digium.com>

	* CDR: Alter destruction pattern for CDR chains.

	  CDRs form chains. When the root of the chain is destroyed, it then
	  unreferences the next CDR in the chain. That CDR is destroyed, and it
	  then unreferences the next CDR in the chain. This repeats until the end
	  of the chain is reached. While this typically does not cause any sort of
	  problems, it is possible in strange scenarios for the CDR chain to grow
	  way longer than expected. In such a scenario, the destruction pattern
	  can result in a stack overflow.

	  This patch fixes the problem by switching from a recursive pattern to an
	  iterative pattern for destruction. When the root CDR is destroyed, it is
	  responsible for iterating over the rest of the CDRs and unreferencing
	  each one. Other CDRs in the chain, since they are not the root, will
	  simply destroy themselves and be done. This causes the stack depth not
	  to increase.

	  ASTERISK-26421 #close
	  Reported by Andrew Nagy

	  Change-Id: I3ca90c2b8051f3b7ead2e0e43f60d2c18fb204b8

2016-10-18 11:51 +0000 [8bd6f695dc]  Joshua Colp <jcolp@digium.com>

	* ari: Update model validator based on addition of asterisk_id.

	  ASTERISK-26470

	  Change-Id: I9c386f7a1c7d969161b28f189eb6298bbc5b7541

2016-10-18 03:01 +0000 [3938e3320f]  Alexander Traud <pabstraud@compuserve.com>

	* cli: Auto-complete File not Module for core set debug.

	  Since Asterisk 1.8, the command "core set debug" on the command-line interface
	  asks not for a file (.c) but a module name. This change shows modules (.so) on
	  the auto-completion via a tabulator or the question mark. Now, when you
	  partially type a module name, TAB or ?, you get the correct candidiates.

	  ASTERISK-26480

	  Change-Id: I1213f1dd409bd4ff8de08ad80cb0c73cafb1bae0

2016-09-11 10:13 +0000 [47ba379e4c]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* menuselect: invalid test for GTK2

	  configuire.ac was only checking for the existence of pkg-config
	  and not the gtk2 package itself.  Now it calls AST_PKG_CONFIG_CHECK
	  for gtk+-2.0.

	  ASTERISK-26356 #close

	  Change-Id: I8079d515d6ea99f9ab320a7eaa71c2aaa101ccd5

2016-10-13 14:09 +0000 [ef8c54238c]  Leandro Dardini <ldardini@gmail.com>

	* app_queue: Added initialization for "context" parameter

	  When using Asterisk Realtime Architecture, empty fields are skipped and the
	  default values are used. If the "context" parameter in queue was set and then
	  cleared from the database, the old value remains in memory and it continues
	  to be used. This change initialize the "context" parameter with an empty value,
	  allowing clearing the parameter.

	  ASTERISK-26462 #close

	  Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905

2016-10-15 20:05 +0000 [ebcbc9ee34]  Matt Jordan <mjordan@digium.com>

	* res/ari: Add the Asterisk EID field to outgoing events

	  This patch adds the Asterisk EID field to all outgoing ARI events.
	  Because this field should be added to all events as they are
	  transmitted, it is appended to the JSON message just prior to it being
	  handed off to the application message handler. This makes it somewhat
	  resilient to both new events being added to ARI, as well as other
	  potential event transport mechanisms.

	  ASTERISK-26470 #close

	  Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d

2016-10-13 02:06 +0000 [3135b15a54]  Moises Silva <moises.silva@gmail.com>

	* chan_rtp: Set a sane default rtp engine for unicast.

	  ASTERISK-26439

	  Change-Id: I7f5ee2eeba8906e9ecb3293dbe3a747770bb5011

2016-10-16 17:25 +0000 [13aa241b9c]  gtjoseph <gjoseph@digium.com>

	* utils.c:  Fix ast_set_default_eid for multiple platforms

	  ast_set_default_eid was searching for ethX, emX, enoX, ensX and even
	  pciD#U interface names.  While this was a good attempt, it wasn't
	  inclusive enough to capture interfaces like enp6s0 or ens6d1, etc.

	  Rather than relying on interface names, we now simply find the first
	  interface returned by the OS that has a hardware address and that
	  address isn't all 0x00 or all 0xff.  The code IS different for BSD,
	  Solaris and Linux based on what method is available for enumerating
	  interfaces.

	  Tested on:
	  FreeBSD9
	  CentOS6
	  Ubuntu14
	  Fedora24

	  I was unable to test on Solaris at this time but the code for Solaris
	  is used elsewhere at Digium.

	  Change-Id: Iaa6db87ca78a9a375e47d70e043ae08c1448cb72

2016-10-11 06:55 +0000 [ca1b197765]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia.

	  In the SIP channel driver chan_sip, auto_comedia was expected to be used in
	  tandem with auto_force_rport. Or stated differently: Only when auto_force_rport
	  was chosen (the default), auto_comedia worked. This change allows auto_comedia
	  to be set independently of the state of (auto_)force_rport. For example,
	  nat=force_rport,auto_comedia is useful for IPv4/IPv6 Dual Stack deployments
	  when IPv6 clients are behind a Firewall.

	  ASTERISK-26457 #close

	  Change-Id: Ib29d66c6dbb61648e371e01fc36c6978ddae5bc2

2016-10-17 20:05 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 14.1.0-rc1 Released.

2016-10-17 11:39 +0000 [230dd6f9af]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Add patch to address SSL crash

	  Addresses crashes when an attempt is made to operate on an SSL socket
	  after the socket has been closed.

	  ASTERISK-26477 #close

	  Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002

2016-10-15 04:58 +0000 [60c84bbad5]  Michael Kuron <m.kuron@gmx.de>

	* chan_sip: Only send video on outgoing channel if incoming channel supports it

	  Previously, the settings videosupport=always and videosupport=yes behaved
	  identically and unconditionally caused a video offer to be sent in the SDP on
	  an outgoing call. This was a regression introduced with commit
	  5a1d90e1fbfc4b48927aad55311f3b38efbf1f54 in Asterisk 1.6.1.

	  This commit restores correct behavior: videosupport=always causes a video offer
	  to be sent unconditionally, while videosupport=yes will only offer video on an
	  outbound channel if the incoming channel it is bridged to also supports video.
	  That way, the device receiving the outgoing call can display the correct user
	  interface elements for audio or video and will not unnecessarily show a blank
	  video window on an audio-only call.

	  ASTERISK-17470 #close

	  Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae

2016-10-14 00:18 +0000 [11bc810c86]  Corey Farrell <git@cfware.com>

	* Fix issues with bundled pjproject cached download.

	  Previously when testing I had a preexisting makeopts in ASTTOPDIR.  The
	  ordering of configure.ac causes --with-externals-cache to be processed
	  after third-party configure.  In cases where the Asterisk clone is
	  cleaned it would cause pjproject to be downloaded to /tmp.  This
	  moves processing of the externals cache and sounds cache to happen
	  before third-party configure.

	  This also addresses a possible issue with the third-party Makefile.  If
	  TMPDIR is set by the environment it would override the path given to
	  --with-externals-cache.

	  ASTERISK-26416

	  Change-Id: Ifab7f35bfcd5a31a31a3a4353cc26a68c8c6592d

2016-10-12 16:24 +0000 [aba27b5a60]  Richard Mudgett <rmudgett@digium.com>

	* Audit ast_json_pack() calls for needed UTF-8 checks.

	  Added needed UTF-8 checks before constructing json objects in various
	  files for strings obtained outside the system.  In this case string values
	  from a channel driver's peer and not from the user setting channel
	  variables.

	  * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
	  object construction.

	  ASTERISK-26466
	  Reported by: Richard Mudgett

	  Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096

2016-10-12 16:20 +0000 [beae5022bf]  Richard Mudgett <rmudgett@digium.com>

	* json: Check party id name, number, subaddresses for UTF-8.

	  * Updated unit test as ast_json_name_number() is now NULL tolerant.

	  ASTERISK-26466 #close
	  Reported by: Richard Mudgett

	  Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6

2016-10-11 18:14 +0000 [2433a9d0c0]  Richard Mudgett <rmudgett@digium.com>

	* json: Add UTF-8 check call.

	  Since the json library does not make the check function public we
	  recreate/copy the function in our interface module.

	  ASTERISK-26466
	  Reported by: Richard Mudgett

	  Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99

2016-10-12 17:42 +0000 [b288b8b997]  Richard Mudgett <rmudgett@digium.com>

	* aoc.c: Whitespace cleanup

	  * In s_to_json() removed unnecessary ast_json_ref() to ast_json_null()
	  when creating the type json object.  The ref is a noop.

	  Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a

2016-10-12 16:22 +0000 [857a9936f4]  Richard Mudgett <rmudgett@digium.com>

	* app_minivm.c: Fix malformed ast_json_pack() call.

	  Change-Id: I082b239022fac462666e52a14a44304748908dc0

2016-10-12 17:27 +0000 [030852e1b3]  Richard Mudgett <rmudgett@digium.com>

	* app_queue.c: Fix clearing of pause reason string.

	  The pause reason is not always cleared when it should be cleared.

	  * Made set_queue_member_pause() always clear pause reason if not pausing
	  with a reason string.

	  Change-Id: I993dad19626ec017478a230e980989438b778c53

2016-10-12 16:30 +0000 [bb08fae6a9]  gtjoseph <gjoseph@digium.com>

	* res_config_mysql:  Fix several issues related to recent table changes

	  Unlike any of the other database drivers, res_config_mysql checks that
	  the table definition matches the requirements for every insert and
	  update statement.  Since all requirements are forced to 'char', any
	  column that isn't a char, like ps_contacts' expiration_time,
	  qualify_timeout, etc., will throw a warning.  It's kinda harmless but
	  very misleading.  Since no other driver does those checks on insert
	  or update, they've been removed from res_config_mysql.  Also, all
	  the logic that actually attempted to ALTER the table to fix the issue
	  has been removed.  With the move to alembic, the auto-alter
	  functionality is not only unnecessary, it's also dangerous.

	  The other issue is that res_config_mysql calls the mysql_insert_id
	  function inside store_mysql.  Presumably the intention was to return
	  the number of rows inserted DESPITE A NOTE IN THE CODE THAT THE VALUE
	  IS NON_PORTABLE AND MAY CHANGE.  That value is then returned to
	  config realtime as the number of rows inserted.  Guess what?  The value
	  changed.  It now only returns the number of rows inserted if there's an
	  auto increment column on the table, which ps_contacts doesn't have.
	  Otherwise it returns 0.  So now, the insert worked but we tell config
	  realtime and sorcery that no rows were inserted.  That call to
	  mysql_insert_id was removed and we now always return 1 if the insert
	  succeeded.  We're only inserting 1 row at a time anyway.  If the insert
	  fails, we still return -1.

	  ASTERISK-26362 #close
	  Reported-by: Carlos Chavez

	  Change-Id: I83ce633efdb477b03c8399946994ee16fefceaf4

2016-09-29 13:08 +0000 [8d2d6361ff]  Torrey Searle <torrey@voxbone.com>

	* res_fax: Fix a tight race condition causing fax to crash in audio fallback

	  When T.38 gets rejected and G711 failback occurs there is a period of
	  time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set,
	  leading to a crash.

	  Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982

2016-09-30 16:29 +0000 [bbbd43924f]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* Add text of cdr directory into README.md for ast-db-manage

	  Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636

2016-10-06 09:58 +0000 [90f8ba8800]  gtjoseph <gjoseph@digium.com>

	* app_dial:  Add the "Q" option to set the cause on unanswered channels

	  The "Q" option will set the cause on the unanswered channels when
	  another channel answers.  It overrides the default of
	  ANSWERED_ELSEWHERE.

	  NOTE:  chan_sip does not support setting the cause on a CANCEL to
	  anything other than ANSWERED_ELSEWHERE.

	  ASTERISK-26446 #close

	  Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47

2016-10-10 16:59 +0000 [012074d46b]  Badalyan Vyacheslav <v.badalyan@open-bs.ru>

	* vector: After remove element recheck index

	  Small fix. It is necessary to double-check
	  the index that we just removed because there
	  is a new element.

	  ASTERISK-26453 #close

	  Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7

2016-09-29 12:52 +0000 [e193217f1a]  Torrey Searle <torrey@voxbone.com>

	* res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge

	  If a bridge switched to P2P when a DTMF was in progress it
	  was possible for the DTMF to continue being sent indefinitely.

	  Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29

2016-10-04 20:46 +0000 [4cf2d9f261]  Michael Walton <mike@farsouthnet.com>

	* audiohooks: Remove redundant codec translations when using audiohooks

	  The main frame read and write handlers in main/channel.c don't use the
	  optimum placement in the processing flow for calling audiohooks
	  callbacks, as far as codec translation is concerned. This change places
	  the audiohooks callback code:
	   * After the channel read translation if the frame is not linear before
	  the translation, thereby increasing the chance that the frame is linear
	  as required by audiohooks
	   * Before the channel write translation if the frame is linear at this
	  point
	  This prevents the audiohooks code from instantiating additional
	  translation paths to/from linear where a linear frame format is already
	  available, saving valuable CPU cycles

	  ASTERISK-26419

	  Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f

2016-10-10 10:59 +0000 [188237fa82]  Badalyan Vyacheslav <v.badalyan@open-bs.ru>

	* res_pjsip_config_wizard: Memory leak in module_unload

	  Fixed a memory leak. It removes only the first element.
	  Added a useful feature in vector.h to remove all items
	  under the CMP through a callback function / macro.

	  ASTERISK-26453 #close

	  Change-Id: I84508353463456d2495678f125738e20052da950

2016-10-09 21:53 +0000 [c3be9f9fa8]  Badalyan Vyacheslav <v.badalyan@open-bs.ru>

	* cel_odbc: Fix memory leak on module unload

	  Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715

2016-10-03 11:30 +0000 [a0d02f3832]  gtjoseph <gjoseph@digium.com>

	* bundled_pjproject:  Add tests for programs used by the Makefile, et al.

	  Added tests for bzip2, tar, patch, sed and nm to configure.ac.

	  Set DOWNLOAD_TO_STDOUT to a working command line regardless of
	  whether the download program is wget, curl or fetch.

	  Added a 'configure.m4' file to the third-party directory which takes
	  care of calling any third-party project setup.  Had to move some
	  pjproject_bundled stuff up in configure.ac so it was called before
	  the third-party configure macro.

	  The pjproject tarball is now downloaded to the externals_cache_dir if
	  it was specified on the ./configure command line

	  Removed regeneration of the pjproject aconfigure file.  It was only
	  needed for an old patch that no longer applies.

	  Converted the tests for symbols to explicit tests since we know that
	  they're now available in the bundled version.  Saves a little time
	  during configure.

	  ASTERISK-26416 #close
	  Reported-by: Corey Farrell

	  Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b
	  (cherry picked from commit e6b0053d7561032b7adbf6f3afaecf30f5046605)

2016-10-09 21:28 +0000 [2f2ef9b8a6]  Corey Farrell <git@cfware.com>

	* logger: Prevent output of verbose messages initiated from rasterisk.

	  Remote asterisk consoles should only display verbose log messages
	  created by the daemon.  The first patch for ASTERISK-26410 caused
	  a couple verbose messages to be printed when the rasterisk process
	  ended.

	  ASTERISK-26410

	  Change-Id: Ie2a1bb3753ad2724c0349ec1a336f52f7117b52a

2016-10-05 14:53 +0000 [b39bbc0c6a]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Add MALLOC_DEBUG capability

	  pjproject_bundled will now use the asterisk memory debugging APIs
	  if MALLOC_DEBUG is turned on in menuselect.

	  Because this required stubs for the executable programs and the python
	  bindings, some Makefile reorganization was needed to properly handle
	  the dependencies.  As a result, the makefile now individually makes
	  each of the pjproject libraries separately instead of making them all
	  in 1 shot.  The only visible change is that there are separate status
	  lines printed for each library instead oif 1 for all libs.  Also, the
	  making of the pjproject dependency files was eliminated.  They're not
	  needed for building unless you're actively modifying pjproject source
	  files and it makes the build process faster.  Finally, any issues with
	  parallel builds should be resolved again making the build faster.

	  Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0

2016-10-04 16:59 +0000 [741c2e1570]  gtjoseph <gjoseph@digium.com>

	* alembic:  Allow cdr, config and voicemail to exist in the same schema

	  cdr, config and voicemail are all separate alembic trees.  Because
	  alembic's default is to use a table named 'alembic_version' to store
	  the current tree revision, the 3 trees can't exist in the same schema
	  without stepping on each other.

	  Now each tree uses 'alembic_version_<tree_name>' as the version table.
	  Each tree's env.py script now first checks for 'alembic_version'.  If
	  it finds it AND its revision is in the tree's history, the script
	  renames it to 'alembic_version_<tree_name>'.  Regardless, the script
	  then continues with the migration using 'alembic_version_<tree_name>'
	  and creates that table if it's not found.  The result is that if an
	  existing 'alembic_version' table was found but it didn't belong to this
	  tree, it's left alone and 'alembic_version_<tree_name>' is used or
	  created.

	  WARNING:  If multiple trees are using the same schema, they MUST NOT
	  CRU or D any objects with names that might exist in the other trees.
	  An example would be 'yesno_values' type.  If two trees perform
	  operations on it, one tree could pull it out from under the other.
	  Thankfully we currently don't share any names among cdr, config and
	  voicemail.

	  NOTE:  Since the env.py scripts in each tree were identical, a common
	  env.py has been placed in the ast-db-manage directory and a symlink
	  to it has been placed in each tree directory.

	  ASTERISK-24311 #close
	  Reported-by: Dafi Ni

	  Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898

2016-09-16 18:54 +0000 [d236973657]  Corey Farrell <git@cfware.com>

	* astobj2: Add backtrace to log_bad_ao2.

	  * Compile __ast_assert_failed unconditionally.
	  * Use __ast_assert_failed to log messages from log_bad_ao2
	  * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run.

	  Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751

2016-10-05 04:25 +0000 [16d9d67329]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: Honor support of Symmetric Response (rport) for SIP requests.

	  In the SIP channel driver chan_sip, the default is "auto_force_rport". When no
	  NAT was detected, for example in case of IPv6, Asterisk uses the IP address
	  from the headers within the SIP-REGISTER for subsequent SIP signaling. When
	  the remote party specifies support for Symmetric Response (RFC 3581) via the
	  parameter "rport", Asterisk should not extract the port from the SIP headers
	  but reuse the port of the transport. This did not happen because of a typo.

	  ASTERISK-26438 #close

	  Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6

2016-09-29 14:02 +0000 [65d0cbea1b]  Kevin Harwell <kharwell@digium.com>

	* Remove "format_ogg_opus: New format"

	  This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c.

	  ASTERISK-26426 #close

	  Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5

2016-09-27 16:10 +0000 [ea29ff9023]  gtjoseph <gjoseph@digium.com>

	* download_externals: Fix issue with re-install

	  Needed to ignore an xmlstarlet return code for optional element.

	  Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873
	  Reported-by: Dan Jenkins

2016-09-27 15:35 +0000 [013414ca4b]  Corey Farrell <git@cfware.com>

	* logger: Output early verbose messages to console.

	  Verbose messages should be printed to the console if the sublevel is
	  less than option_verbose.  This fix ensures the welcome message with
	  copyright and license are printed at daemon and interactive rasterisk
	  startup.

	  ASTERISK-26410 #close

	  Change-Id: Ia44235e30ec328aba92ea2c8a837b094e65c9a03

2016-09-22 09:49 +0000 [f243be6a8c]  gtjoseph <gjoseph@digium.com>

	* codec_opus: Add download ability to menuselect

	  Updated codecs/codecs.xml to add codec_opus to the external
	  download list.

	  ASTERISK-26409

	  Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4
	  (cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c)

2016-07-23 14:50 +0000 [550608db04]  gtjoseph <gjoseph@digium.com>

	* codec_opus: Replace res_format_attr_opus with the one from codec_opus

	  Preparation

	  ASTERISK-26409

	  Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3

2016-07-23 15:56 +0000 [b4642fd6a6]  gtjoseph <gjoseph@digium.com>

	* format_ogg_opus: New format

	  Add Ogg/Opus playback support.

	  This uses libopusfile in order to be able to read .opus files and play
	  them back.

	  Writing/recording support is not present at this time.

	  ASTERISK-26409

	  Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955

2016-09-24 19:05 +0000 [ce6405a234]  gtjoseph <gjoseph@digium.com>

	* build_tools:  Add ability to download variants to download_externals

	  Some external packages have multiple variants that apply to different
	  builds of asterisk.  The DPMA for instance has a "bundled" variant that
	  needs to be downloaded if asterisk was configured with
	  --with-pjproject-bundled.

	  There are 2 ways to specify variants:

	  If you need the user to make the decision about which variant to
	  download, simply create multiple menuselect "member" entries like so...

	  <member name="res_digium_phone" displayname="..snipped..">
	    <support_level>external</support_level>
	    <depend>xmlstarlet</depend>
	    <depend>bash</depend>
	    <defaultenabled>no</defaultenabled>
	  </member>

	  <member name="res_digium_phone-bundled" displayname="..snipped..">
	    <support_level>external</support_level>
	    <depend>xmlstarlet</depend>
	    <depend>bash</depend>
	    <defaultenabled>no</defaultenabled>
	  </member>

	  Note that the second entry has "-<variant>" appended to the name.
	  You can then use the existing menuselect facilities to restrict which
	  members to enable or disable.  Youy probably don't want the user to
	  enable multiple at the same time.

	  If you want to hide the details of the variants, the better way to
	  do it is to create 1 member with "variant" elements.

	  <member name="res_digium_phone" displayname="..snipped..">
	    <support_level>external</support_level>
	    <depend>xmlstarlet</depend>
	    <depend>bash</depend>
	    <defaultenabled>no</defaultenabled>
	    <member_data>
	      <downloader>
	        <variants>
	          <variant tag="bundled"
	            condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/>
	        </variants>
	      </downloader>
	    </member_data>
	  </member>

	  The condition must be a bash expression suitable for use with an "if"
	  statement.  Any environment variable can be used plus those available
	  in makeopts.

	  In this case, if asterisk was configured with --with-pjproject-bundled
	  the bundled variant will be automatically downloaded.  Otherwise the
	  normal version will be downloaded.

	  Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e

2016-09-22 01:40 +0000 [7bcb1fc940]  Aaron An <anjb@ti-net.com.cn>

	* channels/chan_pjsip: fix HANGUPCAUSE function bug.

	  HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered.
	  This patch change the call order of ast_queue_control_data
	  and ast_queue_control in chan_pjsip_incoming_response.

	  ASTERISK-26396 #close
	  Reported by: AaronAn
	  Tested by: AaronAn

	  Change-Id: Ide2d31723d8d425961e985de7de625694580be61

2016-09-23 09:54 +0000 [175975e994]  Alessandro Crespi

	* chan_sip: Resolve externhost not to IPv6; instead go for IPv4.

	  For the channel driver chan_sip, you specify externhost=example.com in sip.conf
	  when your Asterisk is behind a NAT and your IP address is assigned dynamically.
	  Or stated differently: You do not have a static IP address to use "externaddr"
	  directly. This NAT support is quite handy but just about IPv4. Previously,
	  Asterisk resolved "externhost" to any IP version. When the first DNS answer
	  resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and
	  connection (c=). This happened in outgoing SIP-REGISTER and while answering
	  SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an
	  IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost".

	  ASTERISK-18232 #close
	  Reported by: Jacek Kowalski
	  Tested by: Alexander Traud
	  patches:
	   changes.patch submitted by Alessandro Crespi

	  Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac

2016-09-20 09:42 +0000 [d64edafa63]  gtjoseph <gjoseph@digium.com>

	* chan_sip:  Address runaway when realtime peers subscribe to mailboxes

	  Users upgrading from asterisk 13.5 to a later version and who use
	  realtime with peers that have mailboxes were experiencing runaway
	  situations that manifested as a continuous stream of taskprocessor
	  congestion errors, memory leaks and an unresponsive chan_sip.

	  A related issue was that setting rtcachefriends=no NEVER worked in
	  asterisk 13 (since the move to stasis).  In 13.5 and earlier, when a
	  peer tried to register, all of the stasis threads would block and
	  chan_sip would again become unresponsive.  After 13.5, the runaway
	  would happen.

	  There were a number of causes...
	  * mwi_event_cb was (indirectly) calling build_peer even though calls to
	    mwi_event_cb are often caused by build_peer.
	  * In an effort to prevent chan_sip from being unloaded while messages
	    were still in flight, destroy_mailboxes was calling
	    stasis_unsubscribe_and_join but in some cases waited forever for the
	    final message.
	  * add_peer_mailboxes wasn't properly marking the existing mailboxes
	    on a peer as "keep" so build_peer would always delete them all.
	  * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions
	    then just creating them again.

	  All of this was causing a flood of subscribes and unsubscribes on
	  multiple threads all for the same peer and mailbox.

	  Fixes...
	  * add_peer_mailboxes now marks mailboxes correctly and build_peer only
	    deletes the ones that really are no longer needed by the peer.
	  * add_peer_mwi_subs now only adds subscriptions marked as "new" instead
	    of unsubscribing and resubscribing everything.  It also adds the peer
	    object's address to the mailbox instead of its name to the subscription
	    userdata so mwi_event_cb doesn't have to call build_peer.

	  With these changes, with rtcachefriends=yes (the most common setting),
	  there are no leaks, locks, loops or crashes at shutdown.

	  rtcachefriends=no still causes leaks but at least it doesn't lock, loop
	  or crash.  Since making rtcachefriends=no work wasnt in scope for this
	  issue, further work will have to be deferred to a separate patch.

	  Side fixes...
	   * The ast_lock_track structure had a member named "thread" which gdb
	     doesn't like since it conflicts with it's "thread" command.  That
	     member was renamed to "thread_id".

	  ASTERISK-25468 #close

	  Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0

2016-09-21 14:24 +0000 [64df75b02c]  Joshua Colp <jcolp@digium.com>

	* core: Ensure presencestate subtype and message are NULL.

	  When retrieving presence state information there is no
	  guarantee that the subtype and message passed in are
	  set to NULL. This change ensures they are.

	  ASTERISK-26397 #close

	  Change-Id: If38cd730e409e9a9b6eb9adef6591d15a9e61f86

2016-09-21 10:48 +0000 [f63ff293eb]  Joshua Colp <jcolp@digium.com>

	* res_odbc: Make pooling option deprecation notice more useful.

	  This changes the notice for the deprecation of the old
	  pooling options to point to the new option for doing
	  pooling. This gives a clearer direction as to what to
	  look into.

	  ASTERISK-26389 #close

	  Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10

2016-09-12 07:37 +0000 [fc16739600]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* cdr_mysql: fix UTC support

	  * Make 'cdrzone=UTC' work properly.
	  * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone

	  ASTERISK-26359 #close

	  Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778

2016-09-21 08:46 +0000 [2684ffd9cd]  Joshua Colp <jcolp@digium.com>

	* odbc: Remove options that are no longer applicable.

	  The pooling, shared_connection, limit, and idlecheck options
	  are no longer used in res_odbc.

	  ASTERISK-26389

	  Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6

2016-09-20 15:17 +0000 [ba3269f12b]  Corey Farrell <git@cfware.com>

	* core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get.

	  Move the function outside the conditional block that excludes
	  LOW_MEMORY.

	  ASTERISK-26273 #close

	  Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4

2016-08-16 15:21 +0000 [1a7d39818f]  Corey Farrell <git@cfware.com>

	* logger: Simplify ast_callid handling code.

	  Routines responsible for managing ast_callid's are overly complicated.
	  This is left-over code from when ast_callid was an AO2 object.  Now that
	  it is an integer the code can be reduced.

	  ast_callid handler code no longer prints it's own error message upon failure
	  to allocate threadstorage as ast_calloc would have already printed a
	  message.  Debug messages that were printed when TEST_FRAMEWORK was
	  enabled have been also been removed.

	  Change-Id: I65a768a78dc6cf3cfa071e97f33ce3dce280258e

2016-09-20 09:22 +0000 [1b61a21757]  Corey Farrell <git@cfware.com>

	* logger: Always enable verbose for console channel.

	  Previous versions of Asterisk did not require verbose to be specified in
	  logger.conf for the console channel, if it was requested by command line
	  or asterisk.conf it just worked.  This change causes Asterisk to always
	  enable verbose in the console channel level mask.  Verbose is displayed
	  on consoles if requested by command line, option_verbose or 'core set
	  verbose'.

	  This also delays initialization of the logger until after threadstorage
	  is initialized.  Initializing too early can cause messages to be printed
	  multiple times to the console (stdout).

	  ASTERISK-26391 #close

	  Change-Id: I52187d67c2fcb3efd5561bf04b3e5e23e5ee8a04

2016-09-20 10:16 +0000 [68ff7362ae]  Corey Farrell <git@cfware.com>

	* logger: Fix default console settings.

	  When logger.conf is missing or invalid we should be printing notices,
	  warnings and errors to the console.  The logmask was incorrectly
	  calculated.

	  Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3

2016-06-27 14:26 +0000 [29e096cd13]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* sd_notify (systemd status notifications) support

	  sd_notify() is used to notify systemd of changes to the status of the
	  process. This allows the systemd daemon to know when the process
	  finished loading (and thus only start another program after Asterisk has
	  finished loading).

	  To use this, use a systemd unit with 'Type=notify' for Asterisk.

	  This commit also adds the function ast_sd_notify(), a wrapper around
	  sd_notify that does nothing if not built with systemd support.

	  Also adds support for libsystemd detection in the configure script.

	  Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811
	  (cherry picked from commit 07b95f7c65b7c083724f1af2b26f93cc22cad58c)

2016-09-19 14:21 +0000 [2b75559e6c]  Walter Doekes <walter+github@wjd.nu>

	* asterisk.c: Non-root users also get the astcanary after core restart.

	  Without this change, a 'core restart' would kill the astcanary forever
	  if you're not running as root. Both with and without this patch, the
	  scheduling priority was still SCHED_RR after restart.

	  Additionally, the astcanary is now spawned if you start with high
	  priority and Asterisk doesn't get a chance to lower it. For example
	  through: `chrt -r 10 sudo -u asterisk asterisk -c`

	  Also reap killed astcanary processes on core restart.

	  ASTERISK-26352 #close

	  Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55

2016-09-19 09:40 +0000 [bd39d2c0ab]  Walter Doekes <walter+github@wjd.nu>

	* asterisk.c: When astcanary dies on linux, reset priority on all threads.

	  Previously only the canary checking thread itself had its priority set
	  to SCHED_OTHER. Now all threads are traversed and adjusted.

	  ASTERISK-19867 #close
	  Reported by: Xavier Hienne

	  Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39

2016-09-12 18:00 +0000 [4c61d9f629]  Richard Mudgett <rmudgett@digium.com>

	* res_config_odbc.c: Fix buffer size limitation creating invalid SQL.

	  Creating ODBC SQL queries resulted in queries too large to fit into the
	  supplied buffer.  The resulting truncated buffer contained an invalid SQL
	  query.

	  * Made SQL query generation code use a thread storage buffer that can
	  increase in size as needed.

	  * Fixed bad multi-line warning messages.

	  ASTERISK-26263 #close
	  Reported by: Jeppe Ryskov Larsen

	  Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae

2016-09-09 06:35 +0000 [1e9ae453fd]  Timo Teräs <timo.teras@iki.fi>

	* Fix showing of swap details when sysinfo() is available

	  If sysinfo() is available, but not sysctl() or swapctl() the
	  printing code for swap buffer sizes is incorrectly omitted.
	  The above condition happens with musl c-library.

	  Fix #if rule to consider defined(HAVE_SYSINFO). And also
	  remove the redundant || defined(HAVE_SYSCTL) which was
	  incorrectly there to start with. Now swap information is
	  displayed only if an actual libc function to get it is
	  available.

	  This also fixes warnings previously seen with musl libc:

	     [CC] asterisk.c -> asterisk.o
	  asterisk.c: In function 'handle_show_sysinfo':
	  asterisk.c:773:6: warning: variable 'totalswap' set but not used
	   [-Wunused-but-set-variable]
	    int totalswap = 0;
	        ^~~~~~~~~
	  asterisk.c:770:11: warning: variable 'freeswap' set but not used
	   [-Wunused-but-set-variable]
	    uint64_t freeswap = 0;
	             ^~~~~~~~

	  Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca

2016-09-14 06:53 +0000 [696824df89]  Joshua Colp <jcolp@digium.com>

	* rtp: Only accept the first payload for a format in SDP.

	  When receiving an SDP offer with multiple payloads for
	  the same format we would generate an answer with the first
	  payload, but during the payload crossover operation
	  (to set the payloads for receiving) we would remove all
	  payloads but the last. This would result in incoming
	  traffic being matched against the wrong format and outgoing
	  traffic being sent using the wrong payload.

	  This change makes it so that once a format has a payload
	  number put into the mapping all subsequent ones are ignored.
	  This ensures there is only ever one payload in the mapping
	  and that it is the payload placed into the answer SDP.

	  ASTERISK-26365 #close

	  Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60

2016-09-13 05:34 +0000 [58f41ebf93]  Steve Davies <steve@one47.co.uk>

	* chan_sip: Fix session timeout on retransmit of non-UDP packets

	  Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for
	  SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP
	  connections, allowing the TCP layer to handle the retransmits. Unfortunately,
	  this caused sessions to be terminated with a retransmit timeout becasue it
	  stopped at the point of the first retrans call.

	  This patch waits for the 64*T1 timer to expire instead.

	  ASTERISK-19968

	  Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204
	  (cherry picked from commit 98e42cc6624a02bede84c38772412b6ff9d8fa2f)

2016-09-14 08:42 +0000 [f1c06322ba]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_multihomed: Change Contact port to listening port.

	  The res_pjsip_multihomed module determines what interface and transport
	  a request is going out on and updates the SIP message accordingly with
	  the address information. This currently incorrectly updates the Contact
	  header for connectionful protocols to the ephemeral connection port,
	  instead of the bound address for the listening socket which can actually
	  accept the connection back. If the remote side attempts to connect back on
	  the epehemeral port it will fail.

	  This change makes it so the port is updated to the bound port on
	  connectionful protocols and is maintained on UDP (as there can be
	  multiple of those).

	  ASTERISK-26374 #close

	  Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab

2016-09-07 14:48 +0000 [6a5683cc27]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Prevent SERVFAIL from marking name server bad

	  A name server that returns "Server Failure" is indicating only that
	  the server couldn't process that particular request.  We should NOT
	  assume that the name server is incapable of serving other requests.

	  Here's the scenario we've been encountering...

	  * 2 local name servers configured in resolv.conf.
	  * An OPTIONS request causes a request for A and AAAA records to go out
	    to both nameservers.
	  * The A responses both come back successfully resolved.
	  * Because of an issue at some upstream nameserver, the AAAA responses
	    for that particular query come back as "SERVFAIL" from both local
	    name servers.
	  * Both local servers are marked as bad and no further queries can be
	    sent until the 60 second ttl expires.  Only previously cached results
	    can be used.
	  * In this case, 60 seconds is just enough time for another OPTIONS
	    request to go out to the same host so the cycle repeats.

	  We could set the bad ttl really low but that also affects REFUSED and
	  NOTAUTH which probably DO signal a real server issue.  Besides, even
	  a really low bad ttl would be an issue on a pbx.

	  Although we use our own resolver in 14 and master and don't have this
	  issue there, Teluu has merged this patch upstream so it's appropriate
	  to cherry-pick to 14 and master to keep pjproject consistent.


	  Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0

2016-09-14 07:59 +0000 [34e82d16b4]  Joshua Colp <jcolp@digium.com>

	* rtp: Preserve timestamps on video frames.

	  Currently when receiving video over RTP we store only
	  a calculated samples on the frame. When starting the video
	  it can take some time for this calculation to actually yield
	  a value as it requires constant changing timestamps. As well
	  if a video frame passes over multiple RTP packets this calculation
	  will fail as the timestamp is the same as the previous RTP
	  packet and the number of samples calculated will be 0.

	  This change preserves the timestamp on the frame and allows
	  it to pass through the core. When sending the video this timestamp
	  is used instead of a new one being calculated.

	  ASTERISK-26367 #close

	  Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd

2016-09-14 09:51 +0000 [70e6963c5f]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_transport_management: Convert time in log message to seconds.

	  ASTERISK-26375 #close

	  Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc

2016-07-22 05:46 +0000 [b3802e68b8]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: Enable Session-Timers for SIP over TCP (and TLS).

	  Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that
	  scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables
	  Session-Timers for SIP over TCP (and for SIP over TLS).

	  However with longer international calls via TCP, the SIP channel might break,
	  because all hops on the Internet route must stay online (have not a single power
	  outage, for example). Therefore with Session-Timers enabled (which are enabled
	  at default), you might see dropped calls. Consequently even with this change,
	  you might be better-off going for session-timers=refuse in your sip.conf.

	  ASTERISK-19968 #close

	  Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957
	  (cherry picked from commit 66c9dfb272322b21192f58383ae519ceb44e474c)

2016-09-13 06:08 +0000 [b6207cab24]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Don't assume a request will have any addresses.

	  When performing DNS resolution the failover code present in
	  res_pjsip currently assumes that a request will always have
	  at least one viable address. In practice this is not true.
	  A domain may be used that has no records.

	  The code now checks that at least one address exists on the
	  request which prevents looping.

	  ASTERISK-26364 #close

	  Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c

2016-09-12 12:25 +0000 [d910a51b33]  Richard Mudgett <rmudgett@digium.com>

	* app_queue: Fix CLI "queue show" and AMI Queues action output truncation.

	  The output of CLI "queue show" and AMI Queues action is truncated and
	  "failed to extend from 240 to 327" messages are generated if the queue
	  member and interface names are lengthy.

	  * Increase the string buffer size from 240 to 512 in order to accommodate
	  for more information fields added to the output since v1.8.

	  ASTERISK-26360 #close
	  Reported by: Richard Mudgett

	  Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d

2016-09-12 03:28 +0000 [52e6d03401]  Walter Doekes <walter+github@wjd.nu>

	* chan_sip: Allow target refresh (Contact update) on re-INVITE.

	  Previously, the Contact was stored only on initial INVITE and on any
	  18X and 200. That meant that after re-INVITEs from *us* the Contact
	  could get updated, but after re-INVITEs from the *peer*, it did not.

	  This changeset fixes this inconsistency, properly allowing target
	  refreshes through re-INVITES (RFC3261, 12.2).

	  If your strictrtp setting allows it, this change allows you to switch
	  the source IP of a connected/calling device mid-call with a simple
	  re-INVITE from the new IP.

	  ASTERISK-26358 #close

	  Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435

2016-08-31 15:22 +0000 [fd82304aa9]  Richard Mudgett <rmudgett@digium.com>

	* sip_to_pjsip.py: Map legacy_useroption_parsing.

	  Map the sip.conf general section legacy_useroption_parsing to the
	  new pjsip.conf global ignore_uri_user_options.

	  ASTERISK-26316
	  Reported by: Kevin Harwell

	  Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc

2016-08-29 18:08 +0000 [b7ec070cd7]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Add ignore_uri_user_options option.

	  This implements the chan_sip legacy_useroption_parsing option but with a
	  better name.

	  * Made the caller-id number and redirecting number strings obtained from
	  incoming SIP URI user fields always truncated at the first semicolon.
	  People don't care about anything after the semicolon showing up on their
	  displays even though the RFC allows the semicolon.

	  ASTERISK-26316 #close
	  Reported by: Kevin Harwell

	  Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62

2016-09-09 06:26 +0000 [d980085d38]  Walter Doekes <walter+github@wjd.nu>

	* contrib: Let safe_asterisk script continue without /dev/tty9.

	  If you use the safe_asterisk script, it uses hardcoded defaults before
	  running configurable values from /etc/asterisk/startup.d. The hardcoded
	  default has TTY=9. Some containerized environments don't have such a
	  TTY, and safe_asterisk would stop.

	  The custom configuration from /etc/asterisk/startup.d/* isn't read until
	  after it stopped, so changing TTY in a custom config did not help.

	  This changeset changes safe_asterisk to continue if the TTY setting was
	  untouched and /dev/tty9 and /dev/vc/9 aren't found.

	  Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc

2016-09-09 05:39 +0000 [236761a5b2]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Only invoke unidentified endpoint logic when unidentified.

	  The code was incorrectly invoking the unidentified logic when
	  an endpoint had actually been identified, causing log messages
	  to be output.

	  ASTERISK-26349 #close

	  Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f

2016-08-23 06:35 +0000 [d7d22a2be5]  Corey Farrell <git@cfware.com> (license 5909)

	* chan_sip: Don't allocate new RTP instances on top of old ones.

	  In some scenarios dialog_initialize_rtp can be called multiple times on
	  the same dialog.  This can cause RTP instances to be leaked along with
	  multiple file descriptors for each instance.

	  This change makes it so the existing RTP instances are destroyed and
	  not overwritten, stopping the memory leak.

	  ASTERISK-26272 #close
	  patches:
	    ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909)

	  Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73

2016-08-16 15:34 +0000 [f3e79d5870]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Do not crash on ACKs from unknown endpoints.

	  The endpoint identification PJSIP module is intended to identify which
	  endpoint an incoming request is from. If an endpoint is not identified,
	  then an artificial endpoint is used in its place when proceeding.

	  The problem is that the ACK request type is an exception to the rule.
	  The artificial endpoint is not used when processing an ACK. This results
	  in the possibility of having a NULL endpoint being used further on.

	  The reason ACK is an exception is an attempt not to spam security logs
	  with unidentified requests. Presumably, you've already logged the
	  unidentified request on the preceeding INVITE.

	  Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion
	  didn't cause an issue. A new change in 13.10 added endpoint ACL checking
	  shortly after endpoint identification. Because we are accessing a NULL
	  endpoint, this ACL check resulted in a crash.

	  The fix here is to be sure to retrieve the artificial endpoint for all
	  request types. ACKs still do not generate unidentified request security
	  events.

	  ASTERISK-26264 #close
	  Reported by nappsoft

	  AST-2016-006

	  Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703

2016-09-06 11:46 +0000 [057cfe0cbb]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_messaging.c: Misc cleanups and fixes.

	  * Eliminated RAII_VAR in get_outbound_endpoint().

	  * Simplify update_to() coding.  However, this function can only be a NoOp
	  because the To string can only be a URI and not a name-address formatted
	  string.

	  * Simplify update_from() coding.  Also fixed a code path modifying the
	  from string when the caller could still want to use the original string.

	  * Fixed msg_data_create() incompletely removing the "pjsip:" to then add
	  back the "sip:" string if needed.  The code didn't handle the "pjsip:sip:"
	  case because it left the colon after pjsip in the string.

	  Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db

2016-09-07 16:00 +0000 [f401187282]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Allow global headers to be overridden.

	  Currently when you add global headers from the dialplan both
	  the header in the dialplan and the globally configured header
	  are added to the resulting SIP INVITE. This change makes it
	  so the headers in the dialplan take precedence and are the
	  only ones added.

	  Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad

2016-08-10 15:14 +0000 [999b34301b]  Mark Michelson <mmichelson@digium.com>

	* ConfBridge: Make some announcements asynchronous.

	  Confbridge announcements tend to block a channel while they are being
	  played. In some circumstances, this is warranted since you want that
	  particular channel not to hear the announcement (Example: "John Doe has
	  entered the conference"). For others it makes less sense.

	  This change first introduces methods for playing sounds asynchronously
	  into the conference. This is very similar to how synchronous sounds are
	  played, except the channel initiating the playback does not wait for the
	  sound to complete before moving on.

	  Asynchronous announcements are used for two circumstances:
	  * Sounds played for a user after they have left the bridge
	  * Sounds that play first to a single user and then the rest of the
	    conference (if the channel and conference use the same language)

	  ASTERISK-26289 #close
	  Reported by Mark Michelson

	  Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a

2016-09-07 05:59 +0000 [7ec7d53983]  Joshua Colp <jcolp@digium.com>

	* res_resolver_unbound: Fix config documentation.

	  The code was referencing the config section as 'globals'
	  instead of 'general'. This change swaps it over to 'general'.

	  Change-Id: I9dfe7788f41c4a6754c77e103880dc1a747de7fe

2016-09-06 15:25 +0000 [827b235357]  Matt Jordan <mjordan@digium.com>

	* res/res_stasis_playback: Cancel the entire playlist when a stop occurs

	  Prior to this patch, a stop issued by a delete of a Playback resource
	  (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop
	  the current media URI playing. Subsequent URIs specified by a playback
	  operation would then proceed on, even though we had just indicated to
	  the User that the Playback was finished *and* after they had just
	  'deleted' the resource. Whoops.

	  This patch corrects it by bailing out of the sequence of URIs to play if
	  one of them is terminated with an AST_CONTROL_STREAM_STOP indication.

	  ASTERISK-26341 #close

	  Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42

2016-08-01 20:55 +0000 [8b6f9dde14]  gtjoseph <gjoseph@digium.com>

	* build: Add download capability for external packages

	  The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
	  http://downloads.digium.com/pub/telephony/ are now listed in the
	  "External" sections of the "Resource Modules" and "Codec Translators"
	  pages in menuselect.  Any that are selected will automatically be
	  downloaded and installed when "make install" is run.  Their LICENSE and
	  README (if avaialble) files will be installed to
	  ASTVARLIBDIR/documentation/thirdparty/<product_name>.

	  Example use with codecs:

	  The codecs/codecs.xml file is a menuselect style xml file that lists
	  the codecs to be included.  Their support levels are 'external', which
	  triggers the download and install, and defaultenabled is no.  Also
	  because codec_g729a is actually in a directory named codec_g729 on the
	  download server, the newly added 'member_data' element is used to
	  override the default of the directory name being the package name.  You
	  can use the 'directory_name' attribute to keep default base URL
	  (http://downloads.digium.com/pub/telephony/) but use the new directory,
	  or you use the 'remote_url' attribute to specify a full URL to the
	  download directory.  In this case, you must still follow the same
	  subdirectory naming conventions as that used for the packages located
	  at 'http://downloads.digium.com/pub/telephony'.

	  A new configure option '--with-externals-cache' was added and like
	  '--with-sounds-cache' it allows the installer to cache tarballs so
	  they're not downloaded every time.

	  To assist with the download and install process, each external package
	  now has a manifest.xml file that, among other things, contains a package
	  version and checksums for each file in the tarball.  The manifest is
	  saved to both the cache directory and ASTMODDIR and together with the
	  manifest.xml on the downloads site, tells the install scripts whether
	  a download and/or update is needed.

	  bash and xmlstarlet are required for downloader operation.  If they're
	  not installed, the external items in menuselect will be unavailable.

	  Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a

2016-08-18 14:45 +0000 [09e9dca0c5]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip_session: segfault on already disconnected session

	  On heavy loaded system the TCP/TLS incoming calls could be
	  disconnected by pjproject while these calls are being
	  processed by asterisk which could use the session's memory pools.
	  If the session in the disconnected state then the session memory
	  pools were already freed, so we get segfault.

	  This patch adds a lifetime control on an INVITE session to pjproject.
	  The lifetime of the session is manipulated by calling
	  pjsip_inv_add_ref/pjsip_inv_dec_ref.
	  This patch uses these functions to inform pjproject that the
	  session is in use.

	  This patch adds check if the session state is not disconnected
	  and also checks if the memory pool is not NULL.

	  This patch also places tasks 'session_end' and 'session_end_completion'
	  into session's serializer to avoid race condition.

	  ASTERISK-26291 #close

	  Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7

2016-09-06 02:41 +0000 [9b59ebee2f]  Walter Doekes <walter+github@wjd.nu>

	* chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP.

	  Certain SNOM phones send so-called "optional crypto" in their SDP body.
	  Regular SRTP setup looks like this:

	      m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101
	      a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...

	  SNOM-style "optional crypto" looks like this:

	      m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101
	      a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...

	  A crypto line is supplied, but the m-line does not have SAVP.

	  When res_srtp.so is *not* loaded, then chan_sip.so treats the optional
	  crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the
	  incoming call with the following message:

	      WARNING: process_sdp: Failed to receive SDP offer/answer with
	      required SRTP crypto attributes for audio

	  For platforms that want to start providing SRTP this presents a
	  compatibility problem.

	  This changeset lets chan_sip handle the SDP as if no crypto-line was
	  supplied: i.e. accept the call as regular RTP, just like it did before
	  res_srtp was loaded.

	  Now you'll get this informative warning instead:

	      WARNING: Ignoring crypto attribute in SDP because RTP transport is
	      insecure

	  ASTERISK-23989 #close
	  Reported by: Olle Johansson

	  Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2

2016-09-03 16:04 +0000 [3e054a3826]  Matt Jordan <mjordan@digium.com>

	* apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option

	  In any scenario in which the callee is not connected to the caller, the
	  current code in app_dial will crash due to raising a Dial End Stasis
	  Message after the callee channel has been hung up. This patch corrects
	  the error by simply moving the explicit hangup of the callee (peer)
	  channel until after the dial end message.

	  ASTERISK-25691 #close

	  Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d

2016-09-03 16:02 +0000 [15105c7494]  Matt Jordan <mjordan@digium.com>

	* apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5

	  If the callee selects option '5' using the Dial application's privacy
	  (P) option, the DIALSTATUS is erroneously set to ANSWER. This option
	  reflects the callee sending the caller to VoiceMail one time; the call
	  is definitely *not* ANSWERed in such a scenario. With this patch, the
	  DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that
	  is set when the 'send to VoiceMail every time' option is set.

	  ASTERISK-25691

	  Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358

2016-08-30 16:40 +0000 [323b74c6ca]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_registrar.c: Reduce stack usage in find_aor_name().

	  Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09

2016-08-29 18:06 +0000 [5bc8c91863]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_configuration.c: Ignore repeated identify by methods.

	  Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838

2016-08-30 17:26 +0000 [bdfa4f8a0a]  Richard Mudgett <rmudgett@digium.com>

	* config_global.c: Comments and a default expression adjustment.

	  Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3

2016-08-31 15:14 +0000 [2c5bead338]  Richard Mudgett <rmudgett@digium.com>

	* sip_to_pjsip.py: Map canreinvite as directmedia alias.

	  Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2

2016-08-31 15:37 +0000 [d4f01fb7b6]  Richard Mudgett <rmudgett@digium.com>

	* sip_to_pjsip.py: Fix typo converting outboundproxy registration.

	  Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15

2016-08-31 15:13 +0000 [4ffc59b4d5]  Richard Mudgett <rmudgett@digium.com>

	* sip_to_pjsip.py: Fix comment typo and tabs.

	  Change-Id: If35174614545727817d329c60ba4456c028941b5

2016-08-31 15:56 +0000 [bc5b3a7217]  Richard Mudgett <rmudgett@digium.com>

	* Sample configs: Eliminate false multiline comment block starts.

	  Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6

2016-09-02 11:36 +0000 [2a8d27e020]  Richard Mudgett <rmudgett@digium.com>

	* format_cap.c: Fix CLI "core show channeltype Surrogate" crash.

	  * Make ast_format_cap_get_names() NULL tolerant.

	  ASTERISK-26331 #close
	  Reported by: CGI.NET

	  Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3

2016-08-26 17:22 +0000 [e65f170d9c]  Corey Farrell <git@cfware.com>

	* sorcery: Create function ast_sorcery_lockable_alloc.

	  Create an alternative to ast_sorcery_generic_alloc which uses astobj2
	  shared locking. Use this new method for the 'struct ast_sip_aor' allocator.

	  Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f

2016-08-18 13:28 +0000 [3af3860151]  Corey Farrell <git@cfware.com>

	* named_locks: Use ao2_weakproxy to deal with cleanup from container.

	  This allows standard ao2 functions to be used to release references to
	  an ast_named_lock.  This change can cause less frequent locking of the
	  global named_locks container.  The container is no longer locked when a
	  named_lock reference is being release except when this causes the
	  named_lock to be destroyed.

	  Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6

2016-08-26 13:18 +0000 [e260e507f2]  Corey Farrell <git@cfware.com>

	* astobj2: Support using a separate object for locking.

	  Create ao2_alloc_with_lockobj function to support shared locking.

	  Change-Id: Iba687eb9843922be7e481e23a32c0700ecf88a80

2016-08-31 12:23 +0000 [dd408708c9]  Michael Kuron <m.kuron@gmx.de>

	* app_mp3: Use correct buffer size and the same sample rate as the channel

	  Previously, the buffer used for MP3 streamed from HTTP servers had a size of
	  1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1
	  minute. Only when the buffer is full does audio start to play.
	  For MP3 files streamed from a server, that is usually not a big deal as long as
	  the connection to the server is fast enough to supply that much data within a
	  second or two. For MP3 live streams however, it takes 1 minute to download 1
	  minute of audio, so without this change, app_mp3 wasn't really usable for MP3
	  live streams.
	  This commit changes the buffer size so that it covers 6 seconds of an MP3 file
	  streamed from a server and 0.5 seconds of an MP3 live stream. The latter is
	  identified by the use of a .m3u file extension.

	  app_mp3 so far only supported 8 kHz audio.
	  Now it always runs at the sample rate of the channel.

	  ASTERISK-26085 #close

	  Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0

2016-08-26 10:39 +0000 [411efb9754]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: qualify/unqualify added/deleted realtime endpoints

	  If the PJSIP endpoint's AOR with the permanent contact
	  was deleted from the realtime storage the res_pjsip module
	  continues trying to qualify this contact.
	  The error 'Unable to find an endpoint to qualify contact'
	  appeares every 'qualify_frequency' seconds.
	  This patch deletes this contact in this case.

	  The PJSIP endpoint's AOR with the permanent contact
	  is never qualified if it is added to realtime storage
	  after asterisk started.
	  This patch adds qualifying for the AOR's permanent contacts
	  on the first handling of this AOR.

	  ASTERISK-26319 #close

	  Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe

2016-08-22 17:08 +0000 [4c2fed8814]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Default endpoints to the "offline" status.

	  A recent change attempted to optimize startup by not updating contact
	  status. Instead, code responsible for qualifying contacts updates the
	  status as it becomes known. The code even accounts for contacts/AORs
	  that are not set to be qualified.

	  The problem, though, is when there are no contacts associated with an
	  endpoint. A common case is when an endpoint is set to register its
	  contacts but has not done so yet. In this case, prior to registration,
	  the endpoint's device state will appear to be "not in use" and hints
	  associated with that device will appear to be "idle". In actuality, the
	  device state and hint should both appear as "unavailable". The reason
	  for the failure is that the optimization change made all persistent
	  endpoint states set to "unknown".

	  The fix here is to change the hard-coded "unknown" to be "offline"
	  instead. The default state will be offline until the qualifying code
	  determines that the contact is actually online. This way, if there are
	  no contacts at all, then the state stays as offline, and device state
	  and hints appear correctly.

	  ASTERISK-26269 #close
	  Reported by nappsoft

	  Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a

2016-08-17 02:51 +0000 [a637e2c301]  chris de rock <chris@derock.de>

	* app_macro: Consider '~~s~~' as a macro start extension.

	  As described in issue ASTERISK-26282 the AEL parser creates macros with
	  extension '~~s~~'.  app_macro searches only for extension 's' so the
	  created extension cannot be found.  with this patch app_macro searches for
	  both extensions and performs the right extension.

	  ASTERISK-26282 #close

	  Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb

2016-08-29 07:07 +0000 [f9dc20b8cb]  Etienne Lessard <elessard@proformatique.com>

	* pbx.c: Prevent infinite recursion in manager_show_dialplan_helper.

	  Previously, if context A was including context B and context B was including
	  context A, i.e. if there was a circular dependency between contexts, then
	  calling manager_show_dialplan_helper could lead to an infinite recursion,
	  resulting in a crash.

	  This commit applies the same solution as the one implemented in the
	  show_dialplan_helper function. The manager_show_dialplan_helper and
	  show_dialplan_helper functions contain lots of code in common, but the former
	  was missing the "infinite recursion avoidance" code.

	  ASTERISK-26226 #close

	  Change-Id: I1aea85133c21787226f4f8442253a93000aa0897

2016-08-25 07:06 +0000 [f7b3f628cd]  Joshua Colp <jcolp@digium.com>

	* app_queue: Ensure member is removed from pending when hanging up.

	  When dialing channels it is possible that they may not ever
	  leave the not in use state (Local channels in particular) by
	  the time we cancel them. If this occurs but we know they were
	  dialed we explicitly remove them from the pending members
	  container so that subsequent call attempts occur.

	  ASTERISK-26299 #close

	  Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65

2016-08-26 14:34 +0000 [cfe2675dd4]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Disable srtp use by pjmedia

	  The reason for the disable is that while Asterisk works fine with older
	  libsrtp versions, newer versions of pjproject won't compile with them.
	  Debian 6 for instance, has libsrtp 1.4.4 which is older than what
	  pjproject is expecting.

	  We don't use most of pjmedia but we DO use it for SDP negotiation.
	  Luckily disabling srtp in pjmedia doesn't interfere with it's ability
	  to negitiate a secure channel.  The proper crypto attributes are
	  negotiated in both directions.

	  ASTERISK-26279 #close

	  Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2

2016-08-26 08:41 +0000 [1d2d4e2ae9]  Alexander Traud <pabstraud@compuserve.com>

	* channel: No hung-up on failing security requirements.

	  In your Diaplan, if you specify
	   same => n,Set(CHANNEL(secure_bridge_media)=1)
	   same => n,Set(CHANNEL(secure_bridge_signaling)=1)
	  only the SIP channel driver chan_sip supports this. All other channels drivers
	  like res_pjsip fail. In case of failure, the original sRTP source code released
	  the whole channel, even if not hung-up, yet. This change does not release the
	  channel but instead hangs-up the channel.

	  ASTERISK-26306

	  Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db

2016-08-20 09:04 +0000 [c3ac1651dd]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations.

	  When using the migration script sip_to_pjsip.py, and your sip.conf is
	  configured with bindaddr=::, two transports are written to pjsip.conf, one for
	  0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4
	  and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface
	  like in chan_sip.

	  Furthermore, the script internal functions "build_host" and "split_hostport"
	  did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change
	  makes sure, even such addresses are parsed correctly.

	  ASTERISK-26309

	  Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48

2016-08-04 20:11 +0000 [a8771d7a71]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Cache global config options.

	  We may check a global config option hundreds of times a second or more.
	  Asking sorcery for the global configuration from the config files backend
	  involves several allocations and container traversals.  Using realtime
	  without a memory cache is a lot worse because you have to lookup in the
	  realtime database each time to reconstitute the sorcery object.  With a
	  memory cache for realtime, there is about the same amount of overhead as
	  for config files.  Either way, it is still fairly expensive to access the
	  sorcery object that much.

	  * Cache the global config options so we can access them faster.  You must
	  now always perform a res_pjsip reload to change the global options.

	  Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7

2016-08-23 11:02 +0000 [1a7d5989d6]  Richard Mudgett <rmudgett@digium.com>

	* res_fax: Fix deadlock in ast_channel_get_t38_state().

	  ast_channel_get_t38_state() calls ast_channel_queryoption() with
	  AST_OPTION_T38_STATE.  If the passed in channel is a local channel then a
	  deadlock can happen if a channel lock is held when called.

	  * Made ast_channel_get_t38_state() callers not hold a channel lock before
	  calling.

	  * Update ast_channel_get_t38_state() doxygen to note that no channel locks
	  can be held when calling the function.

	  ASTERISK-26203 #close
	  Reported by: Etienne Lessard

	  ASTERISK-24822 #close
	  Reported by: David Brillert

	  ASTERISK-22732 #close
	  Reported by: Richard Mudgett

	  Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214

2016-08-23 10:39 +0000 [aaef3b7175]  Richard Mudgett <rmudgett@digium.com>

	* res_fax: Fix deadlock setting FAXMODE channel variable.

	  ASTERISK-25980 added the FAXMODE channel variable to res_fax.c.
	  Unfortunately, it also introduced a deadlock potential because
	  set_channel_variables() which sets FAXMODE can be called during a
	  masquerade.  The ast_channel_get_t38_state() which gets the value used to
	  set FAXMODE cannot be called with the channel locked.  As a result, local
	  channels can deadlock because of how they must acquire the locks necessary
	  to operate.

	  The intent of FAXMODE is for dialplan to know how a fax was transferred
	  after the fax completes.  However, the previous patch sets FAXMODE to the
	  channel's current T.38 state AFTER the fax has completed and where T.38
	  may have already disconnected.

	  * Set FAXMODE based upon T.38 negotiations exchanged either with the fax
	  applications or the fax framehooks.

	  ASTERISK-26203
	  Reported by: Etienne Lessard

	  ASTERISK-24822
	  Reported by: David Brillert

	  ASTERISK-22732
	  Reported by: Richard Mudgett

	  Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1

2016-08-22 12:31 +0000 [a53aebabbc]  Richard Mudgett <rmudgett@digium.com>

	* res_fax.c: Fix deadlock in fax_gateway_indicate_t38().

	  fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be
	  called with any channel locks already held.  A deadlock can happen if the
	  function is operating on a local channel.

	  * Made fax_gateway_indicate_t38() unlock the channel before calling
	  ast_indicate_data() since fax_gateway_indicate_t38() is always called with
	  the channel locked.

	  * Made fax_gateway_indicate_t38() return void since nothing cared about
	  its return value.

	  ASTERISK-26203
	  Reported by: Etienne Lessard

	  ASTERISK-24822
	  Reported by: David Brillert

	  ASTERISK-22732
	  Reported by: Richard Mudgett

	  Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407

2016-08-23 11:16 +0000 [a6448b01a2]  Richard Mudgett <rmudgett@digium.com>

	* res_fax.c: Add chan locked precondition comments.

	  Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7

2016-08-23 10:42 +0000 [5869bb22db]  Richard Mudgett <rmudgett@digium.com>

	* ast_framehook_detach() must be called with the channel locked.

	  The framehook container could become corrupted if the channel lock is not
	  held before calling.

	  Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584

2016-08-22 15:01 +0000 [fc859dfee9]  Richard Mudgett <rmudgett@digium.com>

	* ast_framehook_attach() must be called with the channel locked.

	  The framehook container could become corrupted if the channel lock is not
	  held before calling.

	  Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438

2016-08-24 14:42 +0000 [b729072431]  gtjoseph <gjoseph@digium.com>

	* res_rtp_multicast:  Fix SEGV in ast_multicast_rtp_create_options

	  ast_multicast_rtp_create_options now checks for NULL or empty options

	  Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362

2016-08-19 18:19 +0000 [a8b3d8d7f9]  Corey Farrell <git@cfware.com>

	* Fix checks for allocation debugging.

	  MALLOC_DEBUG should not be used to check if debugging is actually
	  enabled, __AST_DEBUG_MALLOC should be used instead.  MALLOC_DEBUG only
	  indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it
	  is active.

	  Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53

2016-08-10 15:14 +0000 [415f95ed00]  Mark Michelson <mmichelson@digium.com>

	* ConfBridge: Rework announcer channel methodology

	  NOTE: This patch was submitted earlier and reverted because of a failing
	  test. The test has been patched so that it adjusts for the changes here,
	  so this is being resubmitted for review.

	  One feature that confbridge has is the ability to play sounds to all
	  participants in the conference. Prior to this commit, the algorithm for
	  this was as follows:

	  * Grab the playback lock
	  * Push the conference announcer channel into the bridge
	  * Play back the sound
	  * Pull the conference announcer channel from the bridge
	  * Release the playback lock

	  The issue here is that the act of adding the playback channel to the
	  bridge and removing it for each announcement is expensive. Amongst the
	  expenses:

	  * The announcer channel is imparted into the bridge, meaning a new
	    thread is spun up for each playback.
	  * When the announcer is added or removed from the bridge, it results
	    in the BRIDGEPEER channel variable being set on all channels in the
	    bridge. This requires keeping the bridge locked and locking each
	    individual channel in order to set it.
	  * There's also just the general overhead of adding the channel and
	    removing it from the bridge. The bridge potentially has to reconfigure
	    every single time

	  With this commit, the paradigm for playing back announcements has
	  shifted.

	  * The announcer channel is now added to the bridge when the conference
	    is allocated, and it is hung up when the conference is destroyed.
	  * A taskprocessor is used to queue playbacks onto the announcer channel.
	    This keeps the behavior from before where playbacks do not overlap.
	  * The announcer channel is no longer placed into the bridge as
	    departable. Since we are not constantly removing the channel from
	    the bridge, it is safe to add the channel using an independent thread
	    and simply hang the channel up when it is time for the conference to
	    be destroyed.

	  The use of the taskprocessor for playbacks opens up the interesting
	  possibility of having asynchronous announcements played. In this commit,
	  however, the behavior is still exactly the same as it previously was.

	  ASTERISK-26289
	  Reported by Mark Michelson

	  Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0

2016-08-23 05:54 +0000 [937093f768]  Joshua Colp <jcolp@digium.com>

	* Revert "ConfBridge: Rework announcer channel methodology"

	  This reverts commit d9749887361252f7486bd5aaf3d3a8f046831a3d.

	  Change-Id: I517a38f4d821dbccc335f7cba0e675da5f8ed675

2016-08-19 10:21 +0000 [bba715ecad]  Alexei Gradinari <alex2grad@gmail.com>

	* compilation failed with -Werror=maybe-uninitialized

	  The compilation failed for devmode
	  --enable DONT_OPTIMIZE
	  --enable BETTER_BACKTRACES
	  --enable DO_CRASH
	  --enable TEST_FRAMEWORK

	  res_pjsip/pjsip_configuration.c: In function dtls_handler:
	  res_pjsip/pjsip_configuration.c:974:20: error:
	  back may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  int size = strlen(front);
	             ^
	  cc1: all warnings being treated as errors

	  Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580

2016-08-11 12:10 +0000 [6b6dea53b0]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* followme: initialize all config items on reload

	  Some configuration directives were not initialized on reload, and hence
	  were not reset to default if they were removed from followme.conf.

	  ASTERISK-26288 #close

	  Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150

2016-08-20 14:51 +0000 [9dd4ec7d03]  David M. Lee <dlee@respoke.io>

	* res_odbc_transaction: add dep on generic_odbc

	  When res_odbc_transaction depended on res_odbc, it got the generic_odbc
	  headers and libs implicitly. Now that it no longer depends on res_odbc,
	  its dependency on generic_odbc must be explicit.

	  Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911

2016-08-20 11:18 +0000 [dbfe4a9011]  Alexander Traud <pabstraud@compuserve.com>

	* pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations.

	  PJProject supports a lot of platforms even Windows, some with different defaults
	  when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS,
	  "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows.

	  Because of this, if configured with just an IPv6 address/transport, PJProject
	  listens to both IPv4 and IPv6. However, this is not supported by the PJProject
	  team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP,
	  incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only
	  server which accepts incoming connections on IPv4.

	  If you try to configure two transports, one with IPv4 and one with IPv6 on the
	  same interface, as expected by the PJProject team, the IPv4 transport is not
	  able to bind because the IPv6 transport listens to both already.

	  One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide.
	  Then, you are able to configure two transports, one for each IP version on the
	  same interface. That way, you get a server which works with IPv4 clients and
	  IPv6 clients at the same time over the same interface.

	  Here, this change sets this parameter directly within PJProject to match the
	  expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack
	  servers out of the box like in chan_sip. This change was accepted by the
	  PJProject team as <http://trac.pjsip.org/repos/changeset/5403> and is expected
	  to arrive in the next version, PJProject 2.6.0. Until then, this change is
	  incorporated in the bundled PJProject of Asterisk.

	  ASTERISK-26309

	  Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae

2016-08-17 08:10 +0000 [1466737dbc]  Torrey Searle <torrey@voxbone.com>

	* res_ari: Add http prefix to generated docs

	  updated the uri handler to include the url prefix of the http server
	  this enables res_ari to add it to the uris when generating docs

	  Change-Id: I279335a2625261a8492206c37219698f42591c2e
	  (cherry picked from commit 6f448f32fe9b7379e2630fab7b06205f901f2ded)

2016-08-19 03:59 +0000 [9ca84fa2e8]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Add cert_file.

	  When using the migration script sip_to_pjsip.py, cert_file was not migrated to
	  pjsip.conf. A previous change regarding this contained a copy/paste error.

	  ASTERISK-22374

	  Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b

2016-08-18 09:21 +0000 [d192cd125c]  Alexander Traud <pabstraud@compuserve.com>

	* sip.conf: tlsclientmethod is using sslv23 as default.

	  When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL
	  SSLv23_method. This was documented incorrectly in the file sip.conf.sample.

	  SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method
	  enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that
	  function should have been called 'secure_method' or 'automatic_method' back in
	  the 90s.

	  Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if
	  you face a server which has problems like not falling back to TLSv1.0
	  automatically.

	  ASTERISK-24425

	  Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3

2016-08-16 15:57 +0000 [d2bee6b535]  Jason Parker (license 4993)

	* res_format_attr_g729: Add annexb=no format parameter to SDPs

	  Historically, Asterisk has always specified annexb=no for the g729 format.
	  However, when using res_pjsip no format attribute was specified. This patch
	  makes it so the SDP now contains a format attribute line with annexb=no.

	  Note, that this means only g729a is negotiated. Even for pass through support.
	  According to rfc7261 the type of annex used (a or b) is dependent upon the
	  answerer. However, Asterisk being a back to back user agent makes this tricky
	  to support at this time, thus we only allow annex 'a' for now.

	  ASTERISK-26228 #close
	  patches:
	    res_format_attr_g729.c submitted by Jason Parker (license 4993)

	  Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0

2016-08-18 17:02 +0000 [03c7e5e1ea]  Kevin Harwell <kharwell@digium.com>

	* rest-api: Swagger scripts were not replacing format variable in file brief

	  Given resource paths did not have 'json' substituted in for the '{format}'. For
	  some auto generated documentation/comment strings it resulted in something like
	  the following:

	  "... REST handler for /api-docs/sounds.{format}"

	  This patch makes sure the resource api's path is properly substituted.

	  ASTERISK-25472 #close

	  Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23

2016-08-18 15:15 +0000 [85a1de9f72]  gtjoseph <gjoseph@digium.com>

	* res_odbc:  Correct the dependency relationship with res_odbc_transaction

	  The MODULEINFO dependencies between these 2 modules was reversed.
	  res_odbc should depend on res_odbc_transaction, not the other way
	  around.

	  ASTERISK-25984 #close

	  Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f

2016-08-18 12:04 +0000 [81e3b8f141]  Kevin Harwell <kharwell@digium.com>

	* sip_to_pjsip: Set correct tls transport method

	  A recent update had a copy/paste error where the unused variable 'val' was
	  being passed to the set_value function instead of the 'method' value itself.

	  This patch passes in the right variable.

	  ASTERISK-22374

	  Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06

2016-08-10 15:14 +0000 [d974988736]  Mark Michelson <mmichelson@digium.com>

	* ConfBridge: Rework announcer channel methodology

	  One feature that confbridge has is the ability to play sounds to all
	  participants in the conference. Prior to this commit, the algorithm for
	  this was as follows:

	  * Grab the playback lock
	  * Push the conference announcer channel into the bridge
	  * Play back the sound
	  * Pull the conference announcer channel from the bridge
	  * Release the playback lock

	  The issue here is that the act of adding the playback channel to the
	  bridge and removing it for each announcement is expensive. Amongst the
	  expenses:

	  * The announcer channel is imparted into the bridge, meaning a new
	    thread is spun up for each playback.
	  * When the announcer is added or removed from the bridge, it results
	    in the BRIDGEPEER channel variable being set on all channels in the
	    bridge. This requires keeping the bridge locked and locking each
	    individual channel in order to set it.
	  * There's also just the general overhead of adding the channel and
	    removing it from the bridge. The bridge potentially has to reconfigure
	    every single time

	  With this commit, the paradigm for playing back announcements has
	  shifted.

	  * The announcer channel is now added to the bridge when the conference
	    is allocated, and it is hung up when the conference is destroyed.
	  * A taskprocessor is used to queue playbacks onto the announcer channel.
	    This keeps the behavior from before where playbacks do not overlap.
	  * The announcer channel is no longer placed into the bridge as
	    departable. Since we are not constantly removing the channel from
	    the bridge, it is safe to add the channel using an independent thread
	    and simply hang the channel up when it is time for the conference to
	    be destroyed.

	  The use of the taskprocessor for playbacks opens up the interesting
	  possibility of having asynchronous announcements played. In this commit,
	  however, the behavior is still exactly the same as it previously was.

	  ASTERISK-26289
	  Reported by Mark Michelson

	  Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5

2016-08-18 08:19 +0000 [94d1076e69]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Map the TLS method correctly.

	  When using the migration script sip_to_pjsip.py and tlsclientmethod is not set
	  in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to
	  overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is
	  offering/using not just TLSv1.0 but TLSv1.2 as well.

	  ASTERISK-22374

	  Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f

2016-08-18 08:17 +0000 [9f86d27f60]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent.

	  When using the migration script sip_to_pjsip.py, no section of type=system or
	  type=general were created. Therefore the keys compactheaders, timerb, timert1,
	  and useragent were not migrated to pjsip.conf.

	  ASTERISK-22374

	  Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1

2016-08-18 08:16 +0000 [019eb0454a]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Map (session-)timers correctly.

	  When using the migration script sip_to_pjsip.py, session-timers=accept and
	  session-timers=refuse were mapped to wrong values.

	  ASTERISK-22374

	  Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092

2016-08-18 08:15 +0000 [e054acc723]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Write username even without authname.

	  When using the migration script sip_to_pjsip.py, now the (mandatory) username is
	  written to pjsip.conf, even if there was no (optional) authname in the register
	  string in sip.conf.

	  ASTERISK-22374

	  Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f

2016-08-18 08:14 +0000 [d0c1a4fd33]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Parse register even with transport.

	  When using the migration script sip_to_pjsip.py and the register string
	  started with a transport in sip.conf - like tls://... - register was not parsed
	  correctly and therefore not migrated correctly to pjsip.conf.

	  ASTERISK-22374

	  Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2

2016-08-18 08:13 +0000 [1a556c7618]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit.

	  When using the migration script sip_to_pjsip.py, those keys got missing. These
	  keys might appear several times and the function "merge_value" tried to collect
	  those. However, because these keys have different names in sip.conf and
	  pjsip.conf, "merge_value" was not able to find the new key name in sip.conf.
	  This change lets "merge_value" search with the old key name in sip.conf and
	  write with the new key name in pjsip.conf.

	  ASTERISK-22374

	  Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2

2016-08-18 08:11 +0000 [32fa73d108]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Map externhost/ip to Transports.

	  When using the migration script sip_to_pjsip.py, the externhost or externip of
	  sip.conf were erroneously written to Endpoints instead to Transports.

	  ASTERISK-22374

	  Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4

2016-08-18 08:04 +0000 [db1715a7e7]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry.

	  When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and
	  minexpiry were not migrated to pjsip.conf.

	  ASTERISK-22374

	  Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b

2016-08-18 08:03 +0000 [527ac18ce7]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Write media_encryption.

	  When using the migration script sip_to_pjsip.py, encryption=yes got missing and
	  media_encryption=sdes was not written to pjsip.conf, because of a typo.

	  ASTERISK-22374

	  Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05

2016-08-18 08:02 +0000 [6078499b43]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Write cos and tos.

	  When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got
	  missed, because of a typo. Therefore, cos and tos were not written to
	  pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused
	  by a copy-and-paste error.

	  ASTERISK-22374

	  Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2

2016-08-18 07:55 +0000 [8c63306001]  Alexander Traud <pabstraud@compuserve.com>

	* sip_to_pjsip: Add cert_file and ca_list_path.

	  When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were
	  not migrated to pjsip.conf.

	  ASTERISK-22374

	  Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825

2016-08-16 15:36 +0000 [f87af89396]  gtjoseph <gjoseph@digium.com>

	* res_pjsip:  Add contact_user to endpoint

	  contact_user, when specified on an endpoint, will override the user
	  portion of the Contact header on outgoing requests.

	  Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4

2016-08-17 14:13 +0000 [d833ef6523]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Fix unbound srv failover tests.

	  Commit 1b666549f33d69dc080b212bf92126f3bc3a18b2 broke the srv failover
	  functionality if a TCP connection gets disconnected.  Under these
	  conditions, session_inv_on_state_changed() gets a
	  PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new
	  transport.  Unfortunately, session_inv_on_tsx_state_changed() also gets
	  the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates
	  the session.

	  * Made session_inv_on_tsx_state_changed() complete terminating the session
	  on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still
	  PJSIP_INV_STATE_DISCONNECTED.

	  ASTERISK-26305 #close
	  Reported by: Richard Mudgett

	  Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d

2016-08-17 06:12 +0000 [41e9bf60aa]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Detect ca_list_path capabilities in external PJProject.

	  Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the
	  bundled PJProject but not in an external PJProject. Therefore, ca_list_path
	  could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2
	  is detected again to enable ca_list_path again.

	  ASTERISK-26303 #close

	  Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0

2016-08-16 12:24 +0000 [d938c18733]  gtjoseph <gjoseph@digium.com>

	* ari:  Add documentation that path parameters are case-sensitive

	  Added to api.wiki.mustache so that the generated object pages
	  have the notation in the table header as well as under each method
	  that has path parameters.

	  ASTERISK-25492 #close

	  Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf

2016-08-15 15:29 +0000 [ba91710bc4]  Corey Farrell <git@cfware.com>

	* Refactor usage pattern of xmldoc info tag.

	  This updates func_channel.c and main/message.c to use a generic xpointer
	  include instead of including info from each channel driver.  Now the
	  name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
	  documentation for func_channel.  Setting the name attribute of info to
	  MessageToInfo or MessageFromInfo causes it to be included in the
	  MessageSend application and AMI action.

	  Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea

2016-06-15 17:10 +0000 [19e323a6aa]  Evgeniy Tsybra <cjack@yandex.ru>

	* chan_sip: Fix lastrtprx always updated

	  Packets are read regulary, when there is no data in buffer fr->frametype
	  is AST_FRAME_NULL. There was no check of frametype and lastrtprx always 
	  updated and, therefore, rtptimeout did not work at all.

	  ASTERISK-25270 #close

	  Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d

2016-08-10 14:41 +0000 [811e7d7788]  Alexei Gradinari <alex2grad@gmail.com>

	* core: Entity ID is not set or invalid

	  The Exchanging Device and Mailbox States could not working
	  if the Entity ID (EID) is not set manually and can't be obtained
	  from ethernet interface.

	  This patch replaces debug message to warning
	  and addes missing description about option 'entityid' to
	  asterisk.conf.sample.

	  With this patch the asterisk also:
	  (1) decline loading the modules which won't work without EID:
	      res_corosync and res_pjsip_publish_asterisk.
	  (2) warn if EID is empty on loading next modules:
	      pbx_dundi, res_xmpp

	  Starting with v197 systemd/udev will automatically assign "predictable"
	  names for all local Ethernet interfaces.
	  This patch also addes some new ethernet prefixes "eno" and "ens".

	  ASTERISK-26164 #close

	  Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6

2016-08-04 20:00 +0000 [840d274f4e]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_config.c: Cleanup ao2 container usage idioms.

	  Change-Id: Iad24b335fb121a2bc7f1d048ab7420569edcba5a

2016-08-04 15:57 +0000 [2f8f063d0a]  Richard Mudgett <rmudgett@digium.com>

	* sorcery.c: Minor optimizations.

	  * Remove some unused parameters from internal functions:
	  sorcery_wizard_create()
	  sorcery_wizard_update()
	  sorcery_wizard_delete()

	  * Created the struct sorcery_observer_invocation ao2 object without a lock
	  since it is not needed in sorcery_observer_invocation_alloc().

	  * Cleanup generic ao2 container sorcery object id hash, sort, and cmp
	  functions.

	  Change-Id: Iff71d75f52bc1b8cee955456838c149faaa4f92e

2016-08-01 11:04 +0000 [aed60171f1]  Richard Mudgett <rmudgett@digium.com>

	* sorcery.c: Tweak some container declaration formatting.

	  * Tweak sorcery_object_type_alloc() formatting.
	  * Tweak ast_sorcery_init() formatting.

	  Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944

2016-08-11 23:30 +0000 [f7f9081f7a]  Corey Farrell <git@cfware.com>

	* pbx.c: Additional fixes to ast_context_remove_extension_callerid2.

	  Do not check registrar of the first extension head.  We should only check
	  the registrar when we match the priority.

	  Additionally fix a couple calls to strcmp which used the input callerid
	  instead of the clean version ex.cidmatch.

	  ASTERISK-26233

	  Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1

2016-08-13 22:02 +0000 [7c46e3bfbb]  Matt Jordan <mjordan@digium.com>

	* app_dial: Improve documentation

	  * Add some helpful <literal> and other embedded paragraph tags

	  * Document some of the lesser known channel variables set by Dial

	  * Add examples for some common Dial uses, along with some more
	    challenging but useful options

	  Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1

2016-08-13 20:16 +0000 [2e77b973fb]  Matt Jordan <mjordan@digium.com>

	* manager: Add <see-also> tags to relate interrelated events/actions together

	  Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4

2016-08-13 20:15 +0000 [18065140d5]  Matt Jordan <mjordan@digium.com>

	* manager: Add <see-also> tags to relate Bridge related events,actions, and apps

	  Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85

2016-08-13 20:14 +0000 [f69ae95889]  Matt Jordan <mjordan@digium.com>

	* manager: Add <see-also> tags to relate AoC events and actions

	  Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e

2016-08-13 20:13 +0000 [c5574667c7]  Matt Jordan <mjordan@digium.com>

	* manager: Add <see-also> tags to relate UserEvent actions/apps/events

	  Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4

2016-08-12 15:53 +0000 [1a44ebe8ec]  Matt Jordan <mjordan@digium.com>

	* res_agi: Improve documentation

	  * Groups of AGI commands that have similar functionality now reference
	    each other, and all reference the AGI application for ease of wiki
	    reference.

	  * The documentation for the AGI application has been improved, in
	    particular noting the various AGI types and how they are invoked.

	  * A warning message has been added to DeadAGI, noting that it is
	    deprecated.

	  Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d

2016-08-12 13:53 +0000 [3b3198333a]  Matt Jordan <mjordan@digium.com>

	* manager: Add <see-also> links between related events

	  This patch adds some see-also references between related AMI events. It
	  focuses primarily on those events that are guaranteed to come in pairs,
	  such as DTMFBegin/DTMFEnd, as well as those that occur during the life
	  cycle of an Asterisk channel, such as Newchannel/Hangup.

	  Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3

2016-08-12 11:15 +0000 [89dccb04bc]  Matt Jordan <mjordan@digium.com>

	* func_channel: Reorganize documentation

	  * Following the example of the PJSIP channel driver, the channel
	    technology specific documentation has been moved to the respective
	    channel drivers that provide that functionality. This has the benefit
	    of locating the documentation of items with those modules that provide
	    it.

	  * Examples of using the CHANNEL function for both standard items as well
	    as for PJSIP have been added.

	  * The 'max_forwards' standard item has been documented.

	  Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b

2016-08-15 07:17 +0000 [2ce487585e]  Joshua Colp <jcolp@digium.com>

	* manager: Clarify that dialplan manipulation actions are under system class.

	  ASTERISK-26246 #close

	  Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657

2016-08-11 22:12 +0000 [03b80c5d27]  Corey Farrell <git@cfware.com>

	* Run mandatory cleanup when startup fails.

	  Errors during startup result in an exit.  These error branches should be
	  calling ast_run_atexit(0) to ensure mandatory cleanup is run.

	  ASTERISK-26267 #close

	  Change-Id: If226f2326ae2df7add20040696132214cf2bb680

2016-08-11 11:24 +0000 [33952469f2]  gtjoseph <gjoseph@digium.com>

	* res_pjsip_caller_id:  Copy header name to short header name

	  When compact_headers was set, we were sending a zero-length header name
	  for PAI and RPID because we always forced the short header name length
	  to 0.  We did this because we cloned the header from "From" and wanted
	  to clear "f" from the sname.  By cloning however, we bypass pjproject's
	  automatic logic that sets sname to name if there's no compact form of
	  the header, which there isn't for PAI and RPID.  So now we force sname
	  to be the same as name right after we set name.

	  res_pjsip_diversion needed the same treatment for the Diversion header.

	  ASTERISK-26241 #close

	  Change-Id: I633ec139630cd83809aae00336cee4a10077e467

2016-08-11 11:13 +0000 [82529ae4e6]  Matt Jordan <mjordan@digium.com>

	* app_queue: Prevent crash when a call is forwarded to an invalid location

	  When a call forward attempt is made from a Queue member, the current
	  code will hang up the forwarding channel in an off-nominal condition
	  prior to raising the Stasis events informing the rest of Asterisk that
	  the call was forwarded. This will result in a slew of dreaded FRACKs,
	  most likely leading to a crash.

	  This patch modifies the code such that we don't hang up the forwarding
	  channel even in an off-nominal condition until we've safely raised the
	  Stasis messages.

	  ASTERISK-25797 #close

	  Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38

2016-08-11 12:18 +0000 [2db2d9dbd3]  gtjoseph <gjoseph@digium.com>

	* res_pjsip:  Fail global load if debug or default_from_user are empty

	  If debug was specified in the global configuration but left blank,
	  the logger would treat it as a wildcard and log all hosts.  If
	  default_from_user was empty, a crash would result.

	  The global apply handler now checks for empty strings.

	  ASTERISK-26239 #close
	  ASTERISK-26238 #close

	  Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336

2016-08-01 15:07 +0000 [63863ed0f3]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip res_pjsip_mwi: Misc fixes and cleanups.

	  * Eliminated RAII_VAR() usage in
	  ast_sip_persistent_endpoint_update_state().

	  * Added a missing allocation failure check to
	  persistent_endpoint_find_or_create().

	  * Made persistent_endpoint_find_or_create() create the new object without
	  a lock as it isn't needed.

	  * Cleaned up some ao2 container allocation idioms.

	  * Reordered res_pjsip_mwi.c load_module() and unload_module()

	  Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8

2016-08-04 18:03 +0000 [ab45dfaeb9]  Richard Mudgett <rmudgett@digium.com>

	* location.c: Misc fixes and cleanups.

	  * Eliminated most RAII_VAR() usage.

	  * Added several missing allocation failure checks.

	  * Made ast_sip_for_each_contact() allocate the wrapper ao2 object without
	  a lock as it is not needed.

	  Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc

2016-08-11 12:01 +0000 [96887cc110]  gtjoseph <gjoseph@digium.com>

	* autohints:  Update CHANGES and extensions.conf.sample

	  Make it clear that we're talking about device state hints and add
	  an entry to the sample config.

	  Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433

2016-08-02 13:53 +0000 [7f9789fc39]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Tweak high water checks.

	  * The high water check in ast_taskprocessor_alert_set_levels() would
	  trigger immediately if the new high water level is zero and the queue was
	  empty.

	  * The high water check in taskprocessor_push() was off by one.

	  Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d

2016-08-03 16:24 +0000 [94ff684388]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Make aor named lock a mutex.

	  The named aor lock was always being locked for writes so a rwlock adds no
	  benefit and may be slower because rwlocks are biased toward read locking.

	  Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28

2016-07-29 17:41 +0000 [31a1b066ee]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_distributor.c: Add missing allocation failure check.

	  Change-Id: I932ab2cea845e534d9ff318035b6de39972d3b28

2016-08-11 10:50 +0000 [112de90576]  Kevin Harwell <kharwell@digium.com>

	* alembic: add auth_username to endpoint's identify_by enum

	  A new identify_by option was added recently, auth_username. However, this
	  setting was not added as an allowable choice in the database enumeration
	  value.

	  This patch updates the current enumeration, adding in the new setting.

	  ASTERISK-26268 #close

	  Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8

2016-08-11 10:50 +0000 [919824ebbc]  David M. Lee <dlee@respoke.io>

	* Fixed compile flags for non-module libs

	  The non-module libs libasteriskssl.dylib and libasteriskpj.dylib have
	  long been missing the AST_NOT_MODULE compile flag. This was mostly
	  okay, until a recent fix to improve compiler warnings when the
	  AST_MODULE_SELF_SYM is missing broke the build on OS X/macOS/whatever
	  they are calling it these days.

	  Change-Id: I2cb51c890824f001280a5114f2e775f97c163516

2016-08-08 14:50 +0000 [34d366eb0c]  Richard Mudgett <rmudgett@digium.com>

	* res_srtp: Move SDP SRTP code from the core to res_srtp.

	  A patch made to the master branch (Now the 14 branch) inadvertently made
	  libsrtp a required dependency in order to compile Asterisk.  Rather than
	  create dummy defines to substitute for the defines supplied by libsrtp
	  when libsrtp is not available, most of the code in sdp_srtp.c is moved
	  into res_srtp.c.  This gets more code out of Asterisk's core that isn't
	  used when SRTP is not available.  This also makes another inadvertent
	  required dependency on libsrtp by Asterisk's core unlikely.

	  ASTERISK-26253 #close
	  Reported by: Ben Merrills

	  Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7

2016-08-06 10:57 +0000 [ecad35e147]  Alexei Gradinari <alex2grad@gmail.com>

	* pjsip: Fix deadlock with suspend taskprocessor on masquerade

	  If both channels which should be masqueraded
	  are in the same serializer:
	  1st channel will be locked waiting condition 'complete'
	  2nd channel will be locked waiting condition 'suspended'

	  On heavy load system a chance that both channels will be in
	  the same serializer 'pjsip/distibutor' is very high.

	  To reproduce compile res_pjsip/pjsip_distributor.c with
	  DISTRIBUTOR_POOL_SIZE=1

	  Steps to reproduce:
	  1. Party A calls Party B (bridged call 'AB')
	  2. Party B places Party A on hold
	  3. Party B calls Voicemail app (non-bridged call 'BV')
	  4. Party B attended transfers Party A to voicemail using REFER.
	  5. When asterisk masquerades calls 'AB' and 'BV',
	     a deadlock is happened.

	  This patch adds a suspension indicator to the taskprocessor.
	  When a session suspends/unsuspends the serializer
	  it sets the indicator to the appropriate state.
	  The session checks the suspension indicator before
	  suspend the serializer.

	  ASTERISK-26145 #close

	  Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b

2016-08-09 12:07 +0000 [223b7756a6]  Kevin Harwell <kharwell@digium.com>

	* alembic/sqlalchemy: auto increment only allowed on a single column

	  The extensions table defined two columns (id and priority) as primary key
	  autoincrement columns. However only one is allowed when defining the primary
	  key.

	  This patch removes the autoincrement attribute from the priority column since
	  it does not need to be as such and really should not have been on there in the
	  first place.

	  This patch also removes 'context', 'exten', and 'priority' from the primary key
	  index and creates a new combined unique contraint index on them.

	  ASTERISK-26183 #close

	  Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b

2016-08-10 11:47 +0000 [a8512f9afe]  gtjoseph <gjoseph@digium.com>

	* res_resolver_unbound:  Allow compilation with libunbound version < 1.5

	  libunbound at version 1.4.20 (which CentOS still uses) declared all
	  of their string function parameters as as 'char *'.  1.4.21 changed
	  them all to 'const char *'.  Thankfully 1.4.21 also introduced the
	  UNBOUND_VERSION_MAJOR define so configure now checks for that and
	  sets HAVE_UNBOUND_CONST_PARAMS.  res_resolver_unbound then checks
	  that and casts away the 'const' if it's not set.

	  Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and
	  Fedora24 (1.5.4).  There are a few failing tests to be addressed though.

	  ASTERISK-26283 #close

	  Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148

2016-08-07 09:58 +0000 [eec60dd773]  Matt Jordan <mjordan@digium.com>

	* channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH

	  This patch adds a new PJSIP specific dialplan function,
	  PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media
	  session will be refreshed via either an UPDATE or re-INVITE request.
	  When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function,
	  the formats in use on a PJSIP channel can be re-negotiated and changed
	  dynamically after call setup.

	  ASTERISK-26277 #close

	  Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b

2016-08-05 22:06 +0000 [9ce71b6f9e]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* cdr_adaptive_odbc: Fix DNSs mixed config quote quoted_identifiers

	  When haved more than once DNSs config and one of their dont set
	  quoted_identifiers and before this is with configurated with
	  quoted_identifiers resulting a truncate statement for a reference null
	  for quote character identifier.

	  This patch initializes quoted flag before build SQL Query

	  Example config for this bugfix case in cdr_adaptive_odbc.conf file

	  	[first]
	  	connection=asterisk-server1
	  	table=cdr
	  	quoted_identifiers="

	  	[second]
	  	connection=asterisk-server2
	  	table=cdr

	  	[third]
	  	connection=asterisk-server3
	  	table=cdr
	  	quoted_identifiers=`

	  Change-Id: Ibd95667b468e10d4a19a2b9d88b9934ec7207e1d

2016-08-06 01:37 +0000 [c2652d05d9]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* res_odbc: Show only when there a fail attempt of connection in CLI

	  When is executed CLI command "odbc show all" every time is show
	  information about variable last_negative_connect. If not there  a fail
	  attempt of connection will show date like "1969-12-31 21:00:00".

	  This patch fix there situation for to show only this information when
	  exists a fail attempt before.

	  Change-Id: I7c058b0be6f7642e922de75ee6b82c7276c9f113

2016-08-09 16:19 +0000 [5c203c6ca3]  Mark Michelson <mmichelson@digium.com>

	* res_rtp_asterisk: Cache local RTCP address.

	  When an RTCP packet is sent or received, res_rtp_asterisk generates a
	  Stasis event that contains the RTCP report as well as the local and
	  remote addresses that the report pertains to.

	  The addresses are determined using ast_find_ourip(). For the local
	  address, this will typically result in a lookup of the hostname of the
	  server, and then a DNS lookup of that hostname. If you do not have the
	  host in /etc/hosts, then this results in a full DNS lookup, which can
	  potentially block for some time.

	  This is especially problematic when performing RTCP reads, since those
	  are done on the same thread responsible for reading and writing media.

	  This patch addresses the issue by performing a lookup of the local
	  address when RTCP is allocated. We then use this cached local address
	  for the Stasis events when necessary.

	  ASTERISK-26280 #close
	  Reported by Mark Michelson

	  Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556

2016-08-08 19:14 +0000 [bbaebd75c4]  Corey Farrell <git@cfware.com>

	* Produce friendly error when AST_MODULE_SELF_SYM is not defined.

	  Modules must define AST_MODULE_SELF_SYM to be used as the name of a
	  generated function.  This produces a friendly error when it's not
	  defined.

	  ASTERISK-26278 #close

	  Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b

2016-08-08 12:53 +0000 [4d6d3eb268]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack

	  The PJSIP taskprocessors could be overflowed on startup
	  if there are many (thousands) realtime endpoints
	  configured with unsolicited mwi.
	  The PJSIP stack could be totally unresponsive for a few minutes
	  after boot completed.

	  This patch creates a separate PJSIP serializers pool for mwi
	  and makes unsolicited mwi use serializers from this pool.
	  This patch also adds 2 new global options to tune taskprocessor
	  alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.

	  This patch also adds new global option 'mwi_disable_initial_unsolicited'
	  to disable sending unsolicited mwi to all endpoints on startup.
	  If disabled then unsolicited mwi will start processing
	  on next endpoint's contact update.

	  ASTERISK-26230 #close

	  Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a

2016-08-05 15:26 +0000 [5d74c0ea30]  Alexei Gradinari <alex2grad@gmail.com>

	* app_voicemail: Add taskprocessor alert level options.

	  On heavy loaded system with IMAP or DB storage,
	  'app_voicemail' taskprocessor queue could reach 500 scheduled tasks.
	  It could happen when the IMAP or DB server dies or is unreachable.
	  It could happen on startup when there are many (thousands)
	  realtime endpoints configured with unsolicited mwi.
	  If the taskprocessor queue reaches the high water level
	  then the alert is triggered and pjsip stops processing new requests
	  until the queue reaches the low water level to clear the alert.

	  This patch adds 2 new 'general' configuration options
	  to tune taskprocessor alert levels:
	  'tps_queue_high' - Taskprocessor high water alert trigger level.
	  'tps_queue_low' - Taskprocessor low water clear alert level

	  ASTERISK-26229 #close

	  Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8

2016-08-04 10:16 +0000 [a9592043dc]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_outbound_publish: Use a serializer shutdown group for unload.

	  This change replaces the custom unload process for the outbound
	  publish module with the common serializer shutdown group.

	  ASTERISK-25217 #close

	  Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6

2016-08-04 10:27 +0000 [93aa3f74a2]  Kevin Harwell <kharwell@digium.com>

	* resource_channels: Sync with ARI stubs

	  This file was out of sync with the current ARI definitions.

	  Change-Id: Ie7cb7d6d3c2eeb9cc9d683ca87b43b117e713d0a

2016-08-03 15:41 +0000 [8e6a28e86b]  Corey Farrell <git@cfware.com>

	* Add missing checks during startup.

	  This ensures startup is canceled due to allocation failures from the
	  following initializations.
	  * channel.c: ast_channels_init
	  * config_options.c: aco_init

	  ASTERISK-26265 #close

	  Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611

2016-08-03 09:47 +0000 [d1f973906a]  Joshua Colp <jcolp@digium.com>

	* astconfigparser: Really handle case where line is simply a comment.

	  The regular expression would match causing the code that handled
	  the line if it was merely a comment to never get executed.

	  Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819

2016-08-02 14:26 +0000 [9b926ac4f0]  Kevin Harwell <kharwell@digium.com>

	* make_ari_stubs: Updated WIKI_PREFIX for 14

	  Change-Id: Id5678d20d8892fdcde7849c810d492527e77c15d

2016-08-02 12:55 +0000 [63dd0fab30]  Kevin Harwell <kharwell@digium.com>

	* rest-api: Code out of sync with the model

	  Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006

2016-08-01 11:08 +0000 [b63262e7bf]  Joshua Colp <jcolp@digium.com>

	* sorcery: Use more compatible regex for local expressions.

	  This changes the use of an empty regex for both res_sorcery_config
	  and res_sorcery_memory to "." instead. This is a more compatible
	  regular expression which also works on FreeBSD.

	  ASTERISK-26206 #close

	  Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388

2016-08-02 03:08 +0000 [7e0178925f]  Alexander Traud <pabstraud@compuserve.com>

	* res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports.

	  ASTERISK-26256 #close

	  Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058

2016-08-01 16:13 +0000 [eefd446b9f]  gtjoseph <gjoseph@digium.com>

	* menuselect:  Add an opaque "member_data" string to the acceptable xml

	  Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe

2016-07-29 13:13 +0000 [d50895c7b0]  Mark Michelson <mmichelson@digium.com>

	* Remove SILK payload mappings from Asterisk core.

	  SILK is a bit of a hog when it comes to using up our limited number of
	  dynamic payload types in the RTP engine. By freeing up four slots, it
	  allows for other codecs to potentially take the place.

	  Now, codec_silk.so will dynamically use the payload slots in the RTP
	  engine when it loads.

	  A better fix would be make RTP dynamic payload types actually
	  dynamic. However, at this stage of Asterisk 14 development, this is a
	  risky move that would be imprudent.

	  Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612

2016-07-29 04:48 +0000 [6b5058acd8]  Joshua Colp <jcolp@digium.com>

	* astconfigparser: Handle case where line is simply a comment.

	  Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5

2016-07-28 14:10 +0000 [5597193f62]  Corey Farrell <git@cfware.com>

	* pbx.c: Fix handling of '-' in extension name and callerid

	  This adds a two strings to ast_exten.  name to go with exten and
	  cidmatch_display to go with cidmatch.  The new fields contain input used
	  to add the extension in the first place.  The existing fields now
	  contain stripped input that excludes insignificant spaces and dashes.
	  These stripped fields should always be used for comparisons.  The
	  unstripped fields should normally be used for display, but displaying
	  stripped values will not cause runtime errors.

	  Note the actual string is only stored twice if it contains dashes.  If
	  no dashes are found then both 'char *' fields point to the same memory.
	  So this change has a minimum effect on memory usage.

	  The existing functions ast_get_extension_name and
	  ast_get_extension_cidmatch return unstripped values as they did before
	  this change.  Other similar bugs likely still exist where unstripped
	  extensions are saved outside pbx.c then passed back in.

	  ASTERISK-26233 #close

	  Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f

2016-07-27 17:17 +0000 [5316f49a2d]  Richard Mudgett <rmudgett@digium.com>

	* pbx.c: Allow dangerous functions when adding a hint to dialplan.

	  We can allow dangerous functions when adding a hint since altering
	  dialplan is itself a privileged activity.  Otherwise, we could never
	  execute dangerous functions.

	  ASTERISK-25996 #close
	  Reported by: Andrew Nagy

	  Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba

2016-07-21 10:36 +0000 [fc24faf1d9]  Alexei Gradinari <alex2grad@gmail.com>

	* pjproject: fixed a few bugs

	  This patch fixes the issue in pjsip_tx_data_dec_ref()
	  when tx_data_destroy can be called more than once,
	  and checks if invalid value (e.g. NULL) is passed to.

	  This patch updates array limit checks and docs
	  in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability().

	  Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40

2016-07-17 18:28 +0000 [4cf02b5584]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Update for pjproject 2.5.5

	  Add more --disable-* switches to Makefile.rules including
	  --disable-opus which was causing bundled pjproject to fail with
	  "undefined reference" errors in libasteriskpj.

	  Changed PJ_ENABLE_EXTRA_CHECK to 1.

	  Removed 2 obsolete patches and added a new one.
	  The new one was merged by Teluu on 6/27/2016.

	  ASTERISK-26148 #close

	  Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063

2016-07-27 10:33 +0000 [34087b0d43]  David M. Lee <dlee@respoke.io>

	* Portably sscanf tv_usec

	  In a timeval, tv_usec is defined as a suseconds_t, which could be
	  different underlying types on different platforms. Instead of trying to
	  scanf directly into the timeval, scanf into a long int, then copy that
	  into the timeval.

	  Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95

2016-07-27 12:36 +0000 [e5529c4493]  Kevin Harwell <kharwell@digium.com>

	* rtp_engine: Failed assertion and wrong name given for codec

	  Fixed an assert check that would trigger when the passed in value was negative.
	  The negative value was being cast to an unsigned value. This resulted in the
	  check failing.

	  Also fixed another problem when loading formats in the engine. When setting the
	  mime type the format's name was being passed in instead of the codec's name.

	  Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c

2016-07-21 22:44 +0000 [5f54805e62]  Richard Mudgett <rmudgett@digium.com>

	* dsp.c: Add fax and DTMF detection unit tests.

	  * Add fax amplitude and frequency sweep tests.
	  * Add DTMF amplitude and twist unit tests.

	  Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7

2016-07-21 11:56 +0000 [ebc2b7695d]  Richard Mudgett <rmudgett@digium.com>

	* dsp.c: Added descriptive comments to Goertzel calculations.

	  * Added doxygen to describe some struct members and what is going on in
	  the code.

	  Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d

2016-07-13 13:48 +0000 [a7e747918d]  Richard Mudgett <rmudgett@digium.com>

	* dsp.c: Fix incorrect format reference typo.

	  Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896

2016-07-25 21:18 +0000 [53b46428d1]  Richard Mudgett <rmudgett@digium.com>

	* dsp.c: Correct DTMF twist dsp.conf documentation.

	  Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae

2016-07-27 09:56 +0000 [47689998a9]  David M. Lee <dlee@respoke.io>

	* Replace strdupa with more portable ast_strdupa

	  The strdupa function is a GNU extension, and not widely portable. We
	  have an ast_strdupa function used within Asterisk which is preferred.
	  I pulled the definition up from menuselect.c into the menuselect.h
	  header file so it can be shared across menuselect.

	  Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e

2016-07-22 04:43 +0000 [8ea9fd7fa4]  Joshua Colp <jcolp@digium.com>

	* astconfigparser.py: Update with realtime fixes.

	  When configuring SIP URIs in the pjsip.conf file it is
	  necessary to escape the semicolon so the parser does not
	  treat it as a comment. This change allows this to work in
	  the astconfigparser implementation.

	  A secondary bug where some data was lost if a configuration
	  option included a "=" in its value was also fixed.

	  A bug where sections would be considered equal despite
	  being different has also been fixed.

	  Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8

2016-07-21 22:28 +0000 [ba449e0242]  Richard Mudgett <rmudgett@digium.com>

	* dsp.c: Fix erroneous fax tone detection.

	  The Goertzel calculations get less accurate the lower the signal level
	  being worked with becomes because there is less resolution remaining.
	  If it is too low we can erroneously detect a tone where none really
	  exists.  The searched for fax frequencies not only need to be so much
	  stronger than the background noise they must also be a minimum strength.

	  * Add needed minimum threshold test to tone_detect().

	  * Set TONE_THRESHOLD to allow low volume frequency spread detection.

	  ASTERISK-26237 #close
	  Reported by: Richard Mudgett

	  Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc

2016-09-19 14:18 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 14.0.0-rc1 Released.

2016-09-19 09:17 +0000 [a23b33576f]  Joshua Colp <jcolp@digium.com>

	* Release summaries: Add summaries for 14.0.0-rc1

2016-09-19 09:17 +0000 [e11354b864]  Joshua Colp <jcolp@digium.com>

	* Release summaries: Remove previous versions

2016-09-19 09:17 +0000 [24fac2271a]  Joshua Colp <jcolp@digium.com>

	* .version: Update for 14.0.0-rc1

2016-09-19 09:17 +0000 [52c101d441]  Joshua Colp <jcolp@digium.com>

	* .lastclean: Update for 14.0.0-rc1

2016-09-19 09:17 +0000 [edae56dc65]  Joshua Colp <jcolp@digium.com>

	* realtime: Add database scripts for 14.0.0-rc1

2016-09-14 09:51 +0000 [205e2ea351]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_transport_management: Convert time in log message to seconds.

	  ASTERISK-26375 #close

	  Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc

2016-09-13 06:08 +0000 [bc085bba24]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Don't assume a request will have any addresses.

	  When performing DNS resolution the failover code present in
	  res_pjsip currently assumes that a request will always have
	  at least one viable address. In practice this is not true.
	  A domain may be used that has no records.

	  The code now checks that at least one address exists on the
	  request which prevents looping.

	  ASTERISK-26364 #close

	  Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c

2016-09-09 05:39 +0000 [9a800b24ac]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Only invoke unidentified endpoint logic when unidentified.

	  The code was incorrectly invoking the unidentified logic when
	  an endpoint had actually been identified, causing log messages
	  to be output.

	  ASTERISK-26349 #close

	  Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f

2016-08-16 15:34 +0000 [137aa2f13c]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Do not crash on ACKs from unknown endpoints.

	  The endpoint identification PJSIP module is intended to identify which
	  endpoint an incoming request is from. If an endpoint is not identified,
	  then an artificial endpoint is used in its place when proceeding.

	  The problem is that the ACK request type is an exception to the rule.
	  The artificial endpoint is not used when processing an ACK. This results
	  in the possibility of having a NULL endpoint being used further on.

	  The reason ACK is an exception is an attempt not to spam security logs
	  with unidentified requests. Presumably, you've already logged the
	  unidentified request on the preceeding INVITE.

	  Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion
	  didn't cause an issue. A new change in 13.10 added endpoint ACL checking
	  shortly after endpoint identification. Because we are accessing a NULL
	  endpoint, this ACL check resulted in a crash.

	  The fix here is to be sure to retrieve the artificial endpoint for all
	  request types. ACKs still do not generate unidentified request security
	  events.

	  ASTERISK-26264 #close
	  Reported by nappsoft

	  AST-2016-006

	  Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703

2016-08-23 06:35 +0000 [f877e62cc9]  Corey Farrell <git@cfware.com> (license 5909)

	* chan_sip: Don't allocate new RTP instances on top of old ones.

	  In some scenarios dialog_initialize_rtp can be called multiple times on
	  the same dialog.  This can cause RTP instances to be leaked along with
	  multiple file descriptors for each instance.

	  This change makes it so the existing RTP instances are destroyed and
	  not overwritten, stopping the memory leak.

	  ASTERISK-26272 #close
	  patches:
	    ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909)

	  Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73

2016-09-06 15:25 +0000 [b17ee86148]  Matt Jordan <mjordan@digium.com>

	* res/res_stasis_playback: Cancel the entire playlist when a stop occurs

	  Prior to this patch, a stop issued by a delete of a Playback resource
	  (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop
	  the current media URI playing. Subsequent URIs specified by a playback
	  operation would then proceed on, even though we had just indicated to
	  the User that the Playback was finished *and* after they had just
	  'deleted' the resource. Whoops.

	  This patch corrects it by bailing out of the sequence of URIs to play if
	  one of them is terminated with an AST_CONTROL_STREAM_STOP indication.

	  ASTERISK-26341 #close

	  Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42

2016-08-29 12:30 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 14.0.0-beta2 Released.

2016-08-29 07:29 +0000 [9cdf44668d]  Joshua Colp <jcolp@digium.com>

	* Release summaries: Add summaries for 14.0.0-beta2

2016-08-29 07:29 +0000 [73d39f2029]  Joshua Colp <jcolp@digium.com>

	* Release summaries: Remove previous versions

2016-08-29 07:29 +0000 [e8a97775ee]  Joshua Colp <jcolp@digium.com>

	* .version: Update for 14.0.0-beta2

2016-08-29 07:29 +0000 [345409825a]  Joshua Colp <jcolp@digium.com>

	* .lastclean: Update for 14.0.0-beta2

2016-08-29 07:29 +0000 [105c1168f7]  Joshua Colp <jcolp@digium.com>

	* realtime: Add database scripts for 14.0.0-beta2

2016-08-29 06:31 +0000 [8927b52634]  Joshua Colp <jcolp@digium.com>

	* alembic: Fix downgrade path.

	  The 3772f8f828da version was referencing a previous version
	  that did not exist in the 14.0 branch. It has been fixed to
	  reference the correct previous version.

	  Change-Id: I004d0fcfdfe1d1bb6f01c6dac2b69f6b1f40ae51

2016-08-11 12:18 +0000 [9a95c6dea3]  gtjoseph <gjoseph@digium.com>

	* res_pjsip:  Fail global load if debug or default_from_user are empty

	  If debug was specified in the global configuration but left blank,
	  the logger would treat it as a wildcard and log all hosts.  If
	  default_from_user was empty, a crash would result.

	  The global apply handler now checks for empty strings.

	  ASTERISK-26239 #close
	  ASTERISK-26238 #close

	  Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336

2016-08-11 11:24 +0000 [aaee8160bc]  gtjoseph <gjoseph@digium.com>

	* res_pjsip_caller_id:  Copy header name to short header name

	  When compact_headers was set, we were sending a zero-length header name
	  for PAI and RPID because we always forced the short header name length
	  to 0.  We did this because we cloned the header from "From" and wanted
	  to clear "f" from the sname.  By cloning however, we bypass pjproject's
	  automatic logic that sets sname to name if there's no compact form of
	  the header, which there isn't for PAI and RPID.  So now we force sname
	  to be the same as name right after we set name.

	  res_pjsip_diversion needed the same treatment for the Diversion header.

	  ASTERISK-26241 #close

	  Change-Id: I633ec139630cd83809aae00336cee4a10077e467

2016-08-11 12:01 +0000 [7af0eac02a]  gtjoseph <gjoseph@digium.com>

	* autohints:  Update CHANGES and extensions.conf.sample

	  Make it clear that we're talking about device state hints and add
	  an entry to the sample config.

	  Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433

2016-08-11 10:50 +0000 [ef0bf47bb3]  Kevin Harwell <kharwell@digium.com>

	* alembic: add auth_username to endpoint's identify_by enum

	  A new identify_by option was added recently, auth_username. However, this
	  setting was not added as an allowable choice in the database enumeration
	  value.

	  This patch updates the current enumeration, adding in the new setting.

	  ASTERISK-26268 #close

	  Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8

2016-08-08 14:50 +0000 [a1d6b14c40]  Richard Mudgett <rmudgett@digium.com>

	* res_srtp: Move SDP SRTP code from the core to res_srtp.

	  A patch made to the master branch (Now the 14 branch) inadvertently made
	  libsrtp a required dependency in order to compile Asterisk.  Rather than
	  create dummy defines to substitute for the defines supplied by libsrtp
	  when libsrtp is not available, most of the code in sdp_srtp.c is moved
	  into res_srtp.c.  This gets more code out of Asterisk's core that isn't
	  used when SRTP is not available.  This also makes another inadvertent
	  required dependency on libsrtp by Asterisk's core unlikely.

	  ASTERISK-26253 #close
	  Reported by: Ben Merrills

	  Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7

2016-08-09 12:07 +0000 [a783e1e60d]  Kevin Harwell <kharwell@digium.com>

	* alembic/sqlalchemy: auto increment only allowed on a single column

	  The extensions table defined two columns (id and priority) as primary key
	  autoincrement columns. However only one is allowed when defining the primary
	  key.

	  This patch removes the autoincrement attribute from the priority column since
	  it does not need to be as such and really should not have been on there in the
	  first place.

	  This patch also removes 'context', 'exten', and 'priority' from the primary key
	  index and creates a new combined unique contraint index on them.

	  ASTERISK-26183 #close

	  Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b

2016-08-10 11:47 +0000 [9c56f798f6]  gtjoseph <gjoseph@digium.com>

	* res_resolver_unbound:  Allow compilation with libunbound version < 1.5

	  libunbound at version 1.4.20 (which CentOS still uses) declared all
	  of their string function parameters as as 'char *'.  1.4.21 changed
	  them all to 'const char *'.  Thankfully 1.4.21 also introduced the
	  UNBOUND_VERSION_MAJOR define so configure now checks for that and
	  sets HAVE_UNBOUND_CONST_PARAMS.  res_resolver_unbound then checks
	  that and casts away the 'const' if it's not set.

	  Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and
	  Fedora24 (1.5.4).  There are a few failing tests to be addressed though.

	  ASTERISK-26283 #close

	  Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148

2016-08-01 16:13 +0000 [1ad00c1c30]  gtjoseph <gjoseph@digium.com>

	* menuselect:  Add an opaque "member_data" string to the acceptable xml

	  Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe

2016-07-17 18:28 +0000 [815b6f72f8]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Update for pjproject 2.5.5

	  Add more --disable-* switches to Makefile.rules including
	  --disable-opus which was causing bundled pjproject to fail with
	  "undefined reference" errors in libasteriskpj.

	  Changed PJ_ENABLE_EXTRA_CHECK to 1.

	  Removed 2 obsolete patches and added a new one.
	  The new one was merged by Teluu on 6/27/2016.

	  ASTERISK-26148 #close

	  Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063
	  (cherry picked from commit 4cf02b5584ce33bb0a64408c27bf20c19bc4ce13)

2016-07-29 13:13 +0000 [c95b611a73]  Mark Michelson <mmichelson@digium.com>

	* Remove SILK payload mappings from Asterisk core.

	  SILK is a bit of a hog when it comes to using up our limited number of
	  dynamic payload types in the RTP engine. By freeing up four slots, it
	  allows for other codecs to potentially take the place.

	  Now, codec_silk.so will dynamically use the payload slots in the RTP
	  engine when it loads.

	  A better fix would be make RTP dynamic payload types actually
	  dynamic. However, at this stage of Asterisk 14 development, this is a
	  risky move that would be imprudent.

	  Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612

2016-07-27 12:36 +0000 [bc94ccbcdd]  Kevin Harwell <kharwell@digium.com>

	* rtp_engine: Failed assertion and wrong name given for codec

	  Fixed an assert check that would trigger when the passed in value was negative.
	  The negative value was being cast to an unsigned value. This resulted in the
	  check failing.

	  Also fixed another problem when loading formats in the engine. When setting the
	  mime type the format's name was being passed in instead of the codec's name.

	  Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c

2016-07-26 23:19 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 14.0.0-beta1 Released.

2016-07-26 17:22 +0000 [a7233fbf3e]  Mark Michelson <mmichelson@digium.com>

	* Release summaries: Add summaries for 14.0.0-beta1

2016-07-26 16:24 +0000 [c327430ea0]  Mark Michelson <mmichelson@digium.com>

	* Release summaries: Remove previous versions

2016-07-26 16:24 +0000 [763a18bc9d]  Mark Michelson <mmichelson@digium.com>

	* .version: Update for 14.0.0-beta1

2016-07-26 16:24 +0000 [ce6898bd3c]  Mark Michelson <mmichelson@digium.com>

	* .lastclean: Update for 14.0.0-beta1

2016-07-26 16:24 +0000 [ebc477aa5d]  Mark Michelson <mmichelson@digium.com>

	* realtime: Add database scripts for 14.0.0-beta1

2016-07-26 16:00 +0000 [1838b283aa]  Mark Michelson <mmichelson@digium.com>

	* ChangeLog: Updated for 14.0.0

2016-07-26 15:02 +0000 [f196cf975d]  Mark Michelson <mmichelson@digium.com>

	* Release summaries: Add summaries for 14.0.0

2016-07-26 14:01 +0000 [699a7390eb]  Mark Michelson <mmichelson@digium.com>

	* .version: Update for 14.0.0

2016-07-26 14:01 +0000 [4b17a11d7d]  Mark Michelson <mmichelson@digium.com>

	* .lastclean: Update for 14.0.0

2016-07-26 14:01 +0000 [bb9dcae98c]  Mark Michelson <mmichelson@digium.com>

	* realtime: Add database scripts for 14.0.0

2016-07-24 18:27 +0000 [90f445729d]  gtjoseph <gjoseph@digium.com>

	* menuselect:  Various menuselect enhancements

	  * Add 'external' as a support level.
	  * Add ability for module directories to add entries to the menu
	    by adding members to the <module_prefix>/<module_prefix>.xml file.
	  * Expand the description field to 3 lines in the ncurses implementation.
	  * Allow the description field to wrap in the newt implementation.
	  * Add description field to the gtk implementation.

	  Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808

2016-07-24 16:51 +0000 [f75401b1e3]  Joshua Colp <jcolp@digium.com>

	* ari: Update version.

	  New functionality has been added so the version has been
	  bumped to one over the 13 version.

	  Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69

2016-07-23 08:51 +0000 [58759bd77c]  gtjoseph <gjoseph@digium.com>

	* asterisk.c:  Add auto generation and persistence of UUID

	  Upcoming features will require the generation and persistence
	  of a UUID.

	  Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d

2016-07-22 14:44 +0000 [46b4e673ae]  Mark Michelson <mmichelson@digium.com>

	* Fix sqlalchemy error regarding identifier length.

	  sqlalchemy was complaining:

	  sqlalchemy.exc.IdentifierError: Identifier
	  'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30
	  characters

	  This fixes the problem by changing the index name to be
	  "ps_contacts_qualifyfreq_exp" instead.

	  ASTERISK-26227 #close
	  Reported by Mark Michelson

	  Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9

2016-07-22 07:01 +0000 [633c34c411]  gtjoseph <gjoseph@digium.com>

	* build_tools: Update make_version for 14

	  Also remove svn stuff

	  Change-Id: I95d762f7cbbe5eb01117bde8779515d51a0bb06a

2016-07-19 13:39 +0000 [c82f24f36a]  Alexander Traud <pabstraud@compuserve.com>

	* codecs: Add iLBC 20.

	  Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk
	  defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests
	  this.

	  ASTERISK-26218 #close
	  ASTERISK-26221 #close
	  Reported by: Aaron Meriwether

	  Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa

2016-07-15 16:16 +0000 [6e2e3915c8]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Whitespace and comment cleanup.

	  Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38

2016-07-19 13:18 +0000 [5efb5b38e8]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi.c: Fix deadlock potential in fax redirection.

	  The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to
	  deadlock if an incoming fax happens during the Playback or similar
	  application.

	  * Fixed the potential deadlock by not calling ast_async_goto() with the
	  channel lock held.

	  ASTERISK-26216 #close
	  Reported by: Richard Mudgett

	  Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa

2016-07-13 18:49 +0000 [a1d36c89e0]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix deadlock potential in fax redirection.

	  The sip_read() has the potential to deadlock if an incoming fax happens
	  during the Playback or similar application.

	  * Fixed the potential deadlock by not calling ast_async_goto() with the
	  channel lock held.

	  * Made always eat the fax detection frame whether there is a fax extension
	  or not.

	  ASTERISK-26216
	  Reported by: Richard Mudgett

	  Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e

2016-07-13 18:48 +0000 [4dfadcb025]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip.c: Fix deadlock potential in fax redirection.

	  The chan_pjsip_cng_tone_detected() has the potential to deadlock if an
	  incoming fax happens during the Playback or similar application.

	  * Fixed the potential deadlock by not calling ast_async_goto() with the
	  channel lock held.

	  * Made always eat the fax detection frame whether there is a fax extension
	  or not.

	  ASTERISK-26216
	  Reported by: Richard Mudgett

	  Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5

2016-07-12 17:33 +0000 [964ae54ecf]  Richard Mudgett <rmudgett@digium.com>

	* res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook.

	  The fax_detect_framehook() has the potential to deadlock if an incoming
	  fax happens during the Playback or similar application.

	  * Fixed the potential deadlock by not calling ast_async_goto() with the
	  channel lock held.

	  * Made always eat the fax detection frame whether there is a fax extension
	  or not.

	  * Made only detach the framehook if we detected a fax and not on other
	  possible frames.

	  ASTERISK-26216
	  Reported by: Richard Mudgett

	  Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d

2016-07-12 17:24 +0000 [c3462adeb8]  Richard Mudgett <rmudgett@digium.com>

	* res_fax: Fix FAXOPT(faxdetect) timeout option.

	  The fax detection timeout option did not work because basically the wrong
	  variable was checked in fax_detect_framehook().  As a result, the timer
	  would timeout immediately and disable fax detection.

	  * Fixed ignoring negative timeout values.  We'd complain and then go right
	  on using the negative value.

	  * Fixed destroy_faxdetect() in the off-nominal case of an incomplete
	  object creation.

	  * Added more range checking to FAXOPT(gateway) timeout parameter.

	  ASTERISK-26214 #close
	  Reported by: Richard Mudgett

	  Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976

2016-07-18 16:16 +0000 [c03e27c1c8]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi: Add faxdetect_timeout option.

	  The new option allows the channel driver's faxdetect option to timeout on
	  a call after the specified number of seconds into a call.  The new feature
	  is disabled if the timeout is set to zero.  The option is disabled by
	  default.

	  * Don't clear dsp_features after passing them to the dsp code in
	  my_pri_ss7_open_media().  We should still remember them especially for the
	  new faxdetect_timeout option.

	  ASTERISK-26214
	  Reported by: Richard Mudgett

	  Change-Id: Ieffd3fe788788d56282844774365546dce8ac810

2016-07-15 20:44 +0000 [d11731ac2f]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Add fax_detect_timeout endpoint option.

	  The new endpoint option allows the PJSIP channel driver's fax_detect
	  endpoint option to timeout on a call after the specified number of
	  seconds into a call.  The new feature is disabled if the timeout is set
	  to zero.  The option is disabled by default.

	  ASTERISK-26214
	  Reported by: Richard Mudgett

	  Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d

2016-07-21 10:28 +0000 [56b4112659]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice.

	  This patch removed call of pjsip_tx_data_dec_ref in send_notify
	  if send_request failed.
	  The pjsip_dlg_send_request deletes the message on error by itself.

	  It seems this patch fixes next issues:
	  ASTERISK-26199
	  ASTERISK-26166
	  ASTERISK-26174

	  Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a

2016-07-21 09:05 +0000 [52cbdf2393]  gtjoseph <gjoseph@digium.com>

	* chan_sip: Prevent deadlock when issuing "sip show channels"

	  sip_show_channels locks the dialogs container first then locks each
	  sip_pvt so it can spit out the details.  The rest of sip dialog
	  processing locks the sip_pvt first then locks the dialogs container
	  if it needs to.  Both lock in the order they need but deadlocks can
	  result.  To fix, sip_show_channels and sip_show_channelstats have
	  been converted to use an iterator rather than ao2_callback.  This way
	  the container is locked only while getting the next entry and is
	  unlocked when the callback is called.

	  ASTERISK-23013 #close

	  Change-Id: Id9980419909e811f89484950ed46ef117b9eb990

2016-07-13 05:24 +0000 [2103ad1fec]  Alexander Traud <pabstraud@compuserve.com>

	* res_srtp: Enable AES-256 and AES-GCM.

	  ASTERISK-26190 #close

	  Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b

2016-07-18 22:46 +0000 [05cfe1a76e]  Corey Farrell <git@cfware.com>

	* Add conditional support for noreturn functions.

	  This adds support for tagging functions with the noreturn attribute.
	  If DO_CRASH is enabled then ast_do_crash never returns.  If AST_DEVMODE
	  and DO_CRASH are enabled then failed assertions never return.  This can
	  resolve a large number of false positives with static analyzers.

	  ASTERISK-26220 #close

	  Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753

2016-07-15 19:28 +0000 [0c88fb460f]  Corey Farrell <git@cfware.com>

	* pbx: Create pbx_sw.c for management of 'struct ast_sw'.

	  This changes context switches from a linked list to a vector, makes
	  'struct ast_sw' opaque to pbx.c.

	  Although ast_walk_context_switches is maintained the procedure is no
	  longer efficient except for the first call (inc==NULL).  This
	  functionality is replaced by two new functions implemented by vector
	  macros.
	  * ast_context_switches_count (AST_VECTOR_SIZE)
	  * ast_context_switches_get (AST_VECTOR_GET)

	  As with ast_walk_context_switches callers of these functions are
	  expected to have locked contexts.  Only a few places in Asterisk walked
	  the switches, they have been converted to use the new functions.

	  Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998

2016-07-19 04:48 +0000 [6fca2b3bf0]  Alexander Traud <pabstraud@compuserve.com>

	* Makefile: Retain XML Declaration and DTD in docs.

	  Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo,
	  the XML Declaration and DTD were overwritten by this.

	  ASTERISK-26212 #close

	  Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd

2016-07-18 18:40 +0000 [cf1188a1be]  Corey Farrell <git@cfware.com>

	* Unit tests: Use AST_TEST_DEFINE in conditional code only.

	  If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead
	  code.  This places all existing unit tests into a conditional block if
	  they weren't already.

	  ASTERISK-26211 #close

	  Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686

2016-07-18 09:22 +0000 [e9daa34261]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip_mwi: remove unneeded check on endpoint's contacts.

	  The function create_mwi_subscriptions_for_endpoint checks
	  if there is active contacts by retrieving aors and contacts.

	  This function is used to create all unsolicited mwi subscriptions
	  on startup and is used when contact added.

	  In both cases it's not necessary to check if there are contacts.
	  The contacts are needed when asterisk sends mwi.

	  ASTERISK-26200 #close

	  Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa

2016-07-18 05:13 +0000 [cb5e3445be]  Alexander Traud <pabstraud@compuserve.com>

	* res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets.

	  With this change, the initial RTP sequence number is randomly chosen not between
	  0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over
	  counter (ROC) synchronization is not lost for sRTP, when the very first RTP
	  packets get lost; see http://srtp.sourceforge.net/faq.html#Q6

	  ASTERISK-26207 #close

	  Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464

2016-07-18 04:14 +0000 [6428580e7f]  Alexander Traud <pabstraud@compuserve.com>

	* Makefile: Suppress echoing of target 'config' again.

	  ASTERISK-26038 #close

	  Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f

2016-07-15 02:59 +0000 [e2e8713b84]  Corey Farrell <git@cfware.com>

	* pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'.

	  This changes context ignore patterns from a linked list to a vector,
	  makes 'struct ast_ignorepat' opaque to pbx.c.

	  Although ast_walk_context_ignorepats is maintained the procedure is no
	  longer efficient except for the first call (inc==NULL).  This
	  functionality is replaced by two new functions implemented by vector
	  macros.
	  * ast_context_ignorepats_count (AST_VECTOR_SIZE)
	  * ast_context_ignorepats_get (AST_VECTOR_GET)

	  As with ast_walk_context_ignorepats callers of these functions are
	  expected to have locked contexts.  Only a few places in Asterisk walked
	  the ignorepats, they have been converted to use the new functions.

	  Change-Id: I78f2157d275ef1b7d624b4ff7d770d38e5d7f20a

2016-07-14 13:51 +0000 [be36bd7ca5]  Corey Farrell <git@cfware.com>

	* pbx: Create pbx_include.c for management of 'struct ast_include'.

	  This changes context includes from a linked list to a vector, makes
	  'struct ast_include' opaque to pbx.c.

	  Although ast_walk_context_includes is maintained the procedure is no
	  longer efficient except for the first call (inc==NULL).  This
	  functionality is replaced by two new functions implemented by vector
	  macros.
	  * ast_context_includes_count (AST_VECTOR_SIZE)
	  * ast_context_includes_get (AST_VECTOR_GET)

	  As with ast_walk_context_includes callers of these functions are
	  expected to have locked contexts.  Only a few places in Asterisk walked
	  the includes, they have been converted to use the new functions.

	  const have been applied where possible to parameters for ast_include
	  functions.

	  Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60

2016-07-14 03:25 +0000 [d3348c51b5]  Corey Farrell <git@cfware.com>

	* features.c: Remove unneeded adsi.h include.

	  adsi.h is no longer used by features.c since parking was moved to a
	  module.

	  Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59

2016-06-30 15:58 +0000 [273052f404]  Mark Michelson <mmichelson@digium.com>

	* Update support for SILK format.

	  This commit adds scaffolding in order to support the SILK audio format
	  on calls. Roughly, this is what is added:

	  * Cached silk formats. One for each possible sample rate.
	  * ast_codec structures for each possible sample rate.
	  * RTP payload mappings for "SILK".

	  In addition, this change overhauls the res_format_attr_silk file in the
	  following ways:

	  * The "samplerate" attribute is scrapped. That's native to the format.
	  * There are far more checks to ensure that attributes have been
	    allocated before attempting to reference them.
	  * We do not SDP fmtp lines for attributes set to 0.

	  These changes make way to be able to install a codec_silk module and
	  have it actually work. It also should allow for passthrough silk calls
	  in Asterisk.

	  Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e

2016-07-14 07:45 +0000 [31967dacdf]  Richard Miller (license 5685)

	* app_queue: Only remove queue member from pending when state changes.

	  It is possible for a not in use state change to occur multiple
	  times causing a queue member to be removed from the pending call
	  container prematurely.

	  The first not in use state change will remove the queue member
	  from the container. At this moment the member may be called and
	  placed in the pending container. After this another not in use
	  state change can be received which will remove it from the
	  container. Despite being called at this point the code will
	  incorrectly see that there are no pending calls to it.

	  This change only removes it from the pending container if the
	  state has actually changed.

	  ASTERISK-26133 #close
	  patches:
	    app_queue.diff submitted by Richard Miller (license 5685)

	  Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0

2016-07-14 02:40 +0000 [f3608b50d7]  Corey Farrell <git@cfware.com>

	* pbx: Fix leak of timezone for time based includes.

	  Create include_free to run ast_destroy_timing and ast_free, use that in
	  all places that freed an ast_include structure.  This fixes a couple of
	  paths that previously did not run ast_destroy_timing.

	  ASTERISK-26196 #close

	  Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838

2016-07-13 17:45 +0000 [63ac4c9487]  Kevin Harwell <kharwell@digium.com>

	* translate: explicit format destination not properly set

	  If the destination format's name differed from the codec name then the
	  translator's explict_dst field would be improperly set. In some circumstances
	  it would end up setting it to a newly created format that has the same name
	  as the codec when it actually needed to be the given destination codec.

	  This could cause the translation path to use the wrong format. For instance,
	  if an endpoint had specified 'myulaw' as a format the translator could end up
	  using a 'ulaw' format (with whatever/default settings) instead. If the format
	  attribute settings differed between the two then there may unexpected results
	  during processing.

	  This patch removes the name check when building the translation path. This
	  should make it always set the translator's explicit_dst to the given destination
	  format as long as the sample rate and types match.

	  Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5

2016-07-08 11:46 +0000 [2f26512fd8]  Richard Mudgett <rmudgett@digium.com>

	* stasis_endpoint.c: Fix contactstatus_to_json().

	  The roundtrip_usec json member is optional.  If it isn't present then
	  don't put it into the converted json structure where ast_json_pack()
	  will choke on it.

	  Change-Id: I39bb2f86154ef54591270c58bfda8635070f9ea0

2016-07-11 10:22 +0000 [bc1ff41be7]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_options.c: Fix container operation.

	  aor_observer_deleted() needs to operate on all contacts found for the
	  deleted AOR instead of only the first one found.  This is really only a
	  problem if there is more than one contact for the AOR.

	  Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1

2016-07-11 10:21 +0000 [eabcfeeaa3]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_configuration.c: Misc cleanups.

	  * Fix some whitespace in various routines.

	  * Rename i to iter in persistent_endpoint_update_state().

	  * Fix off-nominal copy/paste message wording in
	  persistent_endpoint_contact_deleted_observer()

	  Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8

2016-07-13 13:45 +0000 [f73ddde7d4]  Corey Farrell <git@cfware.com>

	* chan_sip: Fix reference leak in mwi_event_cb

	  Cleanup the peer reference when stasis_subscription_final_message is
	  true.  Also free peer_name even if peer exists, after reload a new
	  peer_name will be allocated.

	  ASTERISK-26193 #close

	  Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69

2016-07-13 11:30 +0000 [fd54d69feb]  Corey Farrell <git@cfware.com>

	* threadpool: Fix leak in ast_threadpool_serializer_group error path.

	  ast_threadpool_serializer_group leaks a reference to ser when listener
	  is allocated but tps is not.  Although listener takes the reference to
	  ser cleanup functions are not run without tps.

	  ASTERISK-26191 #close

	  Change-Id: Ie3ccf69a3f1e676c2ef62a77067c0cb57dc9a585

2016-06-22 07:13 +0000 [85212f2799]  Eugene Voityuk <eugene@thirdlane.com>,Alexander Traud <pabstraud@compuserve.com>

	* res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS.

	  Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS)
	  support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added
	  for DTLS. The source code from main/tcptls.c should have been re-used to ease
	  security audits. Therefore, this change rolls back the change from July 2015 and
	  re-uses the code from July 2014. This has the additional benefits to work under
	  CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well.

	  ASTERISK-25659 #close
	  Reported by: StefanEng86, urbaniak, pay123
	  Tested by: sarumjanuch, traud
	  patches:
	  res_rtp_asterisk.patch submitted by sarumjanuch
	  dtls_centos_step_1.patch submitted by traud
	  dtls_centos_step_2.patch submitted by traud

	  Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c

2016-06-24 19:55 +0000 [0d487b53b1]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_session: Check for presence of an active negotiator

	  It is possible in a hypothetical situation for a session refresh to be
	  invoked on a PJSIP when the negotiatior on the INVITE session has not
	  yet been established. While this shouldn't occur with existing uses of
	  ast_sip_session_refresh, the crashes that occur due to improperly
	  calling PJSIP functions that expect a non-NULL negotiatior are
	  avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this
	  means that simply checking for the presence of the negotiator before
	  passing it to other PJSIP functions that use it is allowable. As such,
	  this patch adds checks for the presence of the negotiator before calling
	  PJSIP functions that assume it is non-NULL.

	  Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d

2015-10-19 18:55 +0000 [c49833653b]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_pubsub: Add additional debug statements

	  When something very sad and wrong occurs, it's challenging sometimes to
	  figure out why. This patch adds some additional debug statements on
	  off-nominal paths to try and make debugging easier.

	  Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640

2015-10-19 18:55 +0000 [f12311ee69]  Matt Jordan <mjordan@digium.com>

	* res/res_corosync: Raise a Stasis message on node join/leave events

	  When res_corosync detects that a node leaves or joins, it currently is
	  informed of this via Corosync callbacks. However, there are a few
	  limitations with the information presented:
	  (1) While we have information that Corosync is aware of - such as the
	      Corosync nodeid - that information is really only useful inside of
	      Corosync or res_corosync. There's no way to translate a Corosync
	      nodeid to some other internally useful unique identifier for the
	      Asterisk instance that just joined or left the cluster.
	  (2) While res_corosync is notified of the instance joining or leaving
	      the cluster, it has no mechanism to inform the Asterisk core or
	      other modules of this event. This limits the usefulness of res_corosync
	      as a heartbeat mechanism for other modules.

	  This patch addresses both issues.

	  First, it adds the notion of a cluster discovery message both within the
	  Stasis message bus, as well as the binary event messages that
	  res_corosync uses to transmit data back and forth within the cluster.
	  When Asterisk joins the cluster, it sends a discovery message to the other
	  nodes in the cluster, which correlates the Corosync nodeid along with
	  the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids
	  to Asterisk EIDs, such that it can map changes in cluster state with the
	  Asterisk instance that has that nodeid. Likewise, when an Asterisk
	  instance receives a discovery message from a node in the cluster, it now
	  sends its own discovery message back to the originating node with the
	  local Asterisk EID. This lets Asterisk instances within the cluster
	  build a complete picture of the other Asterisk instances within the
	  cluster.

	  Second, it publishes the discovery messages onto the Stasis message bus.
	  Said messages are published whenever a node joins or leaves the cluster.
	  Interested modules can subscribe for the ast_cluster_discovery_type()
	  message under the ast_system_topic() and be notified when changes in
	  cluster state occur.

	  Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465

2016-07-13 08:57 +0000 [a3f4141f6f]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf.

	  Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version.

	  ASTERISK-26046 #close

	  Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7

2016-07-11 20:07 +0000 [886f2cab23]  gtjoseph <gjoseph@digium.com>

	* rest_api/channels:  Fix multiple issues with create and dial

	  * We weren't properly subscribing to the channel and it's originator
	    on create.
	  * We weren't doing a publish_dial after calling ast_call on dial.
	  * We weren't calling depart_bridge when a channel left the dial bridge.

	  The first 2 issues were causing events to not be generated and the third
	  was actually causing channels to not get properly destroyed when hung up.

	  Together these 3 issues were causing the new
	  rest_apichannels/create_dial_bridge tests to fail.

	  As a result of the fixes, the cdr state machine had to be slightly
	  tweaked to allow bridge leave events without asserting and the tests
	  themselves had to be updated to account for the channels now cleaning
	  themselves up.

	  Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8

2016-07-11 10:25 +0000 [b85446d039]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Fix statsd regression.

	  The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
	  patch introduced several regressions when the newly created "Updated"
	  state goes out for each endpoint registration refresh.

	  1) It restarted any OPTIONS RTT ping cycle.

	  2) It would interfere with a currently active ping and throw off that
	  ping's resulting RTT calculation.

	  3) It cleared the RTT time each time the endpoint was refreshed.

	  4) The cleared RTT time was sent out as a statsd update each time.

	  5) It created two AMI events for each update.

	  * Revert the original patch and reimplement it.  Now the current contact
	  status state is re-sent instead of the state being momentarily toggled
	  every time the endpoint refreshes its registration.  The statsd events are
	  not created for the re-sent refresh because they are sent after every
	  OPTIONS ping.

	  ASTERISK-26160 #close
	  Reported by: Matt Jordan

	  Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1

2016-07-10 19:08 +0000 [4ad333bb0e]  Joshua Colp <jcolp@digium.com>

	* func_odbc: Fix connection deadlock.

	  The func_odbc module was modified to ensure that the
	  previous behavior of using a single database connection
	  was maintained. This was done by getting a single database
	  connection and holding on to it. With the new multiple
	  connection support in res_odbc this will actually starve
	  every other thread from getting access to the database as
	  it also maintains the previous behavior of having only
	  a single database connection.

	  This change disables the func_odbc specific behavior if
	  the res_odbc module is running with only a single database
	  connection active. The connection is only kept for the
	  duration of the request.

	  ASTERISK-26177 #close

	  Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f

2016-07-12 03:50 +0000 [110b01a0bc]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Allow own CFLAGS on ./configure.

	  Before this change, make failed with the error
	  Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH
	  when CFLAGS were supplied to the configure script. This was introduced with
	  <https://reviewboard.asterisk.org/r/1852/> which disabled BUILD_NATIVE when
	  CFLAGS were supplied. Those who need different -march= values, please, go for
	  ./configure
	  make menuselect.makeopts or make menuselect
	  ./menuselect/menuselect --disable BUILD_NATIVE

	  ASTERISK-25289 #close

	  Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc

2016-07-11 13:42 +0000 [44f16af7cc]  Richard Mudgett <rmudgett@digium.com>

	* ast_expr2: Fix off-nominal memory leak.

	  Thanks to ibercom for pointing out a memory leak that was missed
	  in the earlier patch for the issue.

	  ASTERISK-26119
	  Reported by: Alexei Gradinari

	  Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71

2016-07-11 10:17 +0000 [8476a9332f]  Alexander Traud <pabstraud@compuserve.com>

	* install_prereq: Checkout of libSRTP 1.5.x.

	  Since 5th November 2014, the master branch of libSRTP changed the prefix of
	  several member names and is not compatible with the source code in Asterisk
	  anymore. Therefore instead, this change checks out the latest version of the
	  libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as
	  backend. This makes AES-GCM and AES-IN possible.

	  ASTERISK-22131 #close

	  Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6

2016-07-09 13:32 +0000 [ad30d60c69]  Corey Farrell <git@cfware.com>

	* chan_sip: Fix reference leaks in error paths.

	  * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error.
	  * build_peer leaks peer on failure to allocate the endpoint.

	  This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed
	  with an unref in the appropriate place.

	  ASTERISK-26184 #close

	  Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12

2016-07-07 12:44 +0000 [7408c51a48]  Corey Farrell <git@cfware.com>

	* REF_DEBUG: Prevent logging of container node objects.

	  Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being
	  recorded to the refs log for the node being replaced.  This prevents
	  logging of those unrefs since they would produce errors in
	  refcounter.py.

	  ASTERISK-26181 #close

	  Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4

2016-07-04 16:38 +0000 [c832f100d9]  Alexei Gradinari <alex2grad@gmail.com>

	* res_sorcery_realtime: fix bug when successful UPDATE is treated as failed

	  If the SQL UPDATE statement changes nothing then SQLRowCount returns 0.
	  This value should be treated as success.
	  But the function sorcery_realtime_update treats it as failed.

	  This bug was found using stress tests on PJSIP.
	  If there are 2 consecutive SIP REGISTER requests with the same contact data
	  during 1 second then res_pjsip_registrar adds contact location on 1st request
	  and tries to update contact location on 2nd.
	  The update fails and res_pjsip_registrar even removes correct contact location.

	  The test "object_update_uncreated" was removed from test_sorcery_realtime.c
	  because it's now a valid situation.

	  This patch also adds missing debug of extra SQL parameter.

	  ASTERISK-26172 #close

	  Change-Id: I05a7f3051455336c9dda29efc229decf86071303

2016-07-07 10:38 +0000 [302be4809a]  Joshua Colp <jcolp@digium.com>

	* chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled.

	  Some T.38 implementations may send another re-invite after the initial
	  one which adds additional negotiation details (such as the max bitrate).
	  Currently this will fail when passthrough is being done in chan_sip as we
	  do nothing if T.38 is already active.

	  Other handlers of T.38 inside of Asterisk (such as res_fax) handle this
	  scenario so this change adds support for it to chan_sip and res_pjsip_t38.
	  If a request to negotiate is received while T.38 is already enabled a
	  new re-INVITE is sent and negotiation is done again.

	  ASTERISK-26179 #close

	  Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c

2016-07-07 10:55 +0000 [fb96492ec4]  Scott Griepentrog <scott@griepentrog.com>

	* PJSIP: provide valid tcp nodelay option for reuse

	  When using TCP transport with chan_pjsip, the TCP_NODELAY
	  option value was allocated on the stack, then passed as a
	  pointer to the tcp transport configuration structure, and
	  later re-used on subsequently created sockets when it was
	  no longer valid.  This patch changes the allocation to be
	  a static.

	  ASTERISK-26180 #close
	  Reported by: Scott Griepentrog

	  Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0

2016-07-06 09:29 +0000 [1c949eea6c]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: Added "subscribe_context" to endpoint

	  If specified, incoming SUBSCRIBE requests will be searched for the matching
	  extension in the indicated context. If no "subscribe_context" is specified,
	  then the "context" setting is used.

	  ASTERISK-25471 #close

	  Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514

2016-07-04 05:58 +0000 [32cb981d04]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf.

	  Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This
	  avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is
	  using AS_HELP_STRING everywhere else already.

	  ASTERISK-26046

	  Change-Id: I8299faf504ceaeee3e39930c59293809e116c631

2016-06-22 17:26 +0000 [9f2c007254]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Don't send extra BYE if SDP invalid.

	  When an answer SDP is invalid we were disconnecting the outgoing call and
	  sending two BYE requests.  The first BYE was sent by PJPROJECT because of
	  the invalid SDP answer.  The second BYE was sent by Asterisk because it
	  thought the canceled call was the result of the RFC5407 section 3.1.2 race
	  condition.

	  * Made not send the BYE on a canceled session if the SDP negotiation is
	  incomplete because PJPROJECT has already sent a BYE for the failed
	  negotiation.

	  ASTERISK-25772 #close
	  Reported by:  Dmitriy Serov

	  Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836

2016-06-27 17:19 +0000 [08d3b9a89e]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: End call on initial invalid SDP negotiation.

	  When an incoming call defers SDP negotiation and then sends us an invalid
	  SDP in the ACK, we need to send a BYE to disconnect the call.  In this
	  case SDP negotiation has failed and we don't have valid media streams
	  negotiated.

	  ASTERISK-25772

	  Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8

2016-06-23 15:13 +0000 [e6e12c752c]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip.c: Register PJMEDIA error code decoder.

	  Registering the PJMEDIA error codes allows errors found when parsing an
	  incoming SDP to be easier to figure out.

	  "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)"
	  is much easier to understand than "Unknown error 220030".

	  ASTERISK-25772

	  Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0

2016-06-27 16:56 +0000 [5d2fc6bab7]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Remove unused parameter from handle_incoming().

	  Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa

2016-06-22 18:02 +0000 [656ed73ac6]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Add missing NULL checks when using pjsip_inv_end_session().

	  pjsip_inv_end_session() is documented as being able to return the
	  passed in tdata parameter set to NULL on success.

	  Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047

2016-06-30 15:17 +0000 [4f7b859726]  Richard Mudgett <rmudgett@digium.com>

	* features: Fix channel datastore access.

	  Found as a result of the testsuite tests/callparking test crashing.

	  Several calls to ast_get_chan_featuremap_config() and
	  ast_get_chan_features_xfer_config() did not lock the channel before
	  calling so the channel's datastore list was accessed without the lock's
	  protection.  Apparently another thread deleted a datastore on the
	  channel's list while the crashing thread was walking the list.  Crash at
	  0xdeaddead due to MALLOC_DEBUG's memory filler value as a result.

	  * Add missing channel locks to calls that were not already protected
	  as the doxygen for those calls indicates.

	  Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1

2016-06-30 08:25 +0000 [5ad7e1c09a]  gtjoseph <gjoseph@digium.com>

	* configure:  Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject

	  There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK
	  from getting set when using an external pjproject.

	  ASTERISK-26099 #close
	  Reported-by: Ross Beer

	  Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae

2016-06-29 15:31 +0000 [dab2a6b689]  Matt Jordan <mjordan@digium.com>

	* hep.conf.sample: Default 'enabled' to 'no'

	  Following the principle of least surprise, we should not be sending
	  massive numbers of PJSIP and RTCP HEP packets out into the ether to some
	  only-slightly-random IP address. Having 'enabled' set to 'no' in the
	  sample configuration file should prevent this from happening for those
	  who run 'make samples'.

	  ASTERISK-26159 #close

	  Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1

2016-06-29 15:09 +0000 [9129ac8e73]  Matt Jordan <mjordan@digium.com>

	* pjproject/patches/config_site: Increase the max number of ICE candidates

	  When negotiating ICE candidates with WebRTC capable endpoints, many
	  networks will result in a browser offering ICE candidates that exceeds
	  the default number of max candidates, 16. This patch bumps the max
	  candidates to 32, with the max checks at twice the number of candidates.
	  In practice, this has shown to be sufficient for browser/WebRTC
	  negotiation.

	  Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5

2016-06-28 09:00 +0000 [4045e6d8ba]  gtjoseph <gjoseph@digium.com>

	* codecs:  Fix ABI incompatibility created by adding format_name to ast_codec

	  Adding format_name even to the end of ast_codec caused issued with
	  binary codec modules because the pointer would be garbage in asterisk
	  when they registered.  So, the ast_codec structure was reverted and an
	  internal_ast_codec structure was created just for use in codec.c.  A new
	  internal-only API was also added (__ast_codec_register_with_format) so
	  that codec_builtin could register codecs with the format_name in a
	  separate parameter rather than in the ast_codec structure.

	  ASTERISK-26144 #close
	  Reported-by: Alexei Gradinari

	  Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba

2016-06-28 08:22 +0000 [651290a809]  gtjoseph <gjoseph@digium.com>

	* BuildSystem:  Fix a few issues hightlighted by gcc 6.x

	  gcc 6.1.1 caught a few more issues.
	  Made sure the unit tests still pass for the func_env and stdtime
	  issues.

	  ASTERISK-26157 #close

	  Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e

2016-06-28 10:33 +0000 [83f2c2573b]  Matt Jordan <mjordan@digium.com>

	* configs/basic-pbx/modules.conf: Remove 'bad' modules

	  This patch removes the following modules:
	   - pbx_functions: It never existed.
	   - res_pjsip_log_forwarder: It no longer exists.
	   - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs
	                    aren't going to be installing HOMER
	   - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't
	                    loaded, and we aren't configured to make use of the
	                    module

	  Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5

2016-06-22 11:19 +0000 [75818b4084]  Joshua Colp <jcolp@digium.com>

	* siren: Add format attribute modules for Siren7 and Siren14.

	  This change removes hardcoded SDP parsing and generation for
	  Siren7 and Siren14 from chan_sip and moves it to format attribute
	  modules so it can also be used by chan_pjsip.

	  With this the fmtp lines for both are added with the bitrate
	  information.

	  ASTERISK-26021

	  Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037

2016-06-23 04:33 +0000 [6e87bf746a]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf.

	  Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C
	  but requires ANSI C anyway.

	  ASTERISK-26046

	  Change-Id: I914c014385e1862102d90fe7650621def78db02e

2016-06-22 15:04 +0000 [8c7017f76e]  Corey Farrell <git@cfware.com>

	* res_fax: Fix reference leak in fax_v21_session_new.

	  fax_v21_session_new created a session details object but only released
	  the allocation reference during error conditions.  fax_session_new adds
	  it's own reference to details if needed so the caller is always
	  responsible for cleaning it's own reference.

	  ASTERISK-26141 #close

	  Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88

2016-06-22 14:25 +0000 [6fa3ed0679]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: improve realtime performance #2

	  The patch removes updating all Endpoints' status on startup.
	  Instead, only non-qualified aors with static contact
	  and non-qualified non-expired contacts are retrieved from the realtime to
	  update the endpoint status to ONLINE.
	  The endpoint name was added to the contact object to simply find the endpoint
	  that created this contact.

	  The status of endpoints with qualified aors will be updated by 'qualify'
	  functions.

	  ASTERISK-26061 #close

	  Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df

2016-06-22 13:41 +0000 [d293ead077]  gtjoseph <gjoseph@digium.com>

	* res_rtp_asterisk:  Fix a self-comparison identified by gcc 6

	  gcc 6 caught a previously unidentified self-comparison in
	  ice_candidate_cmp.  Fixed it and re-ordered the predicates for better
	  short-circuiting.

	  ASTERISK-26140 #close

	  Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7

2016-06-22 10:37 +0000 [c7309a5254]  gtjoseph <gjoseph@digium.com>

	* chan_unistim:  Fix memcpy in get_to_address

	  A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD)
	  was using a pointer to a pointer as the destination of a memcpy and a
	  '&' instead of '*' in the sizeof.

	  ASTERISK-26138 #close

	  Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708

2016-06-20 13:21 +0000 [b6bd97eea2]  Mark Michelson <mmichelson@digium.com>

	* Fix Alembic upgrades.

	  A non-existent constraint was being referenced in the upgrade script.
	  This patch corrects the problem by removing the reference.

	  In addition, the head of the alembic branch referred to a non-existent
	  revision. This has been fixed by referring to the proper revision.

	  This patch fixes another realtime problem as well. Our Alembic scripts
	  store booleans as yes or no values. However, Sorcery tries to insert
	  "true" or "false" instead. This patch introduces a new boolean type that
	  translates to "yes" or "no" instead.

	  ASTERISK-26128 #close

	  Change-Id: I51574736a881189de695a824883a18d66a52dcef

2016-06-22 10:51 +0000 [3b4f5d1345]  gtjoseph <gjoseph@digium.com>

	* test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO

	  Since the file was missing the depends on pjproject, it wasn't
	  picking up the pjproject related include path.  If there was no
	  system installed pjproject and pjproject-bundled was used, a compile
	  would fail because pjsip.h wasn't found.

	  ASTERISK-26139 #close

	  Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757

2016-06-22 10:55 +0000 [5f23aacda4]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf.

	  Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not
	  support the platform SVR2 from the year 1987 anymore.

	  ASTERISK-26046

	  Change-Id: I28161b037feb2d29ab46ed20e785928460226c22

2016-06-21 06:52 +0000 [804005d251]  Torrey Searle <torrey@voxbone.com>

	* res_rtp_asterisk: fix memory leak in dtls

	  ensure that cert bios get freed after creating the fingerprint

	  ASTERISK-26129 #close

	  Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451

2016-06-21 17:42 +0000 [f572b26495]  Richard Mudgett <rmudgett@digium.com>

	* res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro.

	  Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c

2016-06-12 11:19 +0000 [b57cd01404]  gtjoseph <gjoseph@digium.com>

	* res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription

	  Occasionally under load we'll attempt to send a final NOTIFY on a
	  subscription that's already been terminated and a SEGV will occur
	  down in pjproject's evsub_destroy function.  This is a result of a
	  race condition between all the paths that can generate a notify
	  and/or destroy the underlying pjproject evsub object:

	   * The client can send a SUBSCRIBE with Expires: 0.
	   * The client can send a SUBSCRIBE/refresh.
	   * The subscription timer can expire.
	   * An extension state can change.
	   * An MWI event can be generated.
	   * The pjproject transaction timer (timer_b) can expire.

	  Normally when our pubsub_on_evsub_state is called with a terminate,
	  we push a task to the serializer and return at which point the dialog
	  is unlocked.  This is usually not a problem because the task runs
	  immediately and locks the dialog again.  When the system is heavily
	  loaded though, there may be a delay between the unlock and relock
	  during which another event may occur such as the subscription timer
	  or timer_b expiring, an extension state change, etc.  These may also
	  cause a terminate to be processed and if so, we could cause pjproject
	  to try to destroy the evsub structure twice.  There's no way for us to
	  tell that the evsub was already destroyed and the evsub's group lock
	  can't tolerate this and SEGVs.

	  The remedy is twofold.

	   * A patch has been submitted to Teluu and added to the bundled
	     pjproject which adds add/decrement operations on evsub's group lock.

	   * In res_pjsip_pubsub:
	     * configure.ac and pjproject-bundled's configure.m4 were updated
	       to check for the new evsub group lock APIs.
	     * We now add a reference to the evsub group lock when we create
	       the subscription and remove the reference when we clean up the
	       subscription.  This prevents evsub from being destroyed before
	       we're done with it.
	     * A state has been added to the subscription tree structure so
	       termination progress can be tracked through the asyncronous tasks.
	     * The pubsub_on_evsub_state callback has been split so it's not doing
	       double duty.  It now only handles the final cleanup of the
	       subscription tree.  pubsub_on_rx_refresh now handles both client
	       refreshes and client terminates.  It was always being called for
	       both anyway.
	     * The serialized_on_server_timeout task was removed since
	       serialized_pubsub_on_rx_refresh was almost identical.
	     * Missing state checks and ao2_cleanups were added.
	     * Some debug levels were adjusted to make seeing only off-nominal
	       things at level 1 and nominal or progress things at level 2+.

	  ASTERISK-26099 #close
	  Reported-by: Ross Beer.

	  Change-Id: I779d11802cf672a51392e62a74a1216596075ba1

2016-06-21 07:05 +0000 [6eb0354f2d]  Alexander Traud <pabstraud@compuserve.com>

	* res_rtp_asterisk: Use latest DTLS version available by underlying platform.

	  Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the
	  underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for
	  WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based
	  cipher-suites.

	  ASTERISK-26130 #close

	  Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0

2016-06-21 10:53 +0000 [596d0b0bc3]  Scott Griepentrog <scott@griepentrog.com>

	* PJSIP: provide transport type with received messages

	  The receipt of a SIP MESSAGE may occur over any transport including TCP
	  and TLS. When the message is received, the original URI is added to the
	  message in the field PJSIP_RECVADDR, but this is insufficient to ensure
	  a reply message can reach the originating endpoint. This patch adds the
	  PJSIP_TRANSPORT field populated with the transport type.

	  ASTERISK-26132 #close

	  Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e

2016-06-21 08:01 +0000 [9e222efbf2]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf.

	  Some configure scripts used both AC_HELP_STRING and its replacement
	  AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were
	  changed to AS_HELP_STRING.

	  ASTERISK-26046

	  Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f

2016-06-20 10:29 +0000 [e94aae00a7]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_session: Handle race condition at shutdown with timer.

	  When shutting down res_pjsip_session will get unloaded before res_pjsip.
	  The act of unloading unregisters all the PJSIP services and sets
	  their module IDs to -1. In some cases it is possible for a timer to
	  occur after this happens which calls into res_pjsip_session. The
	  res_pjsip_session module can then try to get the session from the
	  INVITE session using the module ID. Since the module ID is now -1
	  this fails.

	  This change stores a copy of the module ID and uses it for the timer
	  callback scenario. If the module ID is -1 the callback immediately
	  returns but if the module ID is valid then it continues as normal.

	  This works as the original ID of the module is guaranteed to still
	  be valid when used with the INVITE session.

	  ASTERISK-26127 #close

	  Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573

2016-06-20 12:13 +0000 [0a30008224]  Richard Mudgett <rmudgett@digium.com>

	* app_voicemail.c: Fix IMAP compile error.

	  Fix compile error introduced by the patch for
	  ASTERISK-26045

	  Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3

2016-06-17 13:51 +0000 [820ed3d4b3]  Alexei Gradinari <alex2grad@gmail.com>

	* fix: memory leaks, resource leaks, out of bounds and bugs

	  ASTERISK-26119 #close

	  Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c

2016-06-13 17:40 +0000 [11caa10cf5]  Mark Michelson <mmichelson@digium.com>

	* ARI: Ensure announcer channels are destroyed.

	  Announcer channels were not being destroyed because the
	  stasis_app_control structure that referenced them was not being
	  destroyed. The control structure was not being destroyed because it was
	  not being unlinked from its container. It was not being unlinked from
	  its container because the after bridge callback for the announcer
	  channel was not being run. The after bridge callback was not being run
	  because the after bridge datastore was not being removed from the
	  channel on destruction. The channel was not being destroyed because the
	  hangup that used to destroy the channel was now only reducing the
	  reference count to one. The reference count of the channel was only
	  being reduced to one because the stasis_app_control structure was
	  holding the final reference...

	  The control structure used to not keep a reference to the channel, so
	  that loop described above did not happen.

	  The solution is to manually remove the control structure from its
	  container when the playback on a bridge is complete.

	  ASTERISK-26083 #close
	  Reported by Joshua Colp

	  Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4

2016-06-20 08:05 +0000 [f72ffc1ff9]  Alexander Traud <pabstraud@compuserve.com>

	* http: leverage 'bindaddr' for TLS in http.conf

	  The internal HTTP/WebSocket server supports both TCP and TLS, which can be
	  activated separately via the file http.conf. The source code intends to re-use
	  the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified
	  explicitly. This did not work because of a typo. This change resolves this typo.

	  ASTERISK-26126 #close

	  Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f

2016-05-18 17:37 +0000 [3c80f84cd0]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_transport_management.c: Misc cleanups to survive shutdown.

	  * In unload_module(), reordered destroying things to minimize the window
	  that the global transports container could be used by other threads on
	  shutdown.  When shutting down you need to stop things in the opposite
	  order of creation.

	  * Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to
	  eliminate the crash potential by other threads using the container on
	  shutdown.

	  * Made struct monitored_transport.sip_received not use
	  ast_atomic_fetchadd_int() since it is used as a boolean value that is only
	  set TRUE.  It was previously incremented for every received SIP message
	  and could theoretically overflow.

	  * In monitored_transport_state_callback(), allocated the monitored
	  transport object without a lock since the lock was unused.

	  * In keepalive_global_loaded(), removed releasing the transports container
	  if the keepalive_thread could not be started.  I set it up to be tried
	  again if the user reloads the configuration.

	  Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff

2016-01-05 19:08 +0000 [7c59f2126f]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip.c: Add check that timer actually got scheduled.

	  Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1

2016-06-13 13:33 +0000 [51cc5c31c4]  Richard Mudgett <rmudgett@digium.com>

	* res_rtp_multicast.c: Fix warning message typo.

	  Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3

2016-02-11 18:15 +0000 [3d0632a9c2]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Reorganize ast_sip_session_terminate().

	  Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b

2016-06-08 06:15 +0000 [ac683f13c9]  Alexander Traud <pabstraud@compuserve.com>

	* core: Not the configured but granted number of possible file descriptors.

	  With CLI "core show settings", simply the parameter maxfiles of the file
	  asterisk.conf was shown. If that parameter was not set, nothing was displayed
	  although the environment might have set a default number itself. Or if maxfiles
	  were not granted (completely), still maxfiles was shown. Now, the maximum number
	  of possible file descriptors in the environment is shown.

	  ASTERISK-26097

	  Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b

2016-06-10 10:39 +0000 [4eb8cf2684]  Joshua Colp <jcolp@digium.com>

	* translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs.

	  This reverts commit 5bfef2a8b4674382f959b21a3b8e14cf1d942bab as it
	  caused fax test failures.

	  ASTERISK-25629

	  Change-Id: I79de974dc4f63a1cafe0d2509169fd9a6b3cbaf4

2016-06-08 06:05 +0000 [0bf1a53db3]  Alexander Traud <pabstraud@compuserve.com>

	* astfd: With RLIMIT_NOFILE only the current value is sensible.

	  With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", both the maximum max
	  and current max of possible file descriptors were shown. Both show the same
	  value always. Not to confuse users, just the current maximum is shown now.

	  ASTERISK-26097

	  Change-Id: I49cf7952d73aec9e3f6a88942842c39be18380fa

2016-06-07 18:45 +0000 [d338343dac]  Joshua Colp <jcolp@digium.com>

	* cel: Ensure only one dial status per channel exists.

	  CEL wrongly assumed that a channel would only have a single dial
	  event on it. This is incorrect. Particularly in a queue each
	  call attempt to a member will result in a dial event, adding
	  a new dial status in CEL without removing the old one. This
	  would cause the container to grow with only one dial status
	  being removed when the channel went away. The other dial status
	  entries would remain leaking memory.

	  This change fixes the memory leak by ensuring that only one dial
	  status will only ever exist for each channel.

	  The behavior during the scenario where multiple events are received
	  has also been improved. For failure cases the first failure will
	  be the dial status. If an answer dial status is received, though,
	  it will take priority and the dial status for the channel will be
	  answer.

	  Memory usage has also been decreased by storing the minimal
	  amount of information and the code has been cleaned up slightly.

	  ASTERISK-25262 #close

	  Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe

2016-06-01 13:48 +0000 [1fd3a7849e]  Mark Michelson <mmichelson@digium.com>

	* ARI: Ensure proper channel state on operations.

	  ARI was recently outfitted with operations to create and dial channels.
	  This leads to the ability to try funny stuff. You could create a channel
	  and then immediately try to play back media on it. You could create a
	  channel, dial it, and while it is ringing attempt to make it continue in
	  the dialplan.

	  This commit attempts to fix this by adding a channel state check to
	  operations that should not be able to operate on outbound channels that
	  have not yet answered. If a channel is in an invalid state, we will send
	  a 412 response.

	  ASTERISK-26047 #close
	  Reported by Mark Michelson

	  Change-Id: I2ca51bf9ef2b44a1dc5a73f2d2de35c62c37dfd8

2016-06-08 11:27 +0000 [10019dc70c]  Mark Michelson <mmichelson@digium.com>

	* test_http_media_cache: Fix failing test.

	  The retrieve_cache_control_directives test has been failing occasionally
	  in Jenkins. The apparent failure occurs when attempting to validate the
	  expiration of the retrieved file.

	  After reproducing, the problem was pretty clear. At the beginning of the
	  test, the current time is retrieved. The seconds value of this timestamp
	  is X. When the file is retrieved, res_http_media_cache calculates the
	  expiration and in doing so retrieves the current time. In most cases,
	  since the test executes quickly, it will also retrieve a timestamp with
	  X seconds. However, if the test starts very near to when the timestamp
	  seconds are set to increment, res_http_media_cache may retrieve a
	  timestamp with X+1 seconds instead.

	  The test attempted to account for this by allowing a tolerance of 1
	  second when validating the expiration. However, the problem was that the
	  comparisons being used in the validation used > and < operations. This
	  meant that values that fell within the tolerance (because they equaled
	  the upper bound of the tolerance) would fail.

	  The solution is to use >= and <= operators in the expiration validation.

	  However, I estimated that while the one second tolerance should be
	  fine on most machines, it would still be possible on a very slow machine
	  to end up falling outside the one second tolerance. So I have also
	  relaxed the tolerance of expiration validation to be three seconds
	  instead.

	  The final change here is to add a debug message when validating
	  expiration so that we can see what values are being compared.

	  ASTERISK-25959 #close
	  Reported by Joshua Colp

	  Change-Id: Ic1a0e10722c1c5d276d5a4d6a67136d6ec26c247

2016-06-03 01:20 +0000 [56bdf048d2]  Timo Teräs <timo.teras@iki.fi>

	* Add support for OGG/Speex file format

	  ASTERISK-18995 #close

	  Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a

2016-06-09 10:33 +0000 [f0855358a6]  gtjoseph <gjoseph@digium.com>

	* cdr.c: Remove assert in base_process_dial_end

	  Scenario: Caller blonde transfer
	  Bob calls Charlie who answers.
	  Bob puts Charlie on hold and calls Alice.
	  Before Alice answers, Bob transfers Charlie to Alice.

	  Charlie's channel triggers an assert because he gets an "ANSWERED"
	  event even though he never dialed anything. With recent changes to dial
	  events, this is now a valid scenario so the assert needed to be removed.

	  ASTERISK-26103 #close

	  Change-Id: I2679b517b696e7952ab7fb29403df9140e7d1de2

2016-06-09 10:37 +0000 [cdb7edbe7b]  Mark Michelson <mmichelson@digium.com>

	* chan_pjsip: Lock channel when checking for RTP changes.

	  bridge_native_rtp can call into an RTP-capable channel driver in order
	  for the driver to update information about who the channel is
	  communicating with. For SIP channel drivers, this means deactivating
	  RTCP and sending a reinvite so that the endpoints can communicate
	  directly.

	  bridge_native_rtp does the right thing and has the channel locked when
	  calling into the channel driver. chan_pjsip can't alter session
	  properties in this thread, though. chan_pjsip queues a task on the
	  session serializer in order to update properties there.

	  The problem is that this queued task was not locking the channel. This
	  meant that the queued task could attempt to deactivate RTCP at the same
	  time that the channel thread was attempting to process an incoming RTCP
	  packet. This could lead to a crash.

	  This patch fixes the issue by locking the channel in the queued task
	  when altering RTP properties.

	  ASTERISK-26092 #close
	  Reported by Niklas Larsson

	  Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159

2016-06-03 22:44 +0000 [04ec9c745e]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_registrar.c: Eliminate rx REGISTER request race condition.

	  This patch fixes a race condition processing received REGISTER requests
	  and their retransmissions caused by REGISTER requests being processed by
	  two threads.  The "sip_transaction Unable to register REGISTER transaction
	  (key exists)" message is a notable symptom of this issue.

	  This issue was more likely to happen before the pjsip/distributor
	  serializers were created.  Instead of steps one and two below placing the
	  REGISTER messages into the same pjsip/distributor they were placed in
	  random pjsip/default serializers.

	  1) REGISTER requests come in and get placed on the pjsip/distributor
	  serializer.

	  2) Before the first request is processed a retransmission comes in and is
	  placed on the same pjsip/distributor serializer.

	  3) The first request goes up the pjsip stack and is then shunted off to
	  the pjsip/aor/<aor> serializer.

	  4) Before the first request is completed processing in the pjsip/aor/<aor>
	  serializer, the second request goes up the pjsip stack and is also shunted
	  off to the pjsip/aor/<aor> serializer.

	  5) The first request completes processing and sends out its response.

	  6) The second request completes processing and tries to send out its
	  response but pjlib complains that the REGISTER transaction key already
	  exists.

	  7) Sadness ensues.

	  * The race is eliminated by removing the pjsip/aor/<aor> serializer and
	  continuing the processing in the pjsip/distributor serializer.  Now any
	  retransmissions queued in the pjsip/distributor serializer will be
	  processed after the first message is completely processed.

	  ASTERISK-26088 #close
	  Reported by:  Richard Mudgett

	  Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a

2016-06-03 11:35 +0000 [dcfef53ee2]  Richard Mudgett <rmudgett@digium.com>

	* stasis: Add setting subscription congestion levels.

	  Stasis subscriptions and message routers create taskprocessors to process
	  the event messages.  API calls are needed to be able to set the congestion
	  levels of these taskprocessors for selected subscriptions and message
	  routers.

	  * Updated CDR, CEL, and manager's stasis subscription congestion levels
	  based upon stress testing.  Increased the congestion levels to reduce the
	  potential for bursty call setup/teardown activity from triggering the
	  taskprocessor overload alert.  CDRs in particular need an extra high
	  congestion level because they can take awhile to process the stasis
	  messages.

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: Id0a716394b4eee746dd158acc63d703902450244

2016-06-02 18:19 +0000 [4879cd875c]  Richard Mudgett <rmudgett@digium.com>

	* sorcery: Add setting object type congestion levels.

	  Sorcery creates taskprocessors for object types to process object observer
	  callbacks.  An API call is needed to be able to set the congestion levels
	  of these taskprocessors for selected object types.

	  * Updated PJSIP's contact and contact_status sorcery object type observer
	  default congestion levels based upon stress testing.  Increased the
	  congestion levels to reduce the potential for bursty register/unregister
	  and subscribe/unsubscribe activity from triggering the taskprocessor
	  overload alert.

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6

2016-06-02 16:08 +0000 [2cd67d5b07]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessors: Implement high/low water mark alerts.

	  When taskprocessors get backed up, there is a good chance that we are
	  being overloaded and need to defer adding new work to the system.

	  * Implemented a high/low water alert mechanism for modules to check if the
	  system is being overloaded and take appropriate action.  When a
	  taskprocessor is created it has default congestion levels set.  A
	  taskprocessor can later have those congestion levels altered for specific
	  needs if stress testing shows that the taskprocessor is a symptom of
	  overloading or needs to handle bursty activity without triggering an
	  overload alert.

	  * Add CLI "core show taskprocessor" low/high water columns.

	  * Fixed __allocate_taskprocessor() to not use RAII_VAR().  RAII_VAR() was
	  never a good thing to use when creating a taskprocessor because of the
	  nature of how its references needed to be cleaned up on a partial
	  creation.

	  * Made res_pjsip's distributor check if the taskprocessor overload alert
	  is active before placing a message representing brand new work onto a
	  distributor serializer.

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: I182f1be603529cd665958661c4c05ff9901825fa

2016-05-27 17:31 +0000 [c966a035e0]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session: Use distributor serializer for incoming calls.

	  We must continue using the serializer that the original INVITE came in on
	  for the dialog.  There may be retransmissions already enqueued in the
	  original serializer that can result in reentrancy and message sequencing
	  problems.

	  Outgoing call legs create the pjsip/outsess/<endpoint> serializers for
	  their dialogs.

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc

2016-05-27 16:28 +0000 [5b7b16a87f]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer.

	  * Resolves potential reentrancy problems if system restarted in the middle
	  of subscription message transactions.

	  * Fixes memory leak recreating persistent subscriptions when the
	  subscription resource tree could not be created.

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be

2016-05-27 12:50 +0000 [c2ae49249c]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions.

	  We must continue using the serializer that the original SUBSCRIBE came in
	  on for the dialog.  There may be retransmissions already enqueued in the
	  original serializer that can result in reentrancy and message sequencing
	  problems.  The "sip_transaction Unable to register SUBSCRIBE transaction
	  (key exists)" message is a notable symptom of this issue.

	  Outgoing subscriptions still create the pjsip/pubsub/<endpoint>
	  serializers for their dialogs.

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0

2016-05-26 17:35 +0000 [2ff26e9746]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_distributor.c: Consistently pick a serializer for messages.

	  Incoming messages that are not part of a dialog or a recognized response
	  to one of our requests need to be sent to a consistent serializer.  Under
	  load we may be queueing retransmissions before we can process the original
	  message.  We don't need to throw these messages onto random serializers
	  and cause reentrancy and message sequencing problems.

	  * Created a pool of pjsip/distributor serializers that get picked by
	  hashing the call-id and remote tag strings of the received messages.

	  * Made ast_sip_destroy_distributor() destroy items in the reverse order of
	  creation.

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: I2ce769389fc060d9f379977f559026fbcb632407

2016-06-02 12:51 +0000 [df2791da8f]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_distributor.c: Ignore messages until fully booted.

	  We should not be processing any incoming messages until we are fully
	  booted.  We may not have dialplan or other needed configuration loaded
	  yet.

	  ASTERISK-26089 #close
	  Reported by: Scott Griepentrog

	  ASTERISK-26088
	  Reported by:  Richard Mudgett

	  Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264

2016-06-09 09:20 +0000 [d21a77b325]  gtjoseph <gjoseph@digium.com>

	* build:  Fix ast_sockaddr initialization to be more portable

	  A change to glibc 2.22 changed the order of the sockadddr_storage
	  members which caused the places where we do an initialization of
	  ast_sockaddr with '{ { 0, 0, } }' to fail compilation.  Those
	  initializers (which we shouldn't have been using anyway) have been
	  replaced with memsets.

	  Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4

2016-06-03 00:59 +0000 [72d190eb69]  Timo Teräs <timo.teras@iki.fi>

	* Detect and use proper libraries for musl toolchains

	  Change-Id: I8d9b212f70813404b82918a3f99439e500d4bfcb

2016-06-03 00:57 +0000 [39b69ab537]  Timo Teräs <timo.teras@iki.fi>

	* Fixes to include signal.h

	  POSIX defines signal.h. sys/signal.h should not be used as it is
	  c-library internal header which may or may not exist. Notably with
	  musl it generates warning of being incorrect.

	  Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc

2016-06-08 12:26 +0000 [7f5ca67e5f]  Matt Jordan <mjordan@digium.com>

	* res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded

	  A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not
	  loaded and does not have a configuration file. Previously when this
	  occurred, checks were put in to see if the configuration was loaded
	  successfully. While this is a good idea - and has been added to the
	  offending function in res_hep - the reality is res_hep_pjsip and
	  res_hep_rtcp have no business running if res_hep isn't also running.

	  As such, this patch also adds a function to res_hep that returns whether
	  or not it successfully loaded. Oddly enough, ast_module_check returns
	  "everything is peachy" even if a module declined its load - so it cannot
	  be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this
	  function to see if they should continue to load; if it fails, they
	  decline their load as well.

	  ASTERISK-26096 #close

	  Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea

2016-06-08 02:11 +0000 [784c18128b]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: No rtpmap for static RTP payload IDs in SDP.

	  This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
	  SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
	  UDP, if many codecs are allowed in Asterisk. This new feature is enabled
	  together with the optional feature compactheaders=yes via the file sip.conf.

	  ASTERISK-25578 #close

	  Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044

2016-06-02 12:04 +0000 [31a5c28339]  Joshua Colp <jcolp@digium.com>

	* res_odbc: Implement a connection pool.

	  Testing has shown that our usage of UnixODBC is problematic
	  due to bugs within UnixODBC itself as well as the heavy weight
	  cost of connecting and disconnecting database connections, even
	  when pooling is enabled.

	  For users of UnixODBC 2.3.1 and earlier crashes would occur due
	  to insufficient protection of the disconnect operation. This was
	  fixed in UnixODBC 2.3.2 and above.

	  For users of UnixODBC 2.3.3 and higher a slow-down would occur
	  under heavy database use due to repeated connection establishment.
	  A regression is present where on each connection the database
	  configuration is cached again, with the cache growing out of
	  control.

	  The connection pool implementation present in this change helps
	  to mitigate these issues by reducing how much we connect and
	  disconnect database connections. We also solve the issue of
	  crashes under UnixODBC 2.3.1 by defaulting the maximum number of
	  connections to 1, returning us to the previous working behavior.
	  For users who may have a fixed version the maximum concurrent
	  connection limit can be increased helping with performance.

	  The connection pool works by keeping a list of active connections.
	  If the connection limit has not been reached a new connection is
	  established. If the connection limit has been reached then the
	  request waits until a connection becomes available before
	  continuing.

	  ASTERISK-26074 #close
	  ASTERISK-26054 #close

	  Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff

2016-05-31 09:10 +0000 [80ff7912a1]  Vasil Kolev <vasil.kolev@securax.org>

	* chan_sip: bigger buffers for headers, better failure mode

	  Currently chan_sip can give weird messages if the contacts don't
	  fit in the From: or To: headers. This fix changes the from,to and
	  invite variables to use ast_str, allocates and deallocates them and
	  resizes them if needed.

	  ASTERISK-26069 #close

	  Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3

2016-06-06 11:13 +0000 [60caebc738]  Örn Arnarson <orn@arnarson.net>

	* apps/app_voicemail.c and main/say.c: Add support for Icelandic language

	  Icelandic has some weird grammar rules when dealing with dates and
	  numbers. There are different genders used depending on which number
	  you're dealing with, and only a handful of numbers do change depending
	  on the gender. There is also an implied gender in several cases.

	  This patch was originally written for asterisk 1.6, and has been in use
	  for several years without crashes. I cleaned it up a bit and rewrote
	  what was necessary for Asterisk 13.

	  The functions were copied from other similar languages and modified
	  where appropriate. If i recall correctly, the German and Danish
	  functions were used as a base.

	  ASTERISK-26087
	  Reported by: Örn Arnarson
	  Tested by: Örn Arnarson

	  Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665

2016-06-07 05:45 +0000 [52120204c9]  Alexander Traud <pabstraud@compuserve.com>

	* res_srtp: Instead of libSRTP use OpenSSL as random source.

	  Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore.
	  Therefore, the symbol RAND_bytes is used instead of crypto_get_random.

	  ASTERISK-24436 #close

	  Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96

2016-06-07 02:16 +0000 [da943ec5c0]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Avoid 'ar cru' and use 'ar cr' instead.

	  In several internal library projects, the files are archived with the help of
	  'ar cr'. Only the projects editline and the Objective Open H.323 stack
	  implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms
	  changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier
	  ignored since `D' is the default (see `U')". For consistency and to avoid this
	  message all projects use 'ar cr' now.

	  ASTERISK-26091 #close

	  Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40

2016-06-01 16:57 +0000 [dca052e531]  Richard Mudgett <rmudgett@digium.com>

	* chan_rtp.c: Simplify options to UnicastRTP channel creation.

	  Change the awkward and not as flexible UnicastRTP options format
	  From:
	  Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
	  To:
	  Dial(UnicastRTP/127.0.0.1[/[<options>]])

	  Where <options> can be standard Asterisk flag options:
	  c(<codec>) - Specify which codec/format to use such as 'ulaw'.
	  e(<engine>) - Specify which RTP engine to use such as 'asterisk'.

	  More option flags can be easily added later such as the codec's RTP
	  payload type to use when the codec does not have a static payload type
	  defined.

	  Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9

2016-05-02 05:57 +0000 [5bfef2a8b4]  Jaco Kroon <jaco@uls.co.za>

	* translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs.

	  ASTERISK-25629 #close

	  Change-Id: Ibfcf0670e094e9718d82fd9920f1fb2dae122006

2016-05-25 10:34 +0000 [3e8d523d88]  Alexei Gradinari <alex2grad@gmail.com>

	* core/dial: New channel variable FORWARDERNAME

	  Added a new channel variable FORWARDERNAME which indicates which
	  channel was responsible for a forwarding requests received on dial attempt.

	  Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.

	  ASTERISK-26059 #close

	  Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2

2016-05-27 14:49 +0000 [a2f820e8dc]  gtjoseph <gjoseph@digium.com>

	* ari/resource_channels:  Add 'formats' to channel create/originate

	  If you create a local channel and don't specify an originator channel
	  to take capabilities from, we automatically add all audio formats to
	  the new channel's capabilities. When we try to make the channel
	  compatible with another, the "best format" functions pick the best
	  format available, which in this case will be slin192.  While this is
	  great for preserving quality, it's the worst for performance and
	  overkill for the vast majority of applications.

	  In the absense of any other information, adding all formats is the
	  correct thing to do and it's not always possible to supply an
	  originator so a new parameter 'formats' has been added to the channel
	  create/originate functions. It's just a comma separated list of formats
	  to make availalble for the channel. Example: "ulaw,slin,slin16".
	  'formats' and 'originator' are mutually exclusive.

	  To facilitate determination of format names, the format name has been
	  added to "core show codecs".

	  ASTERISK-26070 #close

	  Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b

2016-06-03 01:33 +0000 [538c6415c6]  Timo Teräs <timo.teras@iki.fi>

	* chan_sip: Support auth username for callbackextension feature

	  ASTERISK-20527 #close

	  Change-Id: I659cf7f00836a09d09d146ad226a40477d731239

2016-06-03 00:39 +0000 [797695c5cc]  Timo Teräs <timo.teras@iki.fi>

	* Make use of GLOB_BRACE and GLOB_NOMAGIC optional

	  These flags are non-portable GNU extensions. Make their use
	  optional. This fixes complication error on e.g. musl c-library
	  based systems.

	  Change-Id: I0aa06efc62aa8995f091445c8b762a75a91042f3

2016-06-02 14:57 +0000 [3c1fec8099]  Timo Teräs <timo.teras@iki.fi>

	* Fix res_search usage

	  Resolver state is not part of res_search API. This fixes
	  compilation error:

	  dns.c:261:8: error: too many arguments to function 'res_search'
	    ret = res_search(&dns_state,

	  Change-Id: Ia600a58557040df83f744da3dde23225293845a5

2016-06-02 14:53 +0000 [9c1d95e873]  Timo Teräs <timo.teras@iki.fi>

	* Fix #include poll.h and sys/cdefs.h

	  POSIX defines poll.h, sys/poll.h should not be used at is c-library
	  internal header which may or may not exist. Notable in musl it
	  generates warning of being incorrect. And add explict include of
	  sys/cdefs.h where needed.

	  Change-Id: I142930df53fe7585a06b854b6faddc5301e024be

2016-05-25 08:45 +0000 [8a5c2e736c]  Niklas Larsson <niklas@tese.se>

	* core/manager: Add uptime field to FullyBooted

	  Add Uptime and LastReload to event FullyBooted.

	  ASTERISK-26058 #close
	  Reported by: Niklas Larsson

	  Change-Id: I909b330801c0990d78df9b272ab0adc95aecb15e

2016-06-02 04:59 +0000 [4505a59dc9]  Joshua Colp <jcolp@digium.com>

	* alembic: Fix migration.

	  The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting
	  to use UniqueConstraint and failing. It was not imported and after
	  importing it also continued to fail.

	  I've changed the script to use the explicit name of the constraint
	  instead.

	  Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9

2016-06-01 13:57 +0000 [40d19f2e55]  Richard Mudgett <rmudgett@digium.com>

	* logging,cdr,cel: Fix stringfield memory leak.

	  The stringfields refactor to allow adding stringfields to the end of a
	  structure (f6f4cf459f43f072604927209b39646f84aaa2e2) exposed some
	  incomplete cleanup code by some stringfield users.

	  The most noticeable leaker is the logging system where there is a leak for
	  every log message generated.

	  ASTERISK-26078 #close
	  Reported by:  Etienne Lessard
	  Patches:
	        jira_asterisk_26078_v13.patch (license #5621) patch uploaded
	        by Richard Mudgett

	  Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782

2016-05-31 13:02 +0000 [aec7916595]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_distributor.c: Use correct rdata info access method (Part 2).

	  The pjproject doxygen for rdata->msg_info.info says to call
	  pjsip_rx_data_get_info() instead of accessing the struct member directly.
	  You need to call the function mostly because the function will generate
	  the struct member value if it is not already setup.

	  Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799

2016-05-09 15:00 +0000 [205a31f86c]  Mark Michelson <mmichelson@digium.com>

	* Expand the scope of Dial Events

	  Dial events up to this point have come in two flavors
	  * A Dial event with no status to indicate that dialing has begun
	  * A Dial event with a status to indicate that dialing has ended

	  With this change, Dial events have been expanded to also give
	  intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS".
	  This is especially useful for ARI dialing, as it gives the application
	  writer the opportunity to place a channel into an early bridge when
	  early media is detected.

	  AMI handles these in-progress dial events by sending a new event called
	  "DialState" that simply indicates that dial state has changed but has
	  not ended. ARI never distinguished between DialBegin and DialEnd, so no
	  change was made to the event itself.

	  Another change here relates to dial forwards. A forward-related event
	  was previously only sent when a channel was successfully able to forward
	  a call to a new channel. With this set of changes, if forwarding is
	  blocked, we send a Dial event with a forwarding destination but no
	  forwarding channel, since we were prevented from creating one. This is
	  again useful for ARI since application writers can now handle call
	  forward attempts from within their own application.

	  ASTERISK-25925 #close
	  Reported by Mark Michelson

	  Change-Id: I42cbec7730d84640a434d143a0d172a740995543

2016-05-30 19:27 +0000 [8a6a14590d]  gtjoseph <gjoseph@digium.com>

	* res_pjsip_mwi_body_generator:  Re-order the body items

	  Re-ordered the body items so Message-Account is second.

	  Messages-Waiting: no
	  Message-Account: sip:1571@<IP Removed>:5060
	  Voice-Message: 0/0 (0/0)

	  ASTERISK-26065 #close
	  Reported-by: Ross Beer

	  Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3

2016-05-30 10:58 +0000 [7fa5766752]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Move to pjproject 2.5

	  Although all the patches we had against 2.4.5 were applied by Teluu,
	  a new bug was introduced preventing re-use of tcp and tls transports
	  This patch removes all the previous patches against 2.4.5, updates
	  the version to 2.5, and adds a new patch to correct the transport
	  re-use problem.

	  Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068

2016-05-27 12:25 +0000 [b56f611856]  Rusty Newton <rnewton@digium.com>

	* res_pjsip: Add clarifying documentation to PJSIP_HEADER help text

	  Added notes about when you can read or write headers. Specifically
	  about being able to read on the inbound channel and write on an
	  outbound channel.

	  ASTERISK-26063 #close
	  Reported by: Private Name
	  Tested by: Rusty Newton

	  Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5

2016-05-26 15:14 +0000 [bb0f4a6310]  Mark Michelson <mmichelson@digium.com>

	* multicast RTP: Add dialing options

	  This adds a new parameter to the end of a multicast RTP dialing string.
	  This parameter defines the following options:

	  * i: Set the interface from which multicast RTP is sent
	  * l: Set whether multicast packets are looped back to the sender
	  * t: Set the TTL for multicast packets
	  * c: Set the codec to use for RTP

	  ASTERISK-26068 #close
	  Reported by Mark Michelson

	  Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219

2016-05-09 14:48 +0000 [88d997913f]  Mark Michelson <mmichelson@digium.com>

	* ARI: Re-implement the ARI dial command, allowing for early bridging.

	  ARI dial had been implemented using the Dial API. This made great sense
	  when dialing was 100% separate from bridging. However, if a channel were
	  to be added to a bridge during the dial attempt, there would be a
	  conflict between the dialing thread and the bridging thread. Each would
	  be attempting to read frames from the dialed channel and act on them.

	  The initial attempt to make the two play nice was to have the Dial API
	  suspend the channel in the bridge and stay in charge of the channel
	  until the dial was complete. The problem with this was that it was
	  riddled with potential race conditions. It also was not well-suited for
	  the case where the channel changed which bridge it was in during the
	  dial.

	  This new approach removes the use of the Dial API altogether. Instead,
	  the channel we are dialing is placed into an invisible ARI dialing
	  bridge. The bridge channel thread handles incoming frames from the
	  channel. If the channel is added to a real bridge, it is departed from
	  the invisible bridge and then added to the real bridge. Similarly, if
	  the channel is removed from the real bridge, it is automatically added
	  back to the invisible bridge if the dial attempt is still active.

	  This approach keeps the threading simple by always having the channel
	  being handled by bridge channel threads.

	  ASTERISK-25925

	  Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb

2016-05-19 14:56 +0000 [31f17abe44]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: add "via_addr", "via_port", "call_id" to contact

	  As res_pjsip_nat rewrites contact's address, only the last Via header
	  can contain the source address of registered endpoint.
	  Also Call-Id header may contain the source address of registered
	  endpoint.

	  Added "via_addr", "via_port", "call_id" to contact.
	  Added new fields ViaAddress, CallID to AMI event ContactStatus.

	  ASTERISK-26011

	  Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576

2016-05-24 16:56 +0000 [574c9e77eb]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: chatty verbose messages

	  There are a lot of verbose messages about Endpoint and Contact status
	  changes if there are many dynamic endpoints.
	  The patch sets verbose level 2 for Endpoint status changes
	  and verbose level 3 for Contact status changes.

	  ASTERISK-26055 #close

	  Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7

2016-05-20 13:56 +0000 [b3142e99e4]  Alexei Gradinari <alex2grad@gmail.com>

	* app_voicemail: fix bugs, imap mm_status log change to debug

	  Fixed some bugs:
	  - create dirpath when save downloading message from IMAP storage.
	  - create IMAP folder if not exists when saving to IMAP storage
	  - check if file successfully opened before write to it
	  - some IMAP checks
	  - remove non-standard flag 'Unseen'
	  etc

	  Change to debug IMAP mm_status log instead of verbose.

	  Remove unused X-Asterisk-VM-Caller-channel message header
	  for security reason. The clients should not know name of peer/endpoint.

	  ASTERISK-26045 #close

	  Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b

2016-05-25 18:30 +0000 [7d44d12816]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_distributor.c: Use correct rdata info access method.

	  The pjproject doxygen for rdata->msg_info.info says to call
	  pjsip_rx_data_get_info() instead of accessing the struct member directly.
	  You need to call the function mostly because the function will generate
	  the struct member value if it is not already setup.

	  Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2

2016-05-03 11:11 +0000 [1d60bfcdf1]  Tzafrir Cohen <tzafrir@debian.org>

	* followme: allow disabling callee prompt

	  Add the option 'enable_callee_prompt' to followme.conf. Enabled by
	  default. If disabled, a callee is not prompted to accept or reject
	  the forwarded call.

	  ASTERISK-26064 #close

	  Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5
	  Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

2016-02-12 09:59 +0000 [80ff2c2540]  Corey Farrell <git@cfware.com>

	* threadpool: Fix potential data race.

	  worker_start checked for ZOMBIE status without holding a lock.  All
	  other read/write of worker status are performed with a lock, so this
	  check should do the same.

	  ASTERISK-25777 #close

	  Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781

2016-05-24 05:28 +0000 [070eab6ed2]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_outbound_publish: Ensure publish is valid when explicitly destroying.

	  Recent changes to res_pjsip_outbound_publish have introduced a
	  race condition at shutdown where an outbound publish may be shutdown
	  twice. In this case the first succeeds as a result of the unpublish.
	  In the second invocation since it's been unpublished a task is
	  queued to just destroy the client. This task holds no ref to the
	  publish and as a result the publish may be destroyed before the
	  task is run, causing a crash.

	  This explicit destruction task now holds a reference to the publish
	  to ensure it remains valid.

	  ASTERISK-26053 #close

	  Change-Id: I10789b98add3e50292ee3b33a55a1d9061cec94b

2016-05-09 14:27 +0000 [f6c33771f6]  Mark Michelson <mmichelson@digium.com>

	* Bridging: introduce "invisible" bridges.

	  Invisible bridges function the same as normal bridges, but they have the
	  following restrictions:

	  * They never show up in CLI, AMI, or ARI queries.
	  * They do not have Stasis messages published about them.

	  Invisible bridges' main use is for when use of the bridging system is
	  desired, but the bridge should not be known to users of the Asterisk
	  system.

	  ASTERISK-25925

	  Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670

2016-05-22 11:03 +0000 [85d0272e76]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Only check transaction on transaction state events.

	  The send request callback function currently assumes that it
	  will only ever be called on transaction state changes. This is
	  not always true. If our own timer callback occurs we will call
	  the callback with a timer event instead of a transaction state
	  change event. In this case the transaction on the event is
	  invalid and accessing it will result in a crash.

	  ASTERISK-26049 #close

	  Change-Id: I623211c8533eb73056b0250b4580b49ad4174dfc

2016-05-21 05:42 +0000 [31897d2d99]  Jesper (License 5518)

	* func_curl: Don't trim response text on non-ASCII characters

	  The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of
	  a signed comparison.

	  ASTERISK-25669 #close
	  Reported by: Jesper
	  patches:
	    strings.curl.trim.patch submitted by Jesper (License 5518)

	  Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a

2016-05-20 19:03 +0000 [2a77af9ed0]  Richard Mudgett <rmudgett@digium.com>

	* chan_rtp.c: Cleanup ast_request() parameter parsing.

	  * Fixed NULL crash potential if parameters are missing.

	  * Reordered some operations so further diagnostic messages can be
	  more helpful.

	  Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70

2016-05-20 16:59 +0000 [ade5275a3e]  Richard Mudgett <rmudgett@digium.com>

	* parking.h: Update ast_parking_park_call() doxygen to reality.

	  ASTERISK-26029

	  Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260

2016-05-12 15:18 +0000 [c378b00a83]  Alexei Gradinari <alex2grad@gmail.com>

	* func_odbc: single database connection should be optional

	  func_odbc was changed in Asterisk 13.9.0
	  to make func_odbc use a single database connection per DSN
	  because of reported bug ASTERISK-25938
	  with MySQL/MariaDB LAST_INSERT_ID().

	  This is drawback in performance when func_odbc is used
	  very often in dialplan.

	  Single database connection should be optional.

	  ASTERISK-26010

	  Change-Id: I7091783a7150252de8eeb455115bd00514dfe843

2016-05-20 09:39 +0000 [1c02b19b79]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Match dialogs on responses better.

	  When receiving an incoming response to a dialog-starting INVITE, we were
	  not matching the response to the INVITE dialog. Since we had not
	  recorded the to-tag to the dialog structure, the PJSIP-provided method
	  to find the dialog did not match.

	  Most of the time, this was not a problem, because there is a fall-back
	  that makes the response get routed to the same serializer that the
	  request was sent on. However, in cases where an asynchronous DNS lookup
	  occurs in the PJSIP core, the thread that sends the INVITE is not
	  actually a threadpool serializer thread. This means we are unable to
	  record a serializer to handle the incoming response.

	  Now, imagine what happens when an INVITE is sent on a non-serialized
	  thread, and an error response (such as a 486) arrives. The 486 ends up
	  getting put on some random threadpool thread. Eventually, a hangup task
	  gets queued on the INVITE dialog serializer. Since the 486 is being
	  handled on a different thread, the hangup task can execute at the same
	  time that the 486 is being handled. The hangup task assumes that it is
	  the sole owner of the INVITE session and channel, so it ends up
	  potentially freeing the channel and NULLing the session's channel
	  pointer. The thread handling the 486 can crash as a result.

	  This change has the incoming response match the INVITE transaction, and
	  then get the dialog from that transaction. It's the same method we had
	  been using for matching incoming CANCEL requests. By doing this, we get
	  the INVITE dialog and can ensure that the 486 response ends up being
	  handled by the same thread as the hangup, ensuring that the hangup runs
	  after the 486 has been completely handled.

	  ASTERISK-25941 #close
	  Reported by Javier Riveros

	  Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0

2016-05-18 06:19 +0000 [e773e3a9bb]  Matt Jordan <mjordan@digium.com>

	* ARI: Add the ability to download the media associated with a stored recording

	  This patch adds a new feature to ARI that allows a client to download
	  the media associated with a stored recording. The new route is
	  /recordings/stored/{name}/file, and transmits the underlying binary file
	  using Asterisk's HTTP server's underlying file transfer facilities.

	  Because this REST route returns non-JSON, a few small enhancements had
	  to be made to the Python Swagger generation code, as well as the
	  mustache templates that generate the ARI bindings.

	  ASTERISK-26042 #close

	  Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181

2016-05-19 11:41 +0000 [40cb032009]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_astdb: Filter fields to only the registered ones.

	  This change introduces the same filtering that is done in res_sorcery_realtime
	  to the res_sorcery_astdb module. This allows persisted sorcery objects
	  that may contain unknown fields to still be read in from the AstDB
	  and used. This is particularly useful when switching between different
	  versions of Asterisk that may have introduced additional fields.

	  ASTERISK-26014 #close

	  Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2

2016-05-09 21:40 +0000 [9766a12b4c]  snuffy <snuffy22@gmail.com>

	* res_pjsip_empty_info: Respond to empty SIP INFO packets

	  Some SBCs require responses to empty SIP INFO packets
	  after establishing call via INVITE, if not responded to
	  they may drop your call after unspecified timeout of X minutes.

	  They are identified by having no Content-Type, check for this
	  and respond with 200 - OK message.

	  ASTERISK-24986 #close
	  Reported-by: Ilya Trikoz, Federico Santulli

	  Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0

2016-05-18 10:58 +0000 [111c4b0324]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* Makefile: remove OSARCH check for init install

	  There are more specific checks for the platform.

	  Specifically this allows installing OS/X init scripts.

	  ASTERISK-26038 #close

	  Change-Id: If08933621145b10362a0cfe73c079301d9c13f50
	  Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

2016-05-10 11:28 +0000 [d4b77dad1b]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_exten_state: Use the extension for publishing to.

	  This change uses the newly added multi-user support for
	  outbound publish to publish to the specific user that an
	  extension state change is for.

	  This also extends the res_pjsip_outbound_publish support
	  to include the user specific From and To URI information in
	  the outbound publishing of extension state. Since the URI
	  is used when constructing the body it is important to ensure
	  that the correct local and remote URIs are used.

	  Finally the max string growths for the dialog-info+xml
	  body generator has been increased as through testing it has
	  proven to be too conservative.

	  ASTERISK-25965

	  Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1

2016-05-03 16:07 +0000 [3905997bae]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publish: Add multi-user support per configuration

	  Added a new multi_user option that when specified allows a particular
	  configuration to be used for multiple users. It does this by replacing
	  the user portion of the server uri with a dynamically created one.

	  Two new API calls have been added in order to make use of the new
	  functionality:

	  ast_sip_publish_user_send - Sends an outgoing publish message based on the
	  given user. If state for the user already exists it uses that, otherwise
	  it dynamically creates new outbound publishing state for the user at that
	  time.

	  ast_sip_publish_user_remove - Removes all outbound publish state objects
	  associated with the user. This essentially stops outbound publishing for
	  the user.

	  ASTERISK-25965 #close

	  Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc

2016-05-18 07:54 +0000 [6e5e84458f]  gtjoseph <gjoseph@digium.com>

	* udptl:  Don't eat sequence numbers until OK is received

	  Scenario:
	  Local fax -> Asterisk w/ firewall -> Provider -> Remote fax

	  * Local fax starts rtp call to remote fax
	  * Remote fax starts t38 call back to local fax.
	  * Local fax sends t38 no-signal to Asterisk before sending an OK.
	  * udptl processes the frame and increments the expected sequence number.
	  * chan_sip drops the frame because the call isn't up so nothing goes out
	    the external interface to open the port for incoming packets.
	  * Local fax sends OK and Asterisk sends OK to the remote fax.
	  * Remote fax sends t38 packets which are dropped by the firewall.
	  * Local fax re-sends t38 no-signal with the same sequence number.
	  * udptl drops the frame because it thinks it's a dup.
	  * Still no outgoing packets to open the firewall.
	  * t38 negotiation fails.

	  The patch drops frames t38 received before udptl sequence processing
	  when the call hasn't been answered yet.  The second no-signal frame
	  is then seen as new and is relayed out the external interface which
	  opens the port and allows negotiation to continue.

	  ASTERISK-26034 #close

	  Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9

2016-05-15 12:22 +0000 [52148d93f4]  Matt Jordan <mjordan@digium.com>

	* CHANGES: Update formatting of items

	  * Provide consistent indenting of lines in bulleted paragraphs
	  * Respect the 80 character column width
	  * Group all like items together, e.g., all dialplan applications under
	    "Applications", etc.
	  * Use a single blank line to break up functionality changes within a
	    larger section
	  * Use two blanks lines to delineate larger sections

	  Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647

2016-04-18 18:17 +0000 [03d88b5656]  Matt Jordan <mjordan@digium.com>

	* ARI: Add the ability to play multiple media URIs in a single operation

	  Many ARI applications will want to play multiple media files in a row to
	  a resource. The most common use case is when building long-ish IVR prompts
	  made up of multiple, smaller sound files. Today, that requires building a
	  small state machine, listening for each PlaybackFinished event, and triggering
	  the next sound file to play. While not especially challenging, it is tedious
	  work. Since requiring developers to write tedious code to do normal activities
	  stinks, this patch adds the ability to play back a list of media files to a
	  resource.

	  Each of the 'play' operations on supported resources (channels and bridges)
	  now accepts a comma delineated list of media URIs to play. A single Playback
	  resource is created as a handle to the entire list. The operation of playing
	  a list is identical to playing a single media URI, save that a new event,
	  PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final
	  media URI. When the entire list is finished being played, a PlaybackFinished
	  event is raised.

	  In order to help inform applications where they are in the list playback, the
	  Playback resource now includes a new, optional attribute, 'next_media_uri',
	  that contains the next URI in the list to be played.

	  It's important to note the following:
	   - If an offset is provided to the 'play' operations, it only applies to the
	     first media URI, as it would be weird to skip n seconds forward in every
	     media resource.
	   - Operations that control the position of the media only affect the current
	     media being played. For example, once a media resource in the list
	     completes, a 'reverse' operation on a subsequent media resource will not
	     start a previously completed media resource at the appropiate offset.
	   - This patch does not add any new operations to control the list. Hopefully,
	     user feedback and/or future patches would add that if people want it.

	  ASTERISK-26022 #close

	  Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f

2016-05-17 11:14 +0000 [5bd1bf2816]  gtjoseph <gjoseph@digium.com>

	* chan_sip:  Prevent extra Session-Expires headers from being added

	  When chan_sip does a re-INVITE to refresh a session and authentication
	  is required, the INVITE with the Authorization header containes a
	  second Session-Expires header without the ";refersher=" parameter.
	  This is causing some proxies to return a 400.  Also, when Asterisk is
	  the uas and the refresher, it is including the Session-Expires and
	  Min-SE headers in OPTIONS messages which is not allowed per RFC4028.

	  This patch (based on the reporter's) Checks to see if a Session-Expires
	  header is already in the message before adding another one.  It also
	  checks that the method is INVITE or UPDATE.

	  ASTERISK-26030 #close

	  Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9

2016-05-16 15:29 +0000 [ae81b55361]  gtjoseph <gjoseph@digium.com>

	* res_pjsip_outbound_registration:  Clean up state when registration is deleted

	  Nothing was cleaning up the registration state object when ast_sorcery_delete
	  was called on a registration.  So, the registration was deleted from sorcery
	  but the state object went right on refreshing the registration (or failing
	  to refresh the registration) with the peer.

	  * Added a 'deleted' observer on registration that removes the state object.

	  ASTERISK-25964 #close
	  Reported-by Matt Jordan

	  Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23

2016-05-15 19:05 +0000 [8b5cee4a4f]  gtjoseph <gjoseph@digium.com>

	* res_pjsip:  Set TCP_NODELAY on TCP transports

	  Although it's perfectly legal to place multiple SIP messages in the same packet,
	  it can cause problems because the Linux default is to enable Path MTU Discovery
	  which sets the Don't Fragment bit on the packets. If adding a second message to
	  the packet causes the MTU to be exceeded, and the destination isn't equipped to
	  send a FRAGMENTATION NEEDED response to a large packet, the packet will just be
	  dropped.

	  We can't specifically tell the stack to send only 1 message per packet, but we
	  can turn on TCP_NODELAY when we create the transport. This will at least tell
	  the stack to send packets as soon as possible.

	  ASTERISK-26005 #close
	  Reported-by: Ross Beer

	  Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd

2016-05-14 07:24 +0000 [3522376512]  Matt Jordan <mjordan@digium.com>

	* logger: Support JSON logging with Verbose messages

	  When 2d7a4a3357 was merged, it missed the fact that Verbose log messages
	  are formatted and handled by 'verbosers'. Verbosers are registered
	  functions that handle verbose messages only; they exist as a separate
	  class of callbacks. This was done to handle the 'magic' that must be
	  inserted into Verbose messages sent to remote consoles, so that the
	  consoles can format the messages correctly, i.e., the leading
	  tabs/characters.

	  In reality, verbosers are a weird appendage: they're a separate class of
	  formatters/message handlers outside of what handles all other log
	  messages in Asterisk. After some code inspection, it became clear that
	  simply passing a Verbose message along with its 'sublevel' importance
	  through the normal logging mechanisms removes the need for verbosers
	  altogether.

	  This patch removes the verbosers, and makes the default log formatter
	  aware that, if the log channel is a console log, it should simply insert
	  the 'verbose magic' into the log messages itself. This allows the
	  console handlers to interpret and format the verbose message
	  themselves.

	  This simplifies the code quite a lot, and should improve the performance
	  of printing verbose messages by a reasonable factor:
	  (1) It removes a number of memory allocations that were done on each
	      verobse message
	  (2) It removes the need to strip the verbose magic out of the verbose
	      log messages before passing them to non-console log channels
	  (3) It now performs fewer iterations over lists when handling verbose
	      messages

	  Since verbose messages are now handled like other log messages (for the
	  most part), the JSON formatting of the messages works as well.

	  ASTERISK-25425

	  Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96

2016-05-14 21:48 +0000 [a1803cb5f4]  Matt Jordan <mjordan@digium.com>

	* configs/samples/pjsip.conf.sample: Fix typo

	  A ':' is not a valid token for starting a comment.

	  Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad

2016-05-12 07:08 +0000 [d29c17834c]  Matt Jordan <mjordan@digium.com>

	* res/res_hep_pjsip: Fix reported local IP address when bound to 'any'

	  When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its
	  local address the 'any' address, as opposed to the IP address we
	  actually received the packet on. This can cause some confusion in Homer,
	  as it will dutifully report what we send it.

	  This patch uses the PJSIP inspection routines to determine which IP
	  address we probably received the packet on based on the remote party's
	  IP address. In the event that this fails, it falls back to the IP
	  address natively reported by the transport.

	  Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3

2016-05-14 12:29 +0000 [14938184a3]  Sean Bright <sean.bright@gmail.com>

	* res_ari: Correct Location headers returned by some ARI resources

	  The Location headers returned by:

	   * /bridges/{bridgeId}/play
	   * /bridges/{bridgeId}/record
	   * /channels/{channelId}/play
	   * /channels/{channelId}/record

	  Did not have the '/ari' prefix, and in the case of the 'play' resources, were
	  using 'playback' instead of 'playbacks.'

	  Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c

2016-05-11 20:17 +0000 [e06a23681c]  Matt Jordan <mjordan@digium.com>

	* res_hep: Provide an option to pick the UUID type

	  At one point in time, it seemed like a good idea to use the Asterisk
	  channel name as the HEP correlation UUID. In particular, it felt like
	  this would be a useful identifier to tie PJSIP messages and RTCP
	  messages together, along with whatever other data we may eventually send
	  to Homer. This also had the benefit of keeping the correlation UUID
	  channel technology agnostic.

	  In practice, it isn't as useful as hoped, for two reasons:
	  1) The first INVITE request received doesn't have a channel. As a
	     result, there is always an 'odd message out', leading it to be
	     potentially uncorrelated in Homer.
	  2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
	     This causes RTCP information to be uncorrelated to the SIP message
	     traffic seen by those capture nodes.

	  In order to support both (in case someone is trying to use res_hep_rtcp
	  with a non-PJSIP channel), this patch adds a new option, uuid_type, with
	  two valid values - 'call-id' and 'channel'. The uuid_type option is used
	  by a module to determine the preferred UUID type. When available, that
	  source of a correlation UUID is used; when not, the more readily available
	  source is used.

	  For res_hep_pjsip:
	   - uuid_type = call-id: the module uses the SIP Call-ID header value
	   - uuid_type = channel: the module uses the channel name if available,
	                          falling back to SIP Call-ID if not
	  For res_hep_rtcp:
	   - uuid_type = call-id: the module uses the SIP Call-ID header if the
	                          channel type is PJSIP and we have a channel,
	                          falling back to the Stasis event provided
	                          channel name if not
	   - uuid_type = channel: the module uses the channel name

	  ASTERISK-25352 #close

	  Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c

2016-05-13 11:46 +0000 [69a85a519f]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: Endpoint IP Access Controls

	  With the old SIP module we can use IP access controls per peer.
	  PJSIP module missing this feature.

	  This patch added next configuration Endpoint options:
	      "acl" - list of IP ACL section names in acl.conf
	      "deny" - List of IP addresses to deny access from
	      "permit" - List of IP addresses to permit access from
	      "contact_acl" - List of Contact ACL section names in acl.conf
	      "contact_deny" - List of Contact header addresses to deny
	      "contact_permit" - List of Contact header addresses to permit

	  This patch also better logging failed request:
	      add custom message instead of "No matching endpoint found"
	      add SIP method to logging

	  ASTERISK-25900

	  Change-Id: I456dea3909d929d413864fb347d28578415ebf02

2016-05-12 14:36 +0000 [fd3f70598d]  Mark Michelson <mmichelson@digium.com>

	* Use doubles instead of floats for conversions when comparing strings.

	  In 13.9.0, there was an issue where PJSIP contacts added to an AOR would
	  be deleted at seemingly random times.

	  One reason this was happening was because of an operation to retrieve
	  the contacts whose expiration time was less than or equal to the current
	  time. When retrieving existing contacts, the contact's expiration time
	  and the current time were converted from a string to a float, and those
	  two floats were compared.

	  On some systems, including mine, this conversion was horribly off. For
	  instance, I could regularly see the string "1463079214" get converted
	  into 1463079168.000000. When switching from using a float to using a
	  double, the conversion was as expected.

	  Why was the conversion to float off? My best guess is that the
	  conversion to float was attempting to store the entire value in the 23
	  bit significand of the IEEE-754 floating point number. In particular, if
	  you take only the 23 most significant bits of 1463079214, you get the
	  messed up 1463079168 that we were seeing in the conversion. It likely
	  was possible to get a more precise value by composing the number using
	  an exponent, but the conversion did not work that way. With a double,
	  you have a 52 bit significand, allowing the entire value to fit there,
	  and thereby allowing an accurate conversion.

	  ASTERISK-26007 #close
	  Reported by Greg Siemon

	  Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070

2016-05-12 09:13 +0000 [4f8cfa0220]  gtjoseph <gjoseph@digium.com>

	* pjsip_distributor:  Add missing newline to NOTICE

	  There was a newline missing from the end of the "no matching endpoint" notice.

	  Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181

2016-05-10 10:19 +0000 [d14d1ba826]  Sebastian Damm <damm@sipgate.de>

	* res_pjsip_outbound_registration: generate correct Contact URI for TLS

	  There are two types of SIP URIs indicating a secure transport:
	  * sips:user@example.org
	  * sip:user@example.org;transport=tls

	  When using a sips URI, Asterisk checks incoming INVITEs and answers from
	  the other side for sips URIs, and rejects the packet if there are only
	  sip URIs. So Asterisk should only generate a sips Contact URI if the
	  other side supports it.

	  This patch makes Asterisk generate either a sip or sips Contact URI
	  depending on the format of the server URI.

	  If you want a sip URI, use:
	  server_uri=sip:example.org\;transport=tls

	  If you want a sips URI, use:
	  server_uri=sips:example.org

	  ASTERISK-25990 #close
	  Reported-by: Sebastian Damm

	  Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2

2016-05-05 16:41 +0000 [9f996624b0]  Alexei Gradinari <alex2grad@gmail.com>

	* logger: Add PID to syslog messages.

	  During refactoring of this support the addition of
	  the PID to messages was removed. This change adds it
	  back in.

	  ASTERISK-25538 #close

	  Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36

2016-05-11 14:07 +0000 [5236ffed97]  Matt Jordan <mjordan@digium.com>

	* configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER

	  When running on a system that does not support or use AST_UNDEFINED_SANITIZER
	  or AST_LEAK_SANITIZER, the configure script would incorrectly set those
	  constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would
	  cause menuselect to error out, complaining that a blank value is not a
	  valid option. This patch corrects the issue by setting the value to 0 if
	  the options that those constants enable/disable is not found.

	  Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba

2016-05-10 08:17 +0000 [b5c471b339]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* followme: delete the right recorded name file

	  FollowMe with the option a records the name of the caller and plays it
	  to the callee. However it has failed to clean up that recorded file
	  as it tried to delete the file name without the '.sln' extension.

	  ASTERISK-26008 #close

	  Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec
	  Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

2016-05-10 03:10 +0000 [ec85ea3c21]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* basic-cfg: asterisk.conf: don't set languages

	  * No need to set language in a miniml configuration. 'en' will do just
	    fine.
	  * It would be useful to have an example of setting it to a different
	    language.
	  * Setting the documentation language explicitly is likewise not
	    required. Setting it to a different value is not common. At least
	    until there is a set of translated documentation.

	  Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7
	  Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

2016-05-10 03:08 +0000 [1b0a9bb2c4]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* basic-cfg: asterisk.conf: debug level 5 spams

	  Don't suggest users to use debug level 5, which spews (usually
	  non-useful) debug information. Reduce the suggestion to (an
	  arbitrarily-selected) level 2.

	  Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60
	  Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

2016-05-10 03:06 +0000 [d0ba3e8196]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* basic-cfg: asterisk.conf: defaults of options

	  Note the default of remmed-out options. To clarify that those values are
	  not the defaults.

	  Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738
	  Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

2016-05-10 02:56 +0000 [f943a1fd84]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* basic-cfg: asterisk.conf: remove [directories]

	  A minimal configuration does not need to explicitly spell out the
	  directories. The built-in defaults will do just fine. In many cases
	  they are wrong.

	  Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c
	  Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

2016-05-05 11:37 +0000 [1e876d6915]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_authenticator_digest: Don't use source port in nonce verification

	  From the issue reporter:
	  "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of
	  the timestamp, the source address, the source port, a server UUID that is
	  calculated at startup, and the authentication realm.

	  Rather than caching nonces that we create, we instead attempt to re-calculate
	  the nonce when receiving an incoming request with authentication. We then
	  compare the re-calculated nonce to the incoming nonce, and if they don't match,
	  then authentication has failed early.

	  The problem is that it is possible, especially when using TCP, to receive two
	  requests from the same endpoint but have differing source ports for those
	  requests. Asterisk itself commonly will use different source ports for
	  outbound TCP requests."

	  This patch removes the source port dependency when building the nonce.

	  ASTERISK-25978 #close

	  Change-Id: I871b5f4adce102df1c4988066283095ec509dffe

2016-05-07 14:39 +0000 [dfefbf8731]  gtjoseph <gjoseph@digium.com>

	* config_transport:  Tell pjproject to allow all SSL/TLS protocols

	  The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2.
	  SSL is not allowed.   So, even if you specify "sslv3" for a transport method,
	  it's silently ignored and one of the TLS protocols is used.  This was a new
	  behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that
	  we never caught.

	  Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default().
	  This tells pjproject to set the socket protocol to match the method.

	  ASTERISK-26004 #close

	  Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078

2016-05-05 09:14 +0000 [d03e170ae7]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_pubsub: Use common datastores container API.

	  This migrates res_pjsip_pubsub over to using the newly
	  introduce common datastores management API instead of using
	  its own implementations for both subscriptions and
	  publications.

	  As well the extension state data now provides a generic
	  datastores container instead of a subscription. This allows
	  the dialog-info+xml body generator to work for both
	  subscriptions and publications.

	  ASTERISK-25999 #close

	  Change-Id: I773f9e4f35092da0f653566736a8647e8cfebef1

2016-05-05 09:12 +0000 [94cd351ec4]  Joshua Colp <jcolp@digium.com>

	* datastore: Add common container based datastores API.

	  This change introduces a common container based datastores
	  management API. This has been done in a few places across
	  the tree but this consolidates all of the logic into one
	  place in a generic fashion.

	  ASTERISK-25999

	  Change-Id: I72eb15941dcdbc2a37bb00a33ce00f8755bd336a

2016-05-04 02:40 +0000 [8923c9ac96]  Jaco Kroon <jaco@uls.co.za>

	* app_confbridge: Add a regcontext option for confbridge bridge profiles.

	  This patch allows for having app_confbridge register the name of the
	  conference as an extension into a specific context, similar to
	  regcontext for chan_sip.  This variant is not quite as involved as the
	  one in chan_sip and doesn't allow for multiple contexts or custom
	  extensions, you can only specify the context and the conference name
	  will always be used as the extension to register.

	  ASTERISK-25989 #close

	  Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f

2016-05-08 20:19 +0000 [facce6f632]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Check for python-dev and TEST_FRAMEWORK

	  The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
	  The python bindings are now built only if TEST_FRAMEWORK is set and a
	  python development package is installed.

	  libresample was also disabled.

	  ASTERISK-25993 #close
	  Reported-by: Joshua Colp

	  Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03

2016-05-06 11:54 +0000 [322c3b4262]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: module load priority

	  The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_*
	  and res_pjsip_registrar modules should load ASAP
	  to avoid "No matching endpoint found" for legitimate endpoint.

	  ASTERISK-25994

	  Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b

2016-05-05 15:16 +0000 [516f49f316]  Alexei Gradinari <alex2grad@gmail.com>

	* stasis_endpoints: Add new Status and Headers to ContactStatus

	  ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail.
	  ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail.
	  These additions should be also in stasis_endpoints
	  to include in command "manager show event ContactStatus"

	  Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a

2016-05-03 15:43 +0000 [64e058f75a]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches

	  When reloading, or fetching realtime data, if the "apply" failed for any
	  numerous reasons the current state object would not be maintained. This
	  potentially resulted in publishes being stopped for some states/clients when
	  they should not have been.

	  This patch makes it so the current state object is kept upon any type of reload/
	  fetch failures.

	  Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30

2016-05-03 15:35 +0000 [adc82a2260]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publishing: After unloading the library won't load again

	  The same thing was happening in res_pjsip_publish_asterisk. When the library
	  was unloaded it did not unregister the object type from sorcery. Subsequent
	  loads resulted in a failed load due to the sorcery type already existing.

	  Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9

2016-05-03 15:39 +0000 [3b0ce5169d]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publish: Won't unload if condition wait times out

	  When res_pjsip_outbound_publish unloads it has to wait for all current
	  publishing objects to get done. However if the wait condition times out
	  then it does not fail the unload. This sometimes results in an infinite
	  loop check while unloading. This patch now fails the unload operation if
	  the condition times out.

	  Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec

2016-05-03 14:59 +0000 [41fccbfeb1]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publish: Ref leak in off nominal callback paths

	  There were a few spots where the client object's reference was being leaked in
	  sip_outbound_publish_callback. This patch cleans up those leaks.

	  Change-Id: I485d0bc9335090f373026f77c548042e258461df

2016-05-03 15:31 +0000 [dfbb03cc8e]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publish: Potential crash due to off nominal path

	  It was possible for the explicit publish destroy function to be called without
	  the pjsip client ever being initialized. This fix checks to make sure there is
	  a client to destroy before attempting.

	  Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c

2016-05-05 05:07 +0000 [17b6ba49ef]  Joshua Colp <jcolp@digium.com>

	* file: Ensure nativeformats remains valid for lifetime of use.

	  It is possible for the nativeformats of a channel to change
	  throughout its lifetime. As a result a user of it needs to either
	  ensure the channel is locked when accessing the formats or keep
	  a reference to the nativeformats themselves.

	  This change fixes the file playback support so it keeps a
	  reference to the nativeformats when accessing things.

	  ASTERISK-25998 #close

	  Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915

2016-04-15 09:32 +0000 [cc4c5f5693]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: improve realtime performance

	  This patch modified pjsip_options to retrieve only
	  permament contacts for aor if the qualify_frequency is > 0
	  and persisted contacts if the qualify_frequency is > 0.

	  This patch also fixed a bug in res_sorcery_astdb.
	  res_sorcery_astdb doesn't save object data retrived from astdb.

	  ASTERISK-25826

	  Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05

2016-05-02 16:52 +0000 [92f85fe766]  Alexei Gradinari <alex2grad@gmail.com>

	* res_fax/t38_gateway: Peer V.21 session is created on wrong channel

	  The channel and peer V.21 sessions are created on the same channel now.
	  The peer V.21 session should be created only on peer channel
	  when one of channel can handle T.38.

	  Also this patch enable debug for T.38 gateway session
	  if global fax debug enabled.

	  ASTERISK-25982

	  Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e

2016-05-04 16:11 +0000 [4df48581f1]  Alexei Gradinari <alex2grad@gmail.com>

	* pjsip: Added "reg_server" to contacts (fixed alembic)

	  ASTERISK-25931

	  Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549

2016-05-04 03:17 +0000 [02f4ca1079]  Chris Trobridge <christ.trobridge@ultra-aep.com>

	* config_options.c: Expand #ifdef to contain whole if statement.

	  ASTERISK-25956 #close

	  Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38

2016-05-02 16:08 +0000 [380ac201ac]  Alexei Gradinari <alex2grad@gmail.com>

	* res_fax: add FAXMODE variable

	  The app_fax set FAXMODE variable, but res_fax missing this feature.
	  This patch add FAXMODE variable which is set to either "audio" or "T38".

	  ASTERISK-25980

	  Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b

2016-05-02 05:56 +0000 [0c9faaee47]  Jean Aunis <jean.aunis@prescom.fr>

	* app_chanspy: fix audiohook options in non read-only mode

	  When option 'o' was not set, ChanSpy created its audiohook with the flag
	  AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one
	  direction only.

	  ASTERISK-25866 #close

	  Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0

2016-04-07 16:33 +0000 [a4cfcda036]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip/AMI: add contact.updated event

	  With the old SIP module AMI sends PeerStatus event on every
	  successfully REGISTER requests, ie, on start registration,
	  update registration and stop registration.

	  With PJSIP AMI sends ContactStatus only when status is changed.
	  Regarding registration:
	  on start registration - Created
	  on stop registration - Removed
	  but on update registration nothing

	  This patch added contact.updated event.

	  ASTERISK-25904

	  Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f

2016-04-30 17:52 +0000 [e61716b774]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Various fixes discovered during testing of OSes

	  For all OSes:
	  * Disabled third-party codecs in pjproject and added
	    '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
	    configure options since we don't use the pjsip codec capability.

	  FreeBSD:
	  * Added FreeBSD support to install_prereq.
	  * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
	  * Added __progname and environ to asterisk.exports.in.
	  * Reverted the use of ldconfig to create shared library symlinks to ln.
	  * Only enable epoll in pjproject if `uname -s` is Linux.
	  * Added a patch to pjproject to take the name of the 'make' command from
	    an environment variable if supplied.  This is needed for the python bindings.
	    (merged by Teluu into pjproject trunk 5/3/2016)
	  FreeBSD support isn't complete.  Still some general issues regarding
	  make/gmake having nothing to do with pjproject.  With some handholding it DOES
	  build successfully.

	  CentOS:
	  Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
	  CentOS 6/7 32/64 build and run the pjsip testsuite successfully.

	  Ubuntu:
	  No changes required.
	  Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.

	  Debian:
	  No changes required.
	  Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.

	  There will utimately be a follow-up patch to create an install_prereq for
	  the testsuite as I've discovered a few missing requirements.

	  ASTERISK-25968 #close

	  Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c

2016-03-17 14:29 +0000 [080c6216b6]  Andrew Nagy <andrew.nagy@the159.com>

	* app_voicemail: always copy dynamic struct to avoid race condition

	  Voicemail email addresses can be corrupt or voicemail
	  emails can end up being sent to the wrong email address if asterisk is
	  reading voicemail.conf during a reload and processing an email at the
	  same time. This patch always copies the struct that would otherwise only
	  be copied once.

	  ASTERISK-24463 #close
	  Reported by: John Campbell
	  Tested by: Etienne Lessard
	  Tested by: Andrew Nagy
	  Change-Id: I3a0643813116da84e2617291903d0d489b7425fb

2016-04-15 14:26 +0000 [2b1edee772]  Alexei Gradinari <alex2grad@gmail.com>

	* pjsip: Added "reg_server" to contacts.

	  If the Asterisk system name is set in asterisk.conf, it will be stored
	  into the "reg_server" field in the ps_contacts table to facilitate
	  multi-server setups.

	  ASTERISK-25931

	  Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8

2016-05-01 02:21 +0000 [bf13b59062]  Diederik de Groot <dkgroot@talon.nl>

	* configs/basic-pbx/asterisk.conf: contains incorrect path separator

	  Note: When packagers use these files (as an example) the paths are never
	  really used when they are split using '='.

	  Note: Thirdparty applications will also have trouble parsing the file when
	  expecting '=>'.

	  Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004

2016-04-27 17:19 +0000 [2c46063d54]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_exten_state: Create PUBLISH messages.

	  Create PUBLISH messages to update a third party when an extension state
	  changes because of either a device or presence state change.

	  A configuration example:

	  [exten-state-publisher]
	  type=outbound-publish
	  server_uri=sip:instance1@172.16.10.2
	  event=presence
	  ; Optional regex for context filtering, if specified only extension state
	  ; for contexts matching the regex will cause a PUBLISH to be sent.
	  @context=^users
	  ; Optional regex for extension filtering, if specified only extension
	  ; state for extensions matching the regex will cause a PUBLISH to be sent.
	  @exten=^[0-9]*
	  ; Required body type for the PUBLISH message.
	  ;
	  ; Supported values are:
	  ; application/pidf+xml
	  ; application/xpidf+xml
	  ; application/cpim-pidf+xml
	  ; application/dialog-info+xml (Planned support but not yet)
	  @body=application/pidf+xml

	  The '@' extended variables are used because the implementation can't
	  extend the outbound publish type as it is provided by the outbound publish
	  module.  That means you either have to use extended variables, or
	  implement some sort of custom extended variable thing in the outbound
	  publish module.  Another option would be to refactor that stuff to have an
	  option which specifies the use of an alternate implementation's
	  configuration and then have that passed to the implementation.  JColp
	  opted for the extended variables method originally.

	  ASTERISK-25972 #close

	  Change-Id: Ic0dab4022f5cf59302129483ed38398764ee3cca

2016-04-26 16:10 +0000 [0b5292525c]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_exten_state: Check if body generator is available.

	  When starting the extension state publishers, check if the requested
	  message body generator is available.  If not available give error message
	  and skip starting that publisher.

	  * res_pjsip_pubsub.c: Create new API if type/subtype generator
	  registered.

	  * res_pjsip_exten_state.c: Use new body generator API for validation.

	  ASTERISK-25922

	  Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c

2016-04-28 11:35 +0000 [369182d084]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Start body generator users after suppliers.

	  Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb

2016-04-28 16:06 +0000 [3af83ea2fb]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Add useful information to some messages.

	  Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a

2016-04-26 15:58 +0000 [8e1b663b87]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Fix body generator registration race.

	  Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67

2016-04-28 16:54 +0000 [30415944a8]  gtjoseph <gjoseph@digium.com>

	* pjproject_bundled:  Disable PJSIP_UNESCAPE_IN_PLACE

	  When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled,
	  the input uri string will become corrupted if it contains escape sequences.
	  It's not possible to automatically strdup or strdupa the input string because
	  the output uri pj_str_t's will have pointers to chunks of the input string.
	  Getting around this would require more memory management code and wouldn't
	  be worth the savings of doing the unescape in place.

	  ASTERISK-25970 #close
	  Reported-by: Dmitriy Serov

	  Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88

2016-04-26 15:13 +0000 [906ea2c43f]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.h: Fix doxygen association.

	  Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632

2016-04-25 16:00 +0000 [76ea4cfaae]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_publish.c: Remove redundant flag check.

	  Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353

2016-03-07 18:34 +0000 [4ebf9a938d]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Add ability to identify by Authorization username

	  A feature of chan_sip that service providers relied upon was the ability to
	  identify by the Authorization username.  This is most often used when customers
	  have a PBX that needs to register rather than identify by IP address.  From my
	  own experiance, this is pretty common with small businesses who otherwise
	  don't need a static IP.

	  In this scenario, a register from the customer's PBX may succeed because From
	  will usually contain the PBXs account id but an INVITE will contain the caller
	  id.  With nothing recognizable in From, the service provider's Asterisk can
	  never match to an endpoint and the INVITE just stays unauthorized.

	  The fixes:

	  A new value "auth_username" has been added to endpoint/identify_by that
	  will use the username and digest fields in the Authorization header
	  instead of username and domain in the the From header to match an endpoint,
	  or the To header to match an aor.  This code as added to
	  res_pjsip_endpoint_identifier_user rather than creating a new module.

	  Although identify_by was always a comma-separated list, there was only
	  1 choice so order wasn't preserved.  So to keep the order, a vector was added
	  to the end of ast_sip_endpoint.  This is only used by res_pjsip_registrar
	  to find the aor.  The res_pjsip_endpoint_identifier_* modules are called in
	  globals/endpoint_identifier_order.

	  Along the way, the logic in res_pjsip_registrar was corrected to match
	  most-specific to least-specific as res_pjsip_endpoint_identifier_user does.

	  The order is:

	  username@domain
	  username@domain_alias
	  username

	  Auth by username does present 1 problem however, the first INVITE won't have
	  an Authorization header so the distributor, not finding a match on anything,
	  sends a securty_alert.  It still sends a 401 with a challenge so the next
	  INVITE will have the Authorization header and presumably succeed.  As a result
	  though, that first security alert is actually a false alarm.

	  To address this, a new feature has been added to pjsip_distributor that keeps
	  track of unidentified requests and only sends the security alert if a
	  configurable number of unidentified requests come from the same IP in a
	  configurable amout of time.  Those configuration options have been added to
	  the global config object.  This feature is only used when auth_username
	  is enabled.

	  Finally, default_realm was added to the globals object to replace the hard
	  coded "asterisk" used when an endpoint is not yet identified.

	  The testsuite tests all pass but new tests are forthcoming for this new
	  feature.

	  ASTERISK-25835 #close
	  Reported-by: Ross Beer

	  Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d

2016-04-27 13:23 +0000 [2b150f0b80]  Mark Michelson <mmichelson@digium.com>

	* func_odbc: Check connection status before executing queries.

	  A recent change to func_odbc made it so that a single connection was
	  maintained per DSN. The problem was that the code was optimistic about
	  the health of the connection after initially opening it and did nothing
	  to re-connect in case the connection had died.

	  This change adds a check before executing a query to ensure that the
	  connection to the database is still up and running.

	  ASTERISK-25963 #close
	  Reported by Ross Beer

	  Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d

2016-04-15 11:59 +0000 [860b135c88]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: disable multi domain to improve realtime performace

	  This patch added new global pjsip option 'disable_multi_domain'.
	  Disabling Multi Domain can improve Realtime performance by reducing
	  number of database requests.

	  ASTERISK-25930 #close

	  Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7

2016-04-01 07:50 +0000 [7281770710]  Jean Aunis <jean.aunis@prescom.fr>

	* app_chanspy: reduce audio loss on the spying channel.

	  ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC
	  and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when
	  queues grow too large or when read and write queues go out of sync.
	  Now these flags are set conditionally:
	  - AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set
	  - a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not
	  be set on the audiohook

	  ASTERISK-25866

	  Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd

2016-04-14 07:03 +0000 [81ea80b74c]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_exten_state: Add config support for exten state publishers.

	  This change adds the ability to configure outbound publishing of
	  extension state. Right now stuff is merely set up to store the
	  configuration and to register a global extension state callback. The
	  act of constructing the body and sending is not yet complete.

	  Configurable elements right now are a regex for filtering the context,
	  a regex for filtering the extension, and the body type to publish.

	  ASTERISK-25922 #close

	  Change-Id: Ia7e630136dfc355073c1cadff8ad394a08523d78

2016-04-26 11:13 +0000 [c480159045]  Joshua Colp <jcolp@digium.com>

	* chan_sip: Give more time for TCP/TLS threads to stop.

	  The unload process currently tells each TCP/TLS to terminate but
	  does not wait for them to do so. This introduces a race condition
	  where the container holding the threads may be destroyed before
	  the threads are able to remove themselves from it. When they
	  finally do the container is invalid and can't be used causing a
	  crash.

	  A previous change existed which waited a bit to wait for any
	  stranglers to finish. This change extends this and waits longer.

	  ASTERISK-25961 #close

	  Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6

2016-04-26 05:48 +0000 [8ae69cffef]  Joshua Colp <jcolp@digium.com>

	* app_queue: Fix crash when unloading module.

	  When unloading the app_queue module the members in each queue are
	  destroyed and as part of this they are removed from the pending
	  members container. Unfortunately a crash would occur as the container
	  was destroyed before the members were removed.

	  This change tweaks ordering so the container destruction occurs
	  after the members are destroyed.

	  ASTERISK-16115

	  Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b

2016-04-24 22:51 +0000 [284bb814ac]  gtjoseph <gjoseph@digium.com>

	* config:  Fix ast_config_text_file_save2 writability check for missing files

	  A patch I did back in 2014 modified ast_config_text_file_save2 to check the
	  writability of the main file and include files before truncating and re-writing
	  them.  An unintended side-effect of this was that if a file doesn't exist,
	  the check fails and the write is aborted.

	  This patch causes ast_config_text_file_save2 to check the writability of the
	  parent directory of missing files instead of checking the file itself.  This
	  allows missing files to be created again.  A unit test was also added to
	  test_config to test saving of config files.

	  The regression was discovered when app_voicemail's passwordlocation=spooldir
	  feature stopped working.

	  ASTERISK-25917 #close
	  Reported-by: Jonathan Rose

	  Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80

2016-04-25 08:11 +0000 [f99ec857c8]  Javier Acosta <javier.acosta@beeonline.es>

	* Fix case sensitive actions in AMI QueueSummary and QueueStatus

	  ASTERISK-25954 #close
	  Reported by: Javier Acosta

	  Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256
	  (cherry picked from commit c0688a6398f27296ff849848a2e416e036d794e3)

2016-04-21 14:23 +0000 [30ab21d5fa]  Kevin Harwell <kharwell@digium.com>

	* app_queue: queue members can receive multiple calls

	  It was possible for a queue member that is a member of at least 2 or more
	  queues to receive mulitiple calls at the same time. This happened because
	  of a race between when a member was being rung and when the device state
	  notified the other queue(s) member object of the state change.

	  This patch makes it so when a queue member is being rung it gets added to
	  a global pool of queue members. If that same member is tried again, e.g.
	  from another queue, and it is found to already exist in the pending member
	  container then it will not ring that member.

	  ASTERISK-16115 #close

	  Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48

2016-04-22 17:53 +0000 [99fcf2a791]  gtjoseph <gjoseph@digium.com>

	* res_agi:  Prevent run_agi from eating frames it shouldn't

	  The run_agi function is eating control frames when it shouldn't be. This is
	  causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond
	  transfer.

	  Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie
	  answers.

	  Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE
	  and is left thinking he's connected to Bob.

	  In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls
	  an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on
	  Charlie's channel.

	  The fix was to accumulate deferrable frames in the "forever" loop instead of
	  dropping them, and re-queue them just before running the actual agi command
	  or exiting.

	  ASTERISK-25951 #close

	  Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645

2016-04-22 15:25 +0000 [757ec6172b]  Richard Mudgett <rmudgett@digium.com>

	* test_message.c: Wait longer in case dialplan also processes the test message.

	  Bumped the wait from 1 second to 5 seconds.  The test message was hitting my
	  default call handler and failing the test because it took longer.

	  Change-Id: I3a03737f25e92983de00548fcc7bbc50dd7544ba

2016-04-21 23:53 +0000 [41ecf22587]  Kirill Katsnelson <kkm@smartaction.com>

	* chan_sip: Make autocreated peers send PeerStatus events

	  Since Stasis has been introduced, an attempt to send AMI messages by an
	  autocreated peer caused a crash, and all events from autocreated peers were
	  semi-inadvertently disabled altogether in 0b83761. This change restores the
	  disabled functionality.

	  ASTERISK-25950

	  Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974

2016-04-13 17:09 +0000 [b3cc74fda9]  Richard Mudgett <rmudgett@digium.com>

	* manager_channels.c: Fix allocation failure crash.

	  An earlier allocation failure failed to create a channel snapshot for the
	  AMI HangupRequest/SoftHangupRequest event which resulted in a crash in
	  channel_hangup_request_cb().  Where the stasis message gets generated
	  cannot tell if the NULL snapshot returned was because of an allocation
	  failure or the channel was a dummy channel.

	  * Made channel_hangup_request_cb() check if the channel blob has a
	  snapshot and exit if it doesn't.

	  * Eliminated the RAII_VAR usage in channel_hangup_request_cb().

	  Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24

2016-04-13 13:50 +0000 [a63656b419]  Richard Mudgett <rmudgett@digium.com>

	* Bridge system: Fix memory leaks and double frees on impart failure.

	  You cannot reference the passed in features struct after calling
	  ast_bridge_impart().  Even if the call fails.

	  Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21

2016-04-13 13:20 +0000 [71dfa35540]  Richard Mudgett <rmudgett@digium.com>

	* bridge_softmix.c: Fix crash if channel fails to join mixing tech.

	  softmix_bridge_join() failed because of an allocation failure.  To address
	  this, the softmix bridge technology now checks if the channel failed to
	  join softmix successfully.  In addition, the bridge now begins the process
	  of kicking the channel out of the bridge so we don't have channels
	  partially in the bridge for very long.

	  * Fix the test_channel_feature_hooks.c unit tests.  The test channel must
	  have a valid codec to join the simple_bridge technology.  This patch makes
	  joining a bridge more strict by not allowing partially joined channels to
	  remain in the bridge.

	  Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b

2016-04-12 15:29 +0000 [06632a0d11]  Richard Mudgett <rmudgett@digium.com>

	* Manager: Short circuit AMI message processing.

	  Improve AMI message processing performance if there are no consumers
	  listening for the messages.  We now skip creating the AMI event message
	  text strings.

	  Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3

2016-04-13 17:54 +0000 [6ddd856b86]  Richard Mudgett <rmudgett@digium.com>

	* manager.c: Eliminate most RAII_VAR usage.

	  * Made ast_manager_event_blob_create() not allocate the ao2 event object
	  with a lock as it is not needed.

	  Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c

2016-04-22 13:49 +0000 [924738e950]  Mark Michelson <mmichelson@digium.com>

	* func_odbc: Use one connection per DSN.

	  res_odbc was changed in Asterisk 13.8.0 to remove connection management,
	  opting instead to let unixodbc maintain open connections and return
	  those to Asterisk as requested.

	  This was a boon for realtime, since it meant that multiple threads could
	  potentially run parallel queries since they could each be using their
	  own database connections.

	  However, on the user-facing side, func_odbc, there were some inherent
	  behaviors being relied on that no longer hold true after the change.
	  One such reported behavior was that MySQL's LAST_INSERTED_ID() works
	  per-connection. This means that if Asterisk uses separate connections
	  for every database operation, whereas before it used one connection for
	  everything, we have broken expectations and functionality.

	  The fix provided in this patch is to make func_odbc use a single
	  database connection per DSN. This way, user-facing database usage will
	  have the same behavior as it did pre-13.8.0. However, realtime, which is
	  the real workhorse of database interaction, will continue to let
	  unixodbc manage connections.

	  ASTERISK-25938 #close
	  Reported by Edwin Vandamme

	  Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc

2016-04-22 13:02 +0000 [6ede210c98]  Leif Madsen <leif@leifmadsen.com>

	* Remove reference to non-existent sip.conf option

	  Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f

	  ASTERISK-25927 #close

	  Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8

2016-04-21 08:26 +0000 [c991e5472e]  Diederik de Groot <dkgroot@talon.nl>

	* lock.c: Check *lt before dereferencing it

	  *lt is NULL if t->tracking == 0

	  ASTERISK-25948 #close

	  Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba

2016-04-15 14:36 +0000 [6b1a632290]  Richard Mudgett <rmudgett@digium.com>

	* res_stasis: Handle re-enter stasis bridge with swap channel.

	  We lose the fact that there is a swap channel if there is one.  We
	  currently wind up rejoining the stasis bridge as a normal join after the
	  swap channel has already been kicked from the bridge.

	  This patch preserves the swap channel so the AMI/ARI events can note that
	  the channel joining the bridge is swapping with another channel.  Another
	  benefit to swaqpping in one operation is if there are any channels that
	  get lonely (MOH, bridge playback, and bridge record channels).  The lonely
	  channels won't leave before the joining channel has a chance to come back
	  in under stasis if the swap channel is the only reason the lonely channels
	  are staying in the bridge.

	  ASTERISK-25947 #close
	  Reported by: Richard Mudgett

	  ASTERISK-24649
	  Reported by: John Bigelow

	  ASTERISK-24782
	  Reported by: John Bigelow

	  Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee

2016-04-19 16:58 +0000 [1c5248c383]  Richard Mudgett <rmudgett@digium.com>

	* bridge: Hold off more than one imparting channel at a time.

	  An earlier patch blocked the ast_bridge_impart() call until the channel
	  either entered the target bridge or it failed.  Unfortuantely, if the
	  target bridge is stasis and the imprted channel is not a stasis channel,
	  stasis bounces the channel out of the bridge to come back into the bridge
	  as a proper stasis channel.  When the channel is bounced out, that
	  released the block on ast_bridge_impart() to continue.  If the impart was
	  a result of a transfer, then it became a race to see if the swap channel
	  would get hung up before the imparted channel could come back into the
	  stasis bridge.  If the imparted channel won then everything is fine.  If
	  the swap channel gets hung up first then the transfer will fail because
	  the swap channel is leaving the bridge.

	  * Allow a chain of ast_bridge_impart()'s to happen before any are
	  unblocked to prevent the race condition described above.  When the channel
	  finally joins the bridge or completely fails to join the bridge then the
	  ast_bridge_impart() instances are unblocked.

	  ASTERISK-25947
	  Reported by: Richard Mudgett

	  ASTERISK-24649
	  Reported by: John Bigelow

	  ASTERISK-24782
	  Reported by: John Bigelow

	  Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1

2016-04-19 17:52 +0000 [70e860ec49]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_callerid:  Clear out display name if id->name is not valid

	  When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning
	  the From header, then it overwrites the display name and uri from the channel's
	  connected.id.  If the connected.id.name wasn't valid, create_new_id_hdr was
	  leaving the display name from the From header in the new RPID or PAI header.
	  On an attended transfer where the originator had a caller id number set but not
	  a display name, the re-INVITE to the final transferee had the number of the
	  originator but the display name of the transferer.

	  Added a check to clear out the display name in the new header if
	  connected.id.name was invalid.

	  ASTERISK-25942 #close

	  Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b

2016-04-19 13:02 +0000 [d95512a7dd]  Joshua Colp <jcolp@digium.com>

	* app_talkdetect: Make the module core supported.

	  This module is used as part of testsuite tests to confirm
	  stuff works. I'm accordingly marking it as core as it is
	  required by those tests.

	  Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88

2016-04-18 12:12 +0000 [0235a66532]  Mark Michelson <mmichelson@digium.com>

	* PJSIP: Remove PJSIP parsing functions from uri length validation.

	  The PJSIP parsing functions provide a nice concise way to check the
	  length of a hostname in a SIP URI. The problem is that in order to use
	  those parsing functions, it's required to use them from a thread that
	  has registered with PJLib.

	  On startup, when parsing AOR configuration, the permanent URI handler
	  may not be run from a PJLib-registered thread. Specifically, this could
	  happen when Asterisk was started in daemon mode rather than
	  console-mode. If PJProject were compiled with assertions enabled, then
	  this would cause Asterisk to crash on startup.

	  The solution presented here is to do our own parsing of the contact URI
	  in order to ensure that the hostname in the URI is not too long. The
	  parsing does not attempt to perform a full SIP URI parse/validation,
	  since the hostname in the URI is what is important.

	  ASTERISK-25928 #close
	  Reported by Joshua Colp

	  Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60

2016-04-18 17:00 +0000 [b8b60135ec]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_registrar: Fix bad memory-ness with user_agent.

	  Recent changes to the PJSIP registrar resulted in tests failing due to
	  missing AOR_CONTACT_ADDED test events. The reason for this was that the
	  user_agent string had junk values in it, resulting in being unable to
	  generate the event.

	  I'm going to be honest here, I have no idea why this was happening. Here
	  are the steps needed for the user_agent variable to get messed up:
	  * REGISTER is received
	  * First contact in the REGISTER results in a contact being removed
	  * Second contact in the REGISTER results in a contact being added
	  * The contact, AOR, expiration, and user agent all have to be passed as
	    format parameters to the creation of a string. Any subset of those
	    parameters would not be enough to cause the problem.

	  Looking into what was happening, the thing that struck me as odd was
	  that the user_agent variable was meant to be set to the value of the
	  User-Agent SIP header in the incoming REGISTER. However, when removing a
	  contact, the user_agent variable would be set (via ast_strdupa inside a
	  loop) to the stored contact's user_agent. This means that the
	  user_agent's value would be incorrect when attempting to process further
	  contacts in the incoming REGISTER.

	  The fix here is to use a different variable for the stored user agent
	  when removing a contact. Correcting the behavior to be correct also
	  means the memory usage is less weird, and the issue no longer occurs.

	  ASTERISK-25929 #close
	  Reported by Joshua Colp

	  Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08

2016-04-18 13:41 +0000 [6cfa02394f]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_transport_management: Allow unload to occur.

	  At shutdown it is possible for modules to be unloaded that wouldn't
	  normally be unloaded. This allows the environment to be cleaned up.

	  The res_pjsip_transport_management module did not have the unload
	  logic in it to clean itself up causing the res_pjsip module to not
	  get unloaded. As a result the res_pjsip monitor thread kept going
	  processing traffic and timers when it shouldn't.

	  Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a

2016-04-15 11:41 +0000 [6365f0018f]  Richard Mudgett <rmudgett@digium.com>

	* bridge_channel.c: Ignore role setup failure in channel push.

	  We have to setup the channel roles after the bridge class push is called
	  because the bridge class push callback may have set roles on the incoming
	  channel.  Since we have already partially pushed the channel into the
	  bridge and reversing what we have already done could be problematic, the
	  only thing we can do is press on to complete pushing the channel into the
	  bridge.

	  * Ignore any channel role setup errors after pushing the channel into a
	  bridge.  The channel may behave incorrectly in the bridge but we can no
	  longer abort the push at this time.

	  Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00

2016-04-17 15:37 +0000 [f06ce7f90a]  Jaco Kroon <jaco@uls.co.za>

	* chan_sip: Don't verify table if rtupdate=no

	  If rtupdate=no do not verify sipregs/peers table has updatable fields.

	  ASTERISK-25934 #close

	  Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d

2016-04-18 04:53 +0000 [dbb47e0a47]  ibercom <ibercom123@gmail.com>

	* app_queue: Frequent segfaults in function can_ring_entry()

	  ASTERISK-25888 #close

	  Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117

2016-04-15 16:51 +0000 [af114edb8b]  Richard Mudgett <rmudgett@digium.com>

	* stasis_bridge.c: Update stasis bridge push diagnostic messages.

	  Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a

2016-04-12 14:55 +0000 [5e64d7e7a3]  Mark Michelson <mmichelson@digium.com>

	* Dial: Combine frame handling functions.

	  There is a good amount of repetition in the two frame handling routines
	  in the Dial API. This commit combines the two functions into one.

	  This is in preparation for an upcoming commit that adds the ability to
	  handle frames for a channel in a bridge.

	  ASTERISK-25925
	  Reported by Mark Michelson

	  Change-Id: Iaae2f174e3058e774cb44e10659fcdfb85345c58

2016-04-11 16:20 +0000 [a6e2ba187a]  Alexei Gradinari <alex2grad@gmail.com>

	* Codecs: strip codec name while parsing allow/disallow options

	  Failed registration using PJSIP/Realtime if one of the codec name
	  in allow/disallow option is wrong or contains space.

	  This patch strip codec name.

	  ASTERISK-25914

	  Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d

2016-04-14 13:49 +0000 [be4333ddad]  Mark Michelson <mmichelson@digium.com>

	* transport management: Register thread with PJProject.

	  The scheduler thread that kills idle TCP connections was not registering
	  with PJProject properly and causing assertions if PJProject was built in
	  debug mode.

	  This change registers the thread with PJProject the first time that the
	  scheduler callback executes.

	  AST-2016-005

	  Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283

2016-03-17 12:28 +0000 [e83499df56]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Add serialized scheduler (res_pjsip/pjsip_scheduler.c)

	  There are several places that do scheduled tasks or periodic housecleaning,
	  each with its own implementation:

	  * res_pjsip_keepalive has a thread that sends keepalives.
	  * pjsip_distributor has a thread that cleans up expired unidentified requests.
	  * res_pjsip_registrar_expire has a thread that cleans up expired contacts.
	  * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task.
	  * res_pjsip_sdp_rtp also uses ast_sched to send keepalives.

	  There are also places where we should be doing scheduled work but aren't.
	  A good example are the places we have sorcery observers to start registration
	  or qualify.  These don't work when changes are made to a backend database
	  without a pjsip reload.  We need to check periodically.

	  As a first step to solving these issues, a new ast_sip_sched facility has
	  been created.

	  ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue.
	  When a task is ready to run, ast_sip_task_pusk is called for it. This ensures
	  that the task is executed in a PJLIB registered thread and doesn't hold up the
	  ast_sched thread so it can immediately continue processing the queue.  The
	  serializer used by ast_sip_sched is one of your choosing or a random one from
	  the res_pjsip pool if you don't choose one.

	  Another feature is the ability to automatically clean up the task_data when the
	  task expires (if ever).  If it's an ao2 object, it will be dereferenced, if
	  it's a malloc'd object it will be freed.  This is selectable when the task is
	  scheduled.  Even if you choose to not auto dereference an ao2 task data object,
	  the scheduler itself maintains a reference to it while the task is under it's
	  control.  This prevents the data from disappearing out from under the task.

	  There are two scheduling models.

	  AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at
	  the specific interval.  That is, every "interval" milliseconds, regardless of
	  how long the task takes.  If the task takes longer than the interval, it will
	  be scheduled at the next available multiple of interval.  For exmaple: If the
	  task has an interval of 60 secs and the task takes 70 secs (it better not),
	  the next invocation will happen at 120 seconds.

	  AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should
	  start "interval" milliseconds after the current invocation has finished.

	  Also, the same ast_sched facility for fixed or variable intervals exists.  The
	  task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or
	  AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time.

	  One res_pjsip.h housekeeping change was made.  The pjsip header files were
	  added to the top.  There have been a few cases lately where I've needed
	  res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because
	  I didn't add the pjsip header files to my source even though I never referenced
	  any pjsip calls.

	  Finally, a few new convenience APIs were added to astobj2 to make things a
	  little easier in the scheduler.  ao2_ref_and_lock() calls ao2_ref() and
	  ao2_lock() in one go.  ao2_unlock_and_unref() does the reverse. A few macros
	  were also copied from res_phoneprov because I got tired of having to duplicate
	  the same hash, sort and compare functions over and over again. The
	  AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for
	  aor_container_alloc into your source.

	  This facility can be used immediately for the situations where we already have
	  a thread that wakes up periodically or do some scheduled work.  For the
	  registration and qualify issues, additional sorcery and schema changes would
	  need to be made so that we can easily detect changed objects on a periodic
	  basis without having to pull the entire database back to check.  I'm thinking
	  of a last-updated timestamp on the rows but more on this later.

	  Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c

2016-03-08 12:12 +0000 [216f22fd0f]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_transport_management: Kill idle TCP connections.

	  "Idle" here means that someone connects to us and does not send a SIP
	  request. PJProject will not automatically time out such connections, so
	  it's up to Asterisk to do it instead.

	  When we receive an incoming TCP connection, we will start a timer
	  (equivalent to transaction timer D) waiting to receive an incoming
	  request. If we do not receive a request in that timeframe, then we will
	  shut down the TCP connection.

	  ASTERISK-25796 #close
	  Reported by George Joseph

	  AST-2016-005

	  Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6

2016-03-08 10:52 +0000 [d9fba46016]  Mark Michelson <mmichelson@digium.com>

	* Rename res_pjsip_keepalive res_pjsip_transport_management

	  ASTERISK-25796
	  Reported by George Joseph

	  AST-2016-005

	  Change-Id: Id322a05f927392293570599730050bc677d99433

2016-04-14 07:23 +0000 [7b8b6e2e4f]  Mark Michelson <mmichelson@digium.com>

	* AST-2016-004: Fix crash on REGISTER with long URI.

	  Due to some ignored return values, Asterisk could crash if processing an
	  incoming REGISTER whose contact URI was above a certain length.

	  ASTERISK-25707 #close
	  Reported by George Joseph

	  Patches:
	      0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch

	  AST-2016-004

	  Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d

2016-04-12 13:10 +0000 [ff3af764de]  Richard Mudgett <rmudgett@digium.com>

	* bridge_softmix.c: Fix crash if could not allocate the dsp.

	  Fix off nominal crash where we could not setup the channel to process
	  frames for the softmix bridge technology because of allocation failure.

	  Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372

2016-04-13 13:38 +0000 [caa416d5f3]  gtjoseph <george.joseph@fairview5.com>

	* stringfields:  Update extended string fields for master only.

	  In 13, the new ast_string_field_header structure had to be dynamically
	  allocated and assigned to a pointer in ast_string_field_mgr to preserve ABI
	  compatability.  In master, it can be converted to being a structure-in-place in
	  ast_string_field_mgr to eliminate the extra alloc and free calls.

	  Change-Id: Ia97c5345eec68717a15dc16fe2e6746ff2a926f4

2016-04-12 15:41 +0000 [bd3671b397]  gtjoseph <george.joseph@fairview5.com>

	* pjproject:  Add patch for removing strip of '[]' from header params

	  From the patch submitted to Teluu on 4/12/2016
	  <<<<<<<<<
	  The wholesale stripping of '[]' from header parameters causes issues if
	  something (like a port) occurs after the final ']'.

	  '[2001:a::b]' will correctly parse to '2001:a::b'
	  '[2001:a::b]:8080' will correctly parse to '2001:a::b' but the scanner is left
	  with ':8080' and parsing stops with a syntax error.

	  I can't even find a case where stripping the '[]' is a good thing anyway.  Even
	  if you continued to parse and resulted in a string that looks like this...
	  '2001:a::b:8080', it's not valid.

	  This came up in Asterisk because Kamailio sends us a Contact with an alias
	  URI parameter that has an IPv6 address in it like this:
	  Contact: <sip:1171@127.0.0.1:5080;alias=[2001:1:2::3]~43691~6>
	  which should be legal but causes a syntax error because of the characters
	  after the final ']'.  Even if it didn't, the '[]' should still not be stripped.

	  I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6
	  enabled.  No issues were caused by removing the code that strips the '[]'.
	  >>>>>>>>>>>

	  ASTERISK-25123 #close
	  Reported-by: Anthony Messina

	  Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a

2016-04-12 09:10 +0000 [5a0534dc62]  Joshua Colp <jcolp@digium.com>

	* app_voicemail: Fix test_voicemail_notify_endl test.

	  The test_voicemail_notify_endl test checks the end-of-line
	  characters of an email message to confirm that they are consistent.
	  The test wrongfully assumed that reading from the email message
	  into a buffer will always result in more than 1 character being
	  read. This is incorrect. If only 1 character was read the test
	  would go outside of the buffer and access other memory causing
	  a crash.

	  The test now checks to ensure that 2 or more characters are read
	  in ensuring the test stays within the buffer.

	  ASTERISK-25874 #close

	  Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710

2016-04-07 12:02 +0000 [c00c298a0e]  Alexei Gradinari <alex2grad@gmail.com>

	* app_voicemail/IMAP: function 'save_to_folder' creates wrong folder

	  If try to move message to Cust1 (number 5)
	  the function 'save_to_folder' tries to create Greeting folder instead of Cust1.

	  This patch fixed it by setting GREETINGS_FOLDER = -1

	  ASTERISK-24927 #close

	  Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51

2016-04-07 16:18 +0000 [49813bc9e5]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip: Add headers to AMI Event ContactStatusDetail

	  * Added Useragent and RegExpire headers to AMI Event
	  ContactStatusDetail with associated documentation.

	  ASTERISK-25903 #close

	  Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239

2016-04-05 16:56 +0000 [4e00e31ef1]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip_outbound_publish: Add transport for outbound PUBLISH

	  The first available transport of the appropriate type is used now.
	  This patch adds new config option 'transport' for outbound-publish.
	  If transport is set then outbound PUBLISH requests will use this transport.

	  ASTERISK-25901 #close

	  Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151

2016-04-11 14:26 +0000 [2cc56573de]  Jaco Kroon <jaco@uls.co.za>

	* core_unreal: Fix hangupcauses not getting set on Local channels

	  ASTERISK-25912 #close

	  Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa

2016-04-01 13:30 +0000 [a621dd5e96]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip contact:  Lock expiration/addition of contacts

	  Contact expiration can occur in several places:  res_pjsip_registrar,
	  res_pjsip_registrar_expire, and automatically when anyone calls
	  ast_sip_location_retrieve_aor_contact.  At the same time, res_pjsip_registrar
	  may also be attempting to renew or add a contact.  Since none of this was locked
	  it was possible for one thread to be renewing a contact and another thread to
	  expire it immediately because it was working off of stale data.  This was the
	  casue of intermittent registration/inbound/nominal/multiple_contacts test
	  failures.

	  Now, the new named lock functionality is used to lock the aor during contact
	  expire and add operations and res_pjsip_registrar_expire now checks the
	  expiration with the lock held before deleting the contact.

	  ASTERISK-25885 #close
	  Reported-by: Josh Colp

	  Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059

2016-04-10 14:16 +0000 [8637f29d24]  gtjoseph <george.joseph@fairview5.com>

	* pjproject:  Add patch to fix Via IPv6 parsing

	  There's a bug in pjproject's sip_parser where the ":" wasn't correctly
	  interpreted. This is causing IPv6 addresses in the "received" parameter of the
	  Via header to cause a syntax check failure.

	  This patch was submitted to Teluu on 4/10/2016.

	  ASTERISK-25910 #close
	  Reported-by: Anthony Messina

	  Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922

2016-03-31 20:04 +0000 [216abb0ae7]  gtjoseph <george.joseph@fairview5.com>

	* lock:  Add named lock capability

	  Locking some objects like sorcery objects can be tricky because the underlying
	  ao2 object may not be the same for all callers.  For instance, two threads that
	  call ast_sorcery_retrieve_by_id on the same aor name might actually get 2
	  different ao2 objects if the underlying wizard had to rehydrate the aor from a
	  database. Locking one ao2 object doesn't have any effect on the other even if
	  those objects had locks in the first place.

	  Named locks allow access control by keyspace and key strings.  Now an "aor"
	  named "1000" can be locked and any other thread attempting to lock "aor" "1000"
	  will wait regardless of whether the underlying ao2 object is the same or not.
	  Mutex and rwlocks are supported.

	  This capability will initially be used to lock an aor when multiple threads may
	  be attempting to prune expired contacts from it.

	  Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45

2016-04-07 11:37 +0000 [f9dab80816]  Alexei Gradinari <alex2grad@gmail.com>

	* app_voicemail/IMAP: IMAP access FATAL error: Out of memory

	  Sometimes uw-imap function 'mail_fetchbody' returns huge len
	  which then pass to uw-imap function 'rfc822_base64'.
	  uw-imap tries to allocate huge memory and abort() on fail.

	  This patch check the len.
	  If the len more than max size (128 Mbytes) log error.
	  This patch also set variables len, newlen to avoid uninizialezed len.
	  This patch also check pointer returned by rfc822_base64.

	  ASTERISK-25899 #close

	  Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca

2016-04-07 16:39 +0000 [b3be945415]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event

	  BLF pickup isn't working on Cisco SPA and Snom phones
	  if the direction="recipient" attribute is missing in 'dialog' tag.

	  This patch adds direction="recipient" if extension state is
	  Ringing.

	  ASTERISK-24601 #close

	  Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c

2016-04-06 17:57 +0000 [6138a75e8e]  Richard Mudgett <rmudgett@digium.com>

	* pbx.h: Make ast_state_cb_type take more const.

	  This eliminates some casts that I made a note saying v10 and above
	  would no longer need them.

	  Better late than never :)

	  Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572

2016-04-07 10:59 +0000 [72c19f7dc5]  Richard Mudgett <rmudgett@digium.com>

	* pbx.c: Minor code rearangements.

	  * Pull out a loop invariant.

	  * Convert an else-if ladder to a switch statement.

	  Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95

2016-04-07 12:26 +0000 [28cefc3e88]  Richard Mudgett <rmudgett@digium.com>

	* pbx: Update doxygen for extension state watchers.

	  Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e

2016-04-07 11:49 +0000 [751d7a5a49]  gtjoseph <george.joseph@fairview5.com>

	* alembic:  Remove batch operations (and sqlite support)

	  Because SQLite doesn't support full ALTER capabilities, alembic scripts
	  require batch operations.  However, that capability wasn't available until
	  0.7.0 which some distributions haven't reached yet.  Therefore, the batch
	  operations introduced in commit 86d6e44cc (review 2319) have been reverted
	  and SQLite is unsupported again, for now anyway.

	  Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql.

	  ASTERISK-25890 #close
	  Reported-by: Harley Peters

	  Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80

2016-04-07 11:05 +0000 [2eaeea690d]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_registrar_expire: Fix race condition at shutdown.

	  When shutting down, the PJSIP sorcery is destroyed. The registrar
	  expiration module queries the PJSIP sorcery to determine what
	  to expire. As there was no synchronization between termination
	  of the expiration thread and the unloading of the module it was
	  possible for the thread to try to access the PJSIP sorcery after
	  it had been destroyed.

	  This change ensures that the thread is shut down before allowing
	  the module to be considered unloaded.

	  Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b

2016-04-06 16:28 +0000 [3e5672d843]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Fix configuration setting of "regcontext".

	  Due to a merge problem two options were swapped causing the
	  regcontext setting to not get set.

	  Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1

2016-04-06 08:01 +0000 [8ed5f61152]  Jacek Konieczny <jkonieczny@eggsoft.pl>

	* frame.c: Copy the whole subclass in ast_frdup().

	  The problem is ast_frdup() does not copy whole frame.subclass for voice,
	  video and image frames, only the format is copied.  For video frames, the
	  subclass structure contains the .frame_ending flag used to put the RTP
	  marker where it needs to be.

	  ASTERISK-25894 #close

	  Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33

2016-03-30 17:18 +0000 [abbb2edd4c]  Mark Michelson <mmichelson@digium.com>

	* ARI: Add method to Dial a created channel.

	  This adds a new ARI method that allows for you to dial a channel that
	  you previously created in ARI.

	  By combining this with the create method for channels, it allows for a
	  workflow where a channel can be created, manipulated, and then dialed.
	  The channel is under control of the ARI application during all stages of
	  the Dial and can even be manipulated based on channel state changes
	  observed within an ARI application.

	  The overarching goal for this is to eventually be able to add a dialed
	  channel to a Stasis bridge earlier than the "Up" state. However, at the
	  moment more work is needed in the Dial and Bridge APIs in order to
	  facilitate that.

	  ASTERISK-25889 #close

	  Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205

2016-03-30 17:01 +0000 [dd48d60c5b]  Mark Michelson <mmichelson@digium.com>

	* ARI: Add method to create a new channel.

	  This adds a new ARI method to the channels resource that allows for the
	  creation of a new channel. The channel is created and then placed into
	  the specified Stasis application.

	  This is different from the existing originate method that creates a
	  channel, dials it, and then places the answered channel into the
	  dialplan or a Stasis application. This method does not attempt to call
	  the channel at all. Dialing is left as a later step after channel
	  creation. This allows for pre-dialing channel manipulation if desired.

	  ASTERISK-25889

	  Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8

2016-03-28 11:31 +0000 [1dc5e28624]  Joshua Colp <jcolp@digium.com>

	* pbx: Add support for autohints.

	  This change introduces the concept of autohints. These are hints
	  which are created as a result of device state changes occurring within
	  the core. When this happens a hint will be created (if it does not
	  exist already) using the device name as the extension.

	  For example if a device state change is received for "PJSIP/bob"
	  and autohints are enabled on a context then a hint will exist in
	  that context for "bob" with a device of "PJSIP/bob".

	  For virtual or custom device states the name after the type will
	  be used. For example if the device state of "Custom:bob" changes
	  then a hint will exist in that context for "bob" with a device of
	  "Custom:bob".

	  This functionality can be enabled in extensions.conf by placing
	  "autohints=yes" in a context.

	  ASTERISK-25881 #close

	  Change-Id: I7e444c7da41b7b7d33374420fec658beeb18584e

2016-04-05 14:23 +0000 [a098251e7e]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Handle deferred SDP hold/unhold properly.

	  Some SIP devices indicate hold/unhold using deferred SDP reinvites. In
	  other words, they provide no SDP in the reinvite.

	  A typical transaction that starts hold might look something like this:

	  * Device sends reinvite with no SDP
	  * Asterisk sends 200 OK with SDP indicating sendrecv on streams.
	  * Device sends ACK with SDP indicating sendonly on streams.

	  At this point, PJMedia's SDP negotiator saves Asterisk's local state as
	  being recvonly.

	  Now, when the device attempts to unhold, it again uses a deferred SDP
	  reinvite, so we end up doing the following:

	  * Device sends reinvite with no SDP
	  * Asterisk sends 200 OK with SDP indicating recvonly on streams
	  * Device sends ACK with SDP indicating sendonly on streams

	  The problem here is that Asterisk offered recvonly, and by RFC 3264's
	  rules, if an offer is recvonly, the answer has to be sendonly. The
	  result is that the device is not taken off hold.

	  What is supposed to happen is that Asterisk should indicate sendrecv in
	  the 200 OK that it sends. This way, the device has the freedom to
	  indicate sendrecv if it wants the stream taken off hold, or it can
	  continue to respond with sendonly if the purpose of the reinvite was
	  something else (like a session timer refresher).

	  The fix here is to alter the SDP negotiator's state when we receive a
	  reinvite with no SDP. If the negotiator's state is currently in the
	  recvonly or inactive state, then we alter our local state to be
	  sendrecv. This way, we allow the device to indicate the stream state as
	  desired.

	  ASTERISK-25854 #close
	  Reported by Robert McGilvray

	  Change-Id: I7615737276165eef3a593038413d936247dcc6ed

2016-03-30 16:47 +0000 [ef4d3f1328]  Mark Michelson <mmichelson@digium.com>

	* Dial: Add function to append already-created channel.

	  The Dial API takes responsiblity for creating an outbound channel when
	  calling ast_dial_append(). This commit adds a new function,
	  ast_dial_append_channel(), which allows us to create the channel outside
	  the Dial API and then to append the channel to the ast_dial structure.

	  This is useful for situations where the channel's creation and dialing
	  are distinct operations. Upcoming ARI early bridge work will illustrate
	  its usage.

	  ASTERISK-25889

	  Change-Id: Id8179f64f8f99132f80dead8d5db2030fd2c0509

2016-03-27 23:33 +0000 [984d6fd95c]  gtjoseph <george.joseph@fairview5.com>

	* config:  Allow filters when appending to a category

	  In sorcery based config files where there are multiple categories with the same
	  name, you can't use the (+) operator to reliably append to a category because
	  config.c stops looking when it finds the first one with the same name.

	  Example:

	  [1000]
	  type = endpoint

	  [1000]
	  type = aor

	  [1000](+)
	  authenticate_qualify = yes

	  This config will fail because config.c appends authenticate_qualify to the
	  first category it finds, the endpoint, and that's not valid for endpoint.

	  Solution:

	  The capability to find a category that contains a certain variable already
	  exists so the only real change was to parse anything after the '+' that's not a
	  comma, as a filter string.

	  [1000]
	  type = endpoint

	  [1000]
	  type = aor

	  [1000](+type=aor)
	  authenticate_qualify = yes

	  This now works as expected.

	  Although the following example doesn't make any sense for pjsip, you can even
	  specify multiple filters:

	  [1000](+type=aor&qualify_frequency=10)

	  ASTERISK-25868 #close
	  Reported-by: Nick Repin

	  Change-Id: I10773da4c79db36fbf1993961992af63d3441580

2016-04-05 10:21 +0000 [784fb43f43]  Joshua Colp <jcolp@digium.com>

	* res_http_websocket: Make core supported.

	  Websockets are a core part of ARI support and as such this
	  module should also be core supported.

	  Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c

2016-03-25 23:22 +0000 [4d40b161c3]  gtjoseph <george.joseph@fairview5.com>

	* stringfields:  Refactor to allow fields to be added to the end of structures

	  String fields are great, except that you can't add new ones without breaking
	  ABI compatibility because it shifts down everything else in the structure.
	  The only alternative is to add your own char * field to the end of the
	  structure and manage the memory yourself which isn't ideal, especially since
	  you then can't use the OPT_STRINGFIELD_T type.

	  Background:

	  The reason string fields had to be declared inside the
	  AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared
	  fields for initialization, compare and copy.  Since AST_DECLARE_STRING_FIELDS
	  declared the pool, then the fields, then the manager, you could use the offsets
	  of the pool and manager and iterate over the sequential addresses in between to
	  access the fields. The actual pool, field allocation and field set operations
	  don't actually care where the field is.  It's just iteration over the fields
	  that was the problem.

	  Solution: Extended String Fields

	  An extended string field is one that is declared outside the
	  AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent
	  structure.  Other than using AST_STRING_FIELD_EXTENDED instead of
	  AST_STRING_FIELD, it looks the same as other string fields.  It's storage comes
	  from the pool and it participates in string field compare and copy operations
	  peformed on the parent structure. It's also a valid target for the
	  OPT_STRINGFIELD_T aco option type.

	  Implementation:

	  To keep track of the extended fields and make sure that ABI isn't broken, the
	  existing embedded_pool pointer in the manager structure was repurposed to be a
	  pointer to a separate header structure that contains the embedded_pool pointer
	  plus a vector of fields.  The length of the manager structure didn't change and
	  the embedded_pool pointer isn't used in the macros, only the stringfields C
	  code.  A side benefit of this is that changing the header structure in the
	  future won't break ABI.

	  ast_string_fields_init initializes the normal string fields and appends them to
	  the vector, and subsequent calls to ast_string_field_init_extended initialize
	  and append the extended fields. Cleanup, ast_string_fields_cmp, and
	  ast_string_fields_copy can now work on the vector instead of sequentially
	  traversing the addresses between the pool and manager.

	  The total size of a structure using string fields didn't change, whether using
	  extended fields or not, nor have the offsets of any structure members, either
	  inside the original block or outside.  Adding an extended field to the end of a
	  structure is the same as adding a char *.

	  Details:

	  The stringfield C code was pulled out from utils.c and into stringfields.c.
	  It just made sense.

	  Additional work was done in ast_string_field_init and
	  ast_calloc_with_stringfields to handle the allocation of the new header
	  structure and the vector, and the associated cleanup.  In the process some
	  additional NULL pointer checking was added.

	  A lot of work was done in stringfields.h since the logic for compare and copy
	  is there.  Documentation was added as well as somne additional NULL checking.

	  The ability to call ast_calloc_with_stringfields with a number of structures
	  greater than 1 never really worked.  Well, the calloc worked but there was no
	  way to access the additional structures or clean them up.  It was agreed that
	  there was no use case for requesting more than 1 structure so an ast_assert
	  was added to prevent it and the iteration code removed.

	  Testing:

	  The stringfield unit tests were updated to test both normal and extended
	  fields.  Tests for ast_string_field_ptr_set_by_fields and
	  ast_calloc_with_stringfields were also added.

	  As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except
	  res_pjsip itself, saved off.  The patch was then added and a full compile and
	  install was performed.  Then the older res_pjsip_* moduled were copied over the
	  installed versions so res_pjsip was new and the rest were old.  No issues.

	  contact->aor, which is a char * at the end of contact, was then changed to an
	  extended string field and a recompile and reinstall was performed, again
	  leaving stock versions of the the res_pjsip_* modules.  Again, no issues with
	  the res_pjsip_* modules using the old stringfield implementation and with
	  contact->aor as a char *, and res_pjsip itself using the new stringfield
	  implementation and contact->aor being an extended string field.

	  Finally, several existing string fields were converted to extended string
	  fields to test OPT_STRINGFIELD_T.  Again, no issues.

	  Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61

2016-04-04 18:02 +0000 [c07e1190ec]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_mwi:  Fix segv caused by 16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7

	  I forgot the new voicemail_extension wasn't a stringfield and didn't check
	  for NULL where I should have.

	  Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb

2016-04-03 11:47 +0000 [060b7b83bc]  gtjoseph <george.joseph@fairview5.com>

	* install_prereq:  Fix check_installed_debs remove subversion

	  check_installed_debs wasn't handling virtual packages like libsrtp-dev and
	  libresample-dev and on multiarch systems it was accidentally filtering out all
	  packages if any :i386 packages were found instead of just filtering out the
	  :i386 packages themselves.

	  Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda

2016-04-01 13:09 +0000 [433d2c4bbf]  gtjoseph <george.joseph@fairview5.com>

	* utils.c:  Fix typo in handle_show_locks

	  ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e).

	  Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf

2016-03-30 18:34 +0000 [304f81780d]  gtjoseph <george.joseph@fairview5.com>

	* pjproject_bundled:  Fix use of LDCONFIG for shared library link creation

	  LDCONFIG apparently isn't set to something sane on all systems so the creation
	  of the shared library links fails.  Instead of just testing for non-blank,
	  main/Makefile now checks that LDCONFIG is actually executable and reverts to
	  LN if it isn't.

	  This applies to both libasteriskpj and libasteriskssl.

	  Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG.

	  ASTERISK-25873 #close
	  Reported-by: Hans van Eijsden

	  Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729

2016-03-29 13:47 +0000 [0ea742d33a]  Richard Mudgett <rmudgett@digium.com>

	* res_stasis: Add control ref to playback and recording structs.

	  The stasis_app_playback and stasis_app_recording structs need to have a
	  struct stasis_app_control ref.  Other threads can get a reference to the
	  playback and recording structs from their respective global container.
	  These other threads can then use the control pointer they contain after
	  the control struct has gone.

	  * Add control ref to stasis_app_playback and stasis_app_recording structs.

	  With the refs added, the control command queue can now have a circular
	  control reference which will cause the control struct to never get
	  released if the control's command queue is not flushed when the channel
	  leaves the Stasis application.  Also the command queue needs better
	  protection from adding commands if the control->is_done flag is set.

	  * Flush the control command queue on exit.

	  ASTERISK-25882 #close

	  Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d

2016-03-28 18:10 +0000 [53f63ad770]  Richard Mudgett <rmudgett@digium.com>

	* res_stasis: Fix crash on a hanging up channel.

	  * Give the struct stasis_app_control ao2 object a ref to the channel held
	  in the object.  Now the channel will still be around if a thread needs to
	  post a stasis message instead of crash because the topic was destroyed.

	  * Moved stopping any lingering silence generator out of the struct
	  stasis_app_control destructor and made it a part of exiting the Stasis
	  application.  Who knows which thread the destructor will be called under
	  so it cannot affect the channel's silence generator.  Not only was the
	  channel unprotected when the silence generator was stopped, stasis may no
	  longer even control the channel.

	  ASTERISK-25882

	  Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4

2016-03-30 13:31 +0000 [2fab4d7da8]  Richard Mudgett <rmudgett@digium.com>

	* res_stasis.c: Protect channel datastore list from stasis end.

	  Change-Id: Ifadc469590bd4d5368e19d3763db3bd1f80fdb95

2016-03-29 18:06 +0000 [ece2edaa04]  Richard Mudgett <rmudgett@digium.com>

	* res_ari: Cannot get control also means channel is unavailable.

	  The only caller of ari_bridges_play_found() has this note:

	  If ari_bridges_play_found fails because the channel is unavailable for
	  playback, The channel will be removed from the playback list soon.  We can
	  keep trying to get channels from the list until we either get one that
	  will work or else there isn't a channel for this bridge anymore, in which
	  case we'll revert to ari_bridges_play_new.

	  Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6

2016-03-29 14:29 +0000 [2f36cba4b5]  Richard Mudgett <rmudgett@digium.com>

	* res_stasis_recording.c: Cleanup stasis_app_recording_find_by_name().

	  Change-Id: Ic7d93c402c498677a122505558859c853d4e5ac7

2016-03-28 14:23 +0000 [34457dd9db]  Richard Mudgett <rmudgett@digium.com>

	* core_unreal.c: Add clarification comment about channel ref.

	  Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3

2016-03-30 12:38 +0000 [2b3261cd36]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_mwi:  Allow subscribe to vm access extension as an alias

	  Background:

	  If your extension is 1000 and the voicemail access extension is 1571 and you
	  dial 1571, usually a dialplan rule calls voicemailmain with your extension and
	  you are placed directly in your mailbox.  Therefore most admins program the
	  voicemail (or other speed dial) button on their phones to the access extension.
	  Some phones (Snom at least) use whatever is programmed there to also subscribe
	  for MWI and so can't dial one number and subscribe to another.  This works fine
	  in chan_sip because chan_sip completely ignores the user portion of the
	  SUBSCRIBE message request URI.  If it can match the peer, is subscribes to the
	  peer's mailbox.  The user could be set to anything or nothing and you'd still
	  get subscribed to your mailbox.

	  Issue:

	  chan_pjsip actually uses the user portion of the URI to find an aor and its
	  mailboxes.  Therefore a subscribe to 1571 results in a 404.  Sure, you can
	  create an aor for 1571 but you certainly can't add your entire voicemail
	  system's mailboxes to it and everyone would get notified of every MWI.

	  Solution:

	  When an MWI subscribe comes in and an aor can't be found that matches the
	  resource directly, check the resource against the endpoint's aors.  If an aor
	  is found that has a voicemail_extension that matches the resource, use it.

	  ASTERISK-25865
	  Reported-by: Ross Beer

	  Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e

2016-03-24 22:55 +0000 [e2524fcee3]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_mwi:  Add voicemail extension and mwi_subscribe_replaces_unsolicited

	  res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds
	  the Message-Account header to the MWI NOTIFY.  Also, specifying mailboxes
	  on endpoints for unsolicited mwi and on aors for subscriptions required
	  that the admin know in advance which the client wanted.  If you specified
	  mailboxes on the endpoint, subscriptions were rejected even if you also
	  specified mailboxes on the aor.

	  Voicemail extension:
	  * Added a global default_voicemail_extension which defaults to "".
	  * Added voicemail_extension to both endpoint and aor.
	  * Added ast_sip_subscription_get_dialog for support.
	  * Added ast_sip_subscription_get_sip_uri for support.

	  When an unsolicited NOTIFY is constructed, the From header is parsed, the
	  voicemail extension from the endpoint is substituted for the user, and the
	  result placed in the Message-Account field in the body.

	  When a subscribed NOTIFY is constructed, the subscription dialog local uri
	  is parsed, the voicemail_extension from the aor (looked up from the
	  subscription resource name) is substituted for the user, and the result
	  placed in the Message-Account field in the body.

	  If no voicemail extension was defined, the Message-Account field is not added
	  to the NOTIFY body.

	  mwi_subscribe_replaces_unsolicited:
	  * Added mwi_subscribe_replaces_unsolicited to endpoint.

	  The previous behavior was to reject a subscribe if a previous internal
	  subscription for unsolicited MWI was found for the mailbox.  That remains the
	  default.  However, if there are mailboxes also set on the aor and the client
	  subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal
	  subscription is removed and replaced with the external subscription.  This
	  allows an admin to configure mailboxes on both the endpoint and aor and allows
	  the client to select which to use.

	  ASTERISK-25865 #close
	  Reported-by: Ross Beer

	  Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea

2016-03-30 09:46 +0000 [724b9ab28f]  gtjoseph <george.joseph@fairview5.com>

	* res_rtp_asterisk:  Fix placement of txcount increment

	  Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount
	  for rtcp packets as well as rtp packets and that was causing sender reports
	  to be generated instead of receiver reports in cases where no rtp was actually
	  being sent.

	  Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp,
	  to rtp_sento which only handles rtp packets.

	  Discovered by the hep/rtcp-receiver test.

	  Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5

2016-03-26 22:33 +0000 [c4064727d2]  gtjoseph <george.joseph@fairview5.com>

	* chan_pjsip:  Add 'pjsip show channelstats'

	  Added the ability to show channel statistics to chan_pjsip (cli_functions.c)

	  Moved the existing 'pjsip show channel(s)' functionality from
	  pjsip_configuration to cli_functions.c.  The stats needed chan_pjsip's
	  private header so it made sense to move the existing channel commands as well.

	  Now using stasis_cache_dump to get the channel snapshots rather than retrieving
	  all endpoints, then getting each one's channel snapshots.  Much more efficient.

	  Change-Id: I03b114522126d27434030b285bf6d531ddd79869

2016-03-25 10:59 +0000 [970803efcb]  Jacek Konieczny <jkonieczny@eggsoft.pl>

	* res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS

	  Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764
	  explicitly states:

	    There MUST be a separate DTLS-SRTP session for each distinct pair of
	    source and destination ports used by a media session

	  This means RTP keying material cannot be used for DTLS RTCP, which was
	  the reason why RTCP encryption would fail.

	  ASTERISK-25642

	  Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a

2016-03-25 10:42 +0000 [9785e8d090]  Jacek Konieczny <jkonieczny@eggsoft.pl>

	* app_echo: forward and generate VIDUPDATE frames

	  When using app_echo via WebRTC with VP8 video the video would appear
	  only after a few minutes, because there would be nothing to request
	  a full reference frame.

	  This fixes the problem in both ways:
	  - echos any VIDUPDATE frames received on the channel
	  - sends one such frame when first video frame is to be forwarded

	  This makes the echo work with Firefox and Chrome WebRTC implementation.

	  ASTERISK-25867 #close

	  Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e

2016-03-27 12:53 +0000 [44ffb5105a]  gtjoseph <george.joseph@fairview5.com>

	* res_rtp_asterisk:  Fix packet stats on bridged connection

	  rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated
	  for bridged streams because the calulations were being done after the
	  bridged short-circuit.  Actually, rxoctetcount wasn't ever being calculated.

	  Moved the calculations so they occur for all valid received packets and
	  all transmitted packets.  Also added rxoctetcount and txoctetcount to
	  ast_rtp_instance_stat.

	  Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb

2016-03-10 19:52 +0000 [c971a64366]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip/pjsip_options:  Fix From generation on outgoing OPTIONS

	  No one seemed to notice but every time an OPTIONS goes out, it goes
	  out with a From of "asterisk" (or whatever the default from_user is set to),
	  even if you specify an endpoint.

	  The issue had several causes...
	  qualify_contact is only called with an endpoint if called from the CLI.
	  If the endpoint is NULL, qualify_contact only looks up the endpoint if
	  authenticate_qualify=yes. Even then, it never passes it on to
	  ast_sip_create_request where the From header is set.  Therefore From
	  is always "asterisk" (or whatever the default from_user is set to).
	  Even if ast_sip_create_request were to get an endpoint, it only sets
	  the From if endpoint->from_user is set.

	  The fix is 4 parts...

	  First, create_out_of_dialog_request was modified to use the endpoint id
	  if endpoint was specified and from_user is not set.

	  Second, qualify_contact was modified to always look up an endpoint if
	  one wasn't specified regardless of authenticate_qualify.  It then passes
	  the endpoint on to create_out_of_dialog_request.

	  Third (and most importantly), find_an_endpoint was modified to find
	  an endpoint by using an "aors LIKE %contact->aor%" predicate with
	  ast_sorcery_retrieve_by_fields.  As such, this patch will only work
	  if the sorcery realtime optimizations patch goes in.  Otherwise we'd
	  be pulling the entire endpoints database every time we send an OPTIONS.
	  Since we already know the contact's aor, the on_endpoint callback was also
	  modified to just check if the contact->aor is an exact match to one of
	  the endpoint's.

	  Finally, since we now have an endpoint for every OPTIONS request,
	  res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was
	  updated to get the transport from the endpoint and set it on tdata.
	  Now the correct transport is used.

	  Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af

2016-03-08 15:55 +0000 [c948ce9651]  gtjoseph <george.joseph@fairview5.com>

	* sorcery/res_pjsip:  Refactor for realtime performance

	  There were a number of places in the res_pjsip stack that were getting
	  all endpoints or all aors, and then filtering them locally.

	  A good example is pjsip_options which, on startup, retrieves all
	  endpoints, then the aors for those endpoints, then tests the aors to see
	  if the qualify_frequency is > 0.  One issue was that it never did
	  anything with the endpoints other than retrieve the aors so we probably
	  could have skipped a step and just retrieved all aors. But nevermind.

	  This worked reasonably well with local config files but with a realtime
	  backend and thousands of objects, this was a nightmare.  The issue
	  really boiled down to the fact that while realtime supports predicates
	  that are passed to the database engine, the non-realtime sorcery
	  backends didn't.

	  They do now.

	  The realtime engines have a scheme for doing simple comparisons. They
	  take in an ast_variable (or list) for matching, and the name of each
	  variable can contain an operator.  For instance, a name of
	  "qualify_frequency >" and a value of "0" would create a SQL predicate
	  that looks like "where qualify_frequency > '0'".  If there's no operator
	  after the name, the engines add an '=' so a simple name of
	  "qualify_frequency" and a value of "10" would return exact matches.

	  The non-realtime backends decide whether to include an object in a
	  result set by calling ast_sorcery_changeset_create on every object in
	  the internal container.  However, ast_sorcery_changeset_create only does
	  exact string matches though so a name of "qualify_frequency >" and a
	  value of "0" returns nothing because the literal "qualify_frequency >"
	  doesn't match any name in the objset set.

	  So, the real task was to create a generic string matcher that can take a
	  left value, operator and a right value and perform the match. To that
	  end, strings.c has a new ast_strings_match(left, operator, right)
	  function.  Left and right are the strings to operate on and the operator
	  can be a string containing any of the following: = (or NULL or ""), !=,
	  >, >=, <, <=, like or regex.  If the operator is like or regex, the
	  right string should be a %-pattern or a regex expression.  If both left
	  and right can be converted to float, then a numeric comparison is
	  performed, otherwise a string comparison is performed.

	  To use this new function on ast_variables, 2 new functions were added to
	  config.c.  One that compares 2 ast_variables, and one that compares 2
	  ast_variable lists.  The former is useful when you want to compare 2
	  ast_variables that happen to be in a list but don't want to traverse the
	  list.  The latter will traverse the right list and return true if all
	  the variables in it match the left list.

	  Now, the backends' fields_cmp functions call ast_variable_lists_match
	  instead of ast_sorcery_changeset_create and they can now process the
	  same syntax as the realtime engines.  The realtime backend just passes
	  the variable list unaltered to the engine.  The only gotcha is that
	  there's no common realtime engine support for regex so that's been noted
	  in the api docs for ast_sorcery_retrieve_by_fields.

	  Only one more change to sorcery was done...  A new config flag
	  "allow_unqualified_fetch" was added to reg_sorcery_realtime.
	  "no": ignore fetches if no predicate fields were supplied.
	  "error": same as no but emit an error. (good for testing)
	  "yes": allow (the default);
	  "warn": allow but emit a warning. (good for testing)

	  Now on to res_pjsip...

	  pjsip_options was modified to retrieve aors with qualify_frequency > 0
	  rather than all endpoints then all aors.  Not only was this a big
	  improvement in realtime retrieval but even for config files there's an
	  improvement because we're not going through endpoints anymore.

	  res_pjsip_mwi was modified to retieve only endpoints with something in
	  the mailboxes field instead of all endpoints then testing mailboxes.

	  res_pjsip_registrar_expire was completely refactored.  It was retrieving
	  all contacts then setting up scheduler entries to check for expiration.
	  Now, it's a single thread (like keepalive) that periodically retrieves
	  only contacts whose expiration time is < now and deletes them.  A new
	  contact_expiration_check_interval was added to global with a default of
	  30 seconds.

	  Ross Beer reports that with this patch, his Asterisk startup time dropped
	  from around an hour to under 30 seconds.

	  There are still objects that can't be filtered at the database like
	  identifies, transports, and registrations.  These are not going to be
	  anywhere near as numerous as endpoints, aors, auths, contacts however.

	  Back to allow_unqualified_fetch.  If this is set to yes and you have a
	  very large number of objects in the database, the pjsip CLI commands
	  will attempt to retrive ALL of them if not qualified with a LIKE.
	  Worse, if you type "pjsip show endpoint <tab>" guess what's going to
	  happen? :)  Having a cache helps but all the objects will have to be
	  retrieved at least once to fill the cache.  Setting
	  allow_unqualified_fetch=no prevents the mass retrieve and should be used
	  on endpoints, auths, aors, and contacts.  It should NOT be used for
	  identifies, registrations and transports since these MUST be
	  retrieved in bulk.

	  Example sorcery.conf:

	  [res_pjsip]
	  endpoint=config,pjsip.conf,criteria=type=endpoint
	  endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error

	  ASTERISK-25826 #close
	  Reported-by: Ross Beer
	  Tested-by: Ross Beer

	  Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67

2016-03-25 23:19 +0000 [8e8cf80cea]  Philip Correia

	* res_parking: Fix blind transfer dynamic lots creation.

	  Blind transfers to a recognized parking extension need to use the parker's
	  channel variable values to create the dynamic parking lot.  This is
	  because there is always only one parker while the parkee may actually be a
	  multi-party bridge.  A multi-party bridge can never supply the needed
	  channel variables to create the dynamic parking lot.  In the multi-party
	  bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and
	  channel variables are inherited by the local channel used to park the
	  bridge.

	  * In park_common_setup(), make use the parker instead of the parkee to
	  supply the dynamic parking lot channel variable values.  In all but one
	  case, the parkee is the same as the parker.  However, in the recognized
	  parking extension blind transfer scenario for a two party bridge they are
	  different channels.  For consistency, we need to use the parker channel.

	  * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the
	  local channel when blind transferring a multi-party bridge to a recognized
	  parking extension.

	  * When a local channel starts a call, the Local;2 side needs to inherit
	  the CHANNEL(parkinglot) value from Local;1.

	  The DTMF one-touch parking case wasn't even trying to create dynamic
	  parking lots before it aborted the attempt.

	  * In parking_park_call(), add missing code to create a dynamic parking
	  lot.

	  A DTMF bridge hook is documented as returning -1 to remove the hook.
	  Though the hook caller is really coded to accept non-zero.  See the
	  ast_bridge_hook_callback typedef.

	  * In feature_park_call(), don't remove the DTMF one-touch parking hook
	  because of an error.

	  ASTERISK-24605 #close
	  Reported by:  Philip Correia
	  Patches:
	        call_park.patch (license #6672) patch uploaded by Philip Correia

	  Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9

2016-03-23 14:24 +0000 [3cf714031c]  Richard Mudgett <rmudgett@digium.com>

	* res_parking: Cleanup find_channel_parking_lot_name() usage.

	  Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02

2016-03-18 14:01 +0000 [13e75ee04f]  Richard Mudgett <rmudgett@digium.com>

	* res_parking: Misc fixes.

	  res/parking/parking_applications.c:

	  * Add malloc fail checks in setup_park_common_datastore().

	  * Fix playing parking failed announcement to only happen on non-blind
	  transfers in park_app_exec().  It could never go out before because a test
	  was provedly always false.

	  res/parking/parking_bridge.c:

	  * Fix NULL tolerance in generate_parked_user() because
	  bridge_parking_push() can theoretically pass a NULL parker channel if the
	  parker channel went away for some reason.

	  * Clarify some weird code dealing with blind_transfer in
	  bridge_parking_push().

	  res/parking/parking_bridge_features.c:

	  * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel
	  which will be bulk copied to the Local;2 channel on the subsequent
	  ast_call().  The additional advantage is if the parker channel has the
	  BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed
	  to be overridden.

	  res/parking/parking_manager.c:

	  * Fix AMI Park action input range checking of the Timeout header in
	  manager_park().

	  * Reduced locking scope to where needed in manager_park().

	  res/res_parking.c:

	  * Fix some off nominal missing unlocks by eliminating the returns.

	  Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca

2014-12-15 05:23 +0000 [e2853ae337]  Philip Correia

	* res_parking: Update parking documentation for dynamic parking lots.

	  * Remove duplicate res_parking.conf courtesytone config option
	  documentation.

	  ASTERISK-24596 #close
	  Reported by:  Philip Correia

	  ASTERISK-24605
	  Reported by:  Philip Correia
	  Patches:
	        call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia

	  Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5

2016-03-25 06:02 +0000 [72a897c534]  Joshua Colp <jcolp@digium.com>

	* media_cache: Demote warning to debug as it may occur often.

	  The file playback system will now query the media cache and then
	  the old file functionality. Under normal conditions this will result
	  in the cache failing to retrieve a file causing a warning message
	  to get output each time a file is played back.

	  This change demotes this warning to a debug message.

	  Change-Id: Ib72246ba300b5cce32774bfb3c26634bfb708624
2016-03-10 16:58 +0000 [89e94e886c]  Mark Michelson <mmichelson@digium.com>

	* Restrict CLI/AMI commands on shutdown.

	  During stress testing, we have frequently seen crashes occur because a
	  CLI or AMI command attempts to access information that is in the process
	  of being destroyed.

	  When addressing how to fix this issue, we initially considered fixing
	  individual crashes we observed. However, the changes required to fix
	  those problems would introduce considerable overhead to the nominal
	  case. This is not reasonable in order to prevent a crash from occurring
	  while Asterisk is already shutting down.

	  Instead, this change makes it so AMI and CLI commands cannot be executed
	  if Asterisk is being shut down. For AMI, this is absolute. For CLI,
	  though, certain commands can be registered so that they may be run
	  during Asterisk shutdown.

	  ASTERISK-25825 #close

	  Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990

2016-03-24 14:08 +0000 [3f720155b7]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers.

	  Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those
	  codecs, which the caller did not request/support. That fix was not complete
	  because on the second Session Timer all codecs were sent again. Some VoIP/SIP
	  clients interpreted that complete codec-list as a change in the SIP session.
	  Because of that, Asterisk did not send the RTP audio via NAT anymore which
	  created a non-audio scenario after the second Session Timer fired.

	  ASTERISK-24543 #close

	  Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66

2016-03-19 07:34 +0000 [894071ea2c]  Gianluca Merlo <gianluca.merlo@gmail.com>

	* config: fix flags in uint option handler

	  The configuration unsigned integer option handler sets flags for the
	  parser as if the option should be a signed integer (PARSE_INT32),
	  leading to errors on "out of range" values. Fix flags (PARSE_UINT32).

	  A fix to res_pjsip is also present which stops invalid flags from
	  being passed when registering sorcery object fields for qualify
	  status.

	  ASTERISK-25612 #close

	  Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e

2016-03-24 07:51 +0000 [13cdf3e8a1]  Walter Doekes <walter+asterisk@wjd.nu>

	* musiconhold: Only warn if music class is not found in memory and database.

	  The log message when a MusicOnHold music class was not found was changed
	  from debug level to WARNING level in Asterisk 11.19 and 13.5.  For those
	  using realtime musiconhold, this message is wrong because it warns
	  before checking the database.

	  This changeset delays the warning until after the database has been
	  checked.

	  Reported-by: Conrad de Wet
	  ASTERISK-25444 #close

	  Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf

2016-03-24 05:48 +0000 [87c9ab97ea]  Walter Doekes <walter+asterisk@wjd.nu>

	* core/logging: Fix broken syslog levels on older glibc.

	  The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However
	  this macro is broken in older glibc (< 2.17); it would left-shift the
	  facility a second time, causing the resultant priority to become
	  invalid.

	  The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this:

	      The priority argument is formed by ORing the facility and the level
	      values [...].

	  ASTERISK-25510 #close
	  Reported by: Michael Newton

	  Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03
2016-03-24 06:18 +0000 [a72f3b5bb4]  Joshua Colp <jcolp@digium.com>

	* tests/test_http_media_cache: Fix file descriptor leak in test.

	  Change-Id: Ie8a9ae3d13bdeaacafc8d28271adc6707f633a5f

2016-02-28 19:05 +0000 [13efea24f7]  Matt Jordan <mjordan@digium.com>

	* main/app: Only look to end of file if ':end' is specified, and not just ':'

	  There is a little known feature in app_controlplayback that will cause the
	  specified offset to be used relative to the end of a file if a ':end' is
	  detected within the filename.

	  This feature is pretty bad, but okay.

	  However, a bug exists in this code where a ':' detected in the filename
	  will cause the end pointer to be non-NULL, even if the full ':end' isn't
	  specified. This causes us to treat an unspecified offset (0) as being
	  "start playing from the end of the file", resulting in no file playback
	  occurring.

	  This patch fixes this bug by resetting the end pointer if ':end' is not
	  found in the filename.

	  Change-Id: Ib4c7b1b45283e4effd622a970055c51146892f35

2015-12-26 15:29 +0000 [ca14b99e6e]  Matt Jordan <mjordan@digium.com>

	* main/file: Add the ability to play media in the media cache

	  This patch allows applications/APIs that access media through the core file
	  APIs to play media in the media cache. Prior to determining if a 'filename'
	  exists, the filename is passed to the media cache's retrieve API call. If
	  that call succeeds, the local file specified passed back by the API is
	  opened for streaming. When used in this fashion, the 'filename' is actually
	  a URI that the media cache process and understand.

	  ASTERISK-25654 #close

	  Change-Id: I73b6e2e90c3e91b8500581c45cdf9c0dc785f5f0

2015-12-30 10:52 +0000 [01962a3932]  Matt Jordan <mjordan@digium.com>

	* tests/test_http_media_cache: Add unit tests for res_http_media_cache

	  This patch adds unit tests for res_http_media cache, that covers nominal
	  creation and retrieval - and through them as well, staleness and deletion
	  checks. In addition, this patch adds tests that covers the interaction of
	  various HTTP headers, including Expires, Etag, and Cache-Control.

	  ASTERISK-25654

	  Change-Id: I2db101e307c863857fe416d6f5bf4cace9ac7cf5

2015-01-29 08:38 +0000 [22e2340813]  Matt Jordan <mjordan@digium.com>

	* res/res_http_media_cache: Add an HTTP(S) backend for the core media cache

	  This patch adds a bucket backend for the core media cache that interfaces to a
	  remote HTTP server. When a media item is requested in the cache, the cache will
	  query its bucket backends to see if they can provide the media item. If that
	  media item has a scheme of HTTP or HTTPS, this backend will be invoked.

	  The backend provides callbacks for the following:
	   * create - this will always retrieve the URI specified by the provided
	              bucket_file, and store it in the file specified by the object.
	   * retrieve - this will pull the URI specified and store it in a temporary
	                file. It is then up to the media cache to move/rename this file
	                if desired.
	   * delete - destroys the file associated with the bucket_file.
	   * stale - if the bucket_file has expired, based on received HTTP headers from
	             the remote server, or if the ETag on the server no longer matches
	             the ETag stored on the bucket_file, the resource is determined to be
	             stale.

	  Note that the backend respects the ETag, Expires, and Cache-Control headers
	  provided by the HTTP server it is querying.

	  ASTERISK-25654

	  Change-Id: Ie201c2b34cafc0c90a7ee18d7c8359afaccc5250

2015-12-26 15:31 +0000 [791b4c9f81]  Matt Jordan <mjordan@digium.com>

	* main/media_cache: Provide an extension on the local file associated with a URI

	  This patch does the following:

	  First, it addresses file extension handling in the media cache. The media core
	  in Asterisk is a bit interesting in that it wants:
	   * A file to have an extension on it. That extension is used to associate the
	     file with a defined format module.
	   * The filename passed to the core to not have an extension on it. This allows
	     the core to match the available file formats with the format a channel
	     is capable of handling.

	  Unfortunately, this makes the current implementation a bit lacking in the media
	  cache. By default, we do not store the extension of a retrieved URI on the
	  local file that is created. As a result, the media core does not know what
	  format the file is, and the file is ignored. Modifying the file outside of the
	  media core is bad, as we would not be able to update the internal
	  ast_bucket_file's path.

	  At the same time, we do not want to pass the extension out in the file_path
	  parameter in ast_media_cache_retrieve. This parameter is intended to be fed
	  into the media core; if we passed the extension, all callers would have to
	  strip it off.

	  Thus, this patch does the following:
	  * If there is an extension specified in the URL, we append it to the local
	    file name (if a preferred file name isn't specified), and we store that
	    in the local file path.
	  * The extension, however, is stripped off of the file_path parameter passed
	    back out of ast_media_cache_retrieve.

	  Second, this patch causes stale items to be completely removed from the system.
	  Prior to this patch, sound files could be orphaned due to the bucket
	  referencing the file being deleted, but the file itself not being removed. This
	  is now addressed by explicitly calling ast_bucket_file_delete on the
	  bucket_file when it is deemed to be stale. Note that this only happen when we
	  know we will attempt to retrieve the resource again.

	  Finally, this patch changes the AO2 container holding media items to just use
	  a regular mutex. The usage for this container already assumed it was a plain
	  mutex, and - given that retrieval of an item can cause it to be replaced in
	  the container - a mutex makes more sense than a read/write lock.

	  Change-Id: I51667fff86ae8d2e4a663555dfa85b11e935fe0f

2014-10-25 20:21 +0000 [6bbcfb34bd]  Matt Jordan <mjordan@digium.com>

	* funcs/func_curl: Add the ability for CURL to download and store files

	  This patch adds a write option to the CURL dialplan function, allowing it to
	  CURL files and store them locally. The value 'written' to the CURL URL
	  specifies the location on disk to store the file. As an example:

	  same => n,Set(CURL(http://1.1.1.1/foo.wav)=/tmp/foo.wav)

	  Would retrieve the file foo.wav from the remote server and store it in the
	  /tmp directory.

	  Due to the potentially dangerous nature of this function call, APIs are
	  forbidden from using the write functionality unless live_dangerously is set
	  to True in asterisk.conf.

	  ASTERISK-25652 #close

	  Change-Id: I44f4ad823d7d20f04ceaad3698c5c7f653c41b0d

2016-03-23 08:59 +0000 [392341ba37]  gtjoseph <george.joseph@fairview5.com>

	* pjproject-bundled:  Cleanups for reported issues

	  PortAudio should no longer be required
	  PJSIP_MAX_PKT_LEN is now 6000
	  Older autoconf issue fixed. (CentOS 6)

	  Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd

2015-11-20 08:02 +0000 [ac66999971]  Francesco Castellano <francesco.castellano@messagenet.it>

	* chan_sip.c: Space after port causes unnecessary resolution attempt

	  check_via() already skips leading blanks where the sent-by address (with the
	  optional port) should be placed.

	  Since RFC 3261 allows for blanks between the port ant the Via parameters:
	  > https://tools.ietf.org/html/rfc3261#section-20.42
	  (actually it allows a lot of blanks more ;-)). I just switched from
	  ast_skip_blanks() to ast_strip() on the local copy of the string.

	  ASTERISK-21301 #close

	  Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06
2016-03-19 17:49 +0000 [1d3191b118]  gtjoseph <george.joseph@fairview5.com>

	* progdocs:  Exclude ./third-party from documentation generation

	  We don't need pjproject's documentation embedded in Asterisk's.

	  Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17

2016-03-18 20:32 +0000 [8f94f947f5]  Gianluca Merlo <gianluca.merlo@gmail.com>

	* func_aes: fix misuse of strlen on binary data

	  The encryption code for AES_ENCRYPT evaluates the length of the data to
	  be encoded in base64 using strlen. The data is binary, thus the length
	  of it can be underestimated at the first NULL character.
	  Reuse the write pointer offset to evaluate it, instead.

	  ASTERISK-25857 #close

	  Change-Id: If686b5d570473eb926693c73461177b35b13b186
2016-03-18 14:31 +0000 [a3c9a74a02]  Kevin Harwell <kharwell@digium.com>

	* chan_pjsip: ref leak when checking direct_media_glare

	  Fix the reference leak introduced in the following commit:

	  c534bd58075e2e1a1e4f3b23c435186c71b155fd

	  ASTERISK-25849

	  Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85
2016-03-16 12:37 +0000 [c534bd5807]  Kevin Harwell <kharwell@digium.com>

	* chan_pjsip: transfers with direct media reinvite has wrong address/port

	  During a transfer involving direct media a race occurs between when the
	  transferer channel is swapped out, initiating rtp changes/updates, and the
	  subsequent reinvites.

	  When Alice, after speaking with Charlie (Bob is on hold), connects Bob and
	  Charlie invites are sent to each in order to establish the call between them.
	  Bob is taken off hold and Charlie is told to have his media flow through
	  Asterisk. However, if before those invites go out the bridge updates Bob's
	  and/or Charlie's rtp information with direct media data (i.e. address, port)
	  then the invite(s) will contain the remote data in the SDP instead of the
	  Asterisk data.

	  The race occurs in the native bridge glue code when updating the peer. The
	  direct_media_address can get set twice before sending out the first invite
	  during call connection. This can happen because the checking/setting of the
	  direct_media_address happened in one thread while the sending of the invite(s)
	  happened in another thread.

	  This fix removes the race condition by moving the checking/setting of the
	  direct_media_address to be in the same thread as the sending of the invites(s).
	  This serializes the checking/setting and sending so they can no longer happen
	  out of order.

	  ASTERISK-25849 #close

	  Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873

2016-03-03 04:43 +0000 [bdccb81157]  Sergio Medina Toledo <lumasepa@gmail.com>

	* res_pjsip_refer.c: Fix seg fault in process of Refer-to header.

	  The "Refer-to" header of an incoming REFER request is parsed by
	  pjsip_parse_uri().  That function requires the URI parameter to be NULL
	  terminated.  Unfortunately, the previous code added the NULL terminator by
	  overwriting memory that may not be safe.  The overwritten memory results
	  could be benign, memory corruption, or a segmentation fault.  Now the URI
	  is NULL terminated safely by copying the URI to a new chunk of memory with
	  the correct size to be NULL terminated.

	  ASTERISK-25814 #close

	  Change-Id: I32565496684a5a49c3278fce06474b8c94b37342

2016-02-25 10:29 +0000 [0da36fca6b]  Leif Madsen <leif@leifmadsen.com>

	* Add initial support to build Docker images

	  This work-in-progress is the first step to being able to reliably
	  build Asterisk containers from the Asterisk source. I'm submitting
	  this based on feedback gained at AstriDevCon 2015.

	  Information about how to use this is provided in contrib/docker/README.md
	  and will result in a local Asterisk container being built right from
	  your source. I believe this can eventually be automated via
	  hub.docker.com.

	  Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1

2016-03-11 12:22 +0000 [810f92c9dc]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix mwi resub deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  ASTERISK-25023 #close

	  Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6

2016-03-10 17:01 +0000 [72c444ba37]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix registration timeout and expire deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  ASTERISK-25023

	  Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508

2016-03-09 16:26 +0000 [7ea1e181dc]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix waitid deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  * Made always run check_pendings() under the scheduler thread so scheduler
	  ids can be checked safely.

	  ASTERISK-25023

	  Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52

2016-03-10 12:17 +0000 [fbf8e04aed]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix t38id deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  ASTERISK-25023

	  Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f

2016-03-08 15:08 +0000 [02458cc6fd]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix session timers deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  ASTERISK-25023

	  Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900

2016-03-09 16:34 +0000 [c7fdff2e37]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix reinviteid deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  ASTERISK-25023

	  Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1

2016-03-07 13:21 +0000 [69810b306d]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix autokillid deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  * Fix clearing autokillid in __sip_autodestruct() even though we could
	  reschedule.

	  ASTERISK-25023

	  Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f

2016-03-09 16:32 +0000 [f484ddbdfe]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix packet retransid deadlock potential.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  * Fix retrans_pkt() to call check_pendings() with both the owner channel
	  and the private objects locked as required.

	  * Refactor dialog retransmission packet list to safely remove packet
	  nodes.  The list nodes are now ao2 objects.  The list has a ref and the
	  scheduled entry has a ref.

	  ASTERISK-25023

	  Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641

2016-03-07 18:28 +0000 [67c79c326d]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix provisional_keepalive_sched_id deadlock.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Stopping a scheduled event can result in a deadlock if the scheduled event
	  is running when you try to stop the event.  If you hold a lock needed by
	  the scheduled event while trying to stop the scheduled event then a
	  deadlock can happen.  The general strategy for resolving the deadlock
	  potential is to push the actual starting and stopping of the scheduled
	  events off onto the scheduler/do_monitor() thread by scheduling an
	  immediate one shot scheduled event.  Some restructuring may be needed
	  because the code may assume that the start/stop of the scheduled events is
	  immediate.

	  ASTERISK-25023

	  Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48

2016-03-09 11:22 +0000 [76be7093cd]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  * Make dialog_unlink_all() unschedule all items at once in the sched
	  thread.

	  ASTERISK-25023

	  Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4

2016-03-10 21:54 +0000 [52f0932e4c]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Clear scheduled immediate events on unload.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  The reordering of chan_sip's shutdown is to handle any immediate events
	  that get put onto the scheduler so resources aren't leaked.  The typical
	  immediate events at this time are going to be concerned with stopping
	  other scheduled events.

	  ASTERISK-25023

	  Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20

2016-03-15 14:51 +0000 [0987a11cce]  Richard Mudgett <rmudgett@digium.com>

	* sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  Delaying destruction of the chan_sip sip_pvt structures caused the
	  /channels/chan_sip/test_sip_rtpqos unit test to crash.  That test
	  registers a special test ast_rtp_engine with the rtp engine module.  When
	  the unit test completes it cleans up by unregistering the test
	  ast_rtp_engine and exits.  Since the delayed destruction of the sip_pvt
	  happens after the unit test returns, the destructor tries to call the rtp
	  engine destroy callback of the test ast_rtp_engine auto variable which no
	  longer exists on the stack.

	  * Change the test ast_rtp_engine auto variable to a static variable.  Now
	  the variable can still exist after the unit test exits so the delayed
	  sip_pvt destruction can complete successfully.

	  ASTERISK-25023

	  Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13

2016-03-07 15:50 +0000 [9a7cfa2b61]  Richard Mudgett <rmudgett@digium.com>

	* sched.c: Ensure oldest expiring entry runs first.

	  This patch is part of a series to resolve deadlocks in chan_sip.c.

	  * Updated sched unit test to check new behavior.

	  ASTERISK-25023

	  Change-Id: Ib69437327b3cda5e14c4238d9ff91b2531b34ef3

2016-03-15 13:31 +0000 [7964e260d3]  Andrew Nagy <andrew.nagy@the159.com>

	* app_stasis: Don't hang up if app is not registered

	  This prevents pbx_core from hanging up the channel if the app isn't
	  registered.

	  ASTERISK-25846 #close

	  Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce
2016-03-07 18:56 +0000 [cb97198ca6]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Simplify sip_pvt destructor call levels.

	  Remove destructor calling destroy_it calling really_destroy_it
	  for no benefit.  Just make the destructor the really_destroy_it
	  function.

	  Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a

2016-03-04 18:25 +0000 [8be01398d9]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full().

	  Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12

2016-03-14 08:59 +0000 [4df7b3ae80]  Joshua Colp <jcolp@digium.com>

	* build: Add configure check for proto field of PJSIP TLS transport setting.

	  Older versions of PJSIP do not have the proto field on the TLS transport
	  setting structure. This change adds a configure check so even if it is
	  not present we will still be able to build.

	  Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9

2016-03-12 16:02 +0000 [0af6b5de62]  gtjoseph <george.joseph@fairview5.com>

	* build_system:  Split COMPILE_DOUBLE from DONT_OPTIMIZE

	  I can't ever recall actually needing the intermediate files or the checking
	  that a double compile produces.  What I CAN remember is every DONT_OPTIMIZE
	  build needing 3 invocations of gcc instead of 1 just to do the checks and
	  produce those intermediate files.

	  Having said that, Richard pointed out that the reason for the double compile
	  was that there were cases in the past where a submitted patch failed to compile
	  because the submitter never tried it with the optimizations turned on.

	  To get the best of both worlds, COMPILE_DOUBLE has been split into its own
	  option.  If DONT_OPTIMIZE is turned on, COMPILE_DOUBLE will also be selected
	  BUT you can then turn it off if all you need are the debugging symbols.  This
	  way you have to make an informed decision about disabling COMPILE_DOUBLE.

	  To allow COMPILE_DOUBLE to be both auto-selected and turned off, a new feature
	  was added to menuselect.  The <use> element can now contain an "autoselect"
	  attribute which will turn the used member on but not create a hard dependency.
	  The cflags.xml implementation for COMPILE_DOUBLE looks like this...

	  <member name="DONT_OPTIMIZE" displayname="Disable Optimizations ...">
	  	<use autoselect="yes">COMPILE_DOUBLE</use>
	  	<support_level>core</support_level>
	  </member>
	  <member name="COMPILE_DOUBLE" displayname="Pre-compile with ...>
	  	<depend>DONT_OPTIMIZE</depend>
	  	<support_level>core</support_level>
	  </member>

	  When DONT_OPTIMIZE is turned on, COMPILE_DOUBLE is turned on because
	  of the use.
	  When DONT_OPTIMIZE is turned off, COMPILE_DOUBLE is turned off because
	  of the depend.
	  When COMPILE_DOUBLE is turned on, DONT_OPTIMIZE is turned on because
	  of the depend.
	  When COMPILE_DOUBLE is turned off, DONT_OPTIMIZE is left as is because
	  it only uses COMPILE_DOUBLE, it doesn't depend on it.

	  I also made a few tweaks to the ncurses implementation to move things
	  left a bit to allow longer descriptions.

	  Change-Id: Id49ca930ac4b5ec4fc2d8141979ad888da7b1611

2016-03-10 13:09 +0000 [638133131a]  gtjoseph <george.joseph@fairview5.com>

	* pjproject:  Pass (dont_)optimize flags to pjproject and fix pjsua

	  The pjproject Makefile now uses the Asterisk optimization flags which
	  are determined by the setting of the DONT_OPTMIZE menuselect flag.
	  The Makefile was also restructured so a change to the top level
	  menuselect.makeopts will result in a rebuild of pjproject.

	  Also, "--disable-resample" was removed from the pjproject configure
	  options.  Without resample, pjsua (which is used by the testsuite)
	  can't make audio calls.  When it can't, it segfaults.

	  Change-Id: I24b0a4d0872acef00ed89b3c527a713ee4c2ccd4

2016-03-11 16:03 +0000 [dcb25bb057]  Walter Doekes <walter+asterisk@wjd.nu>

	* app_chanspy: Fix occasional deadlock with ChanSpy and Local channels.

	  Channel masquerading had a conflict with autochannel locking.

	  When locking autochannel->channel, the channel is fetched from the
	  autochannel and then locked. During the fetch, the autochannel -- which
	  has no locks itself -- can be modified by someone who owns the channel
	  lock. That means that the value of autochan->channel cannot be trusted
	  until you hold the lock.

	  In practice, this caused problems with Local channels getting
	  masqueraded away while the ChanSpy attempted to get info from that
	  channel. The old channel which was about to get removed got locked, but
	  the new (replaced) channel got unlocked (no-op). Because the replaced
	  channel was now locked (and would never get unlocked), it couldn't get
	  removed from the channel list in a timely manner, and would now cause
	  deadlocks when iterating over the channel list.

	  This change checks the autochannel after locking the channel for changes
	  to the autochannel. If the channel had been changed, the lock is
	  reobtained on the new channel.

	  In theory it seems possible that after this fix, the lock attempt on the
	  old (wrong) channel can be on an already destroyed lock, maybe causing
	  a crash. But that hasn't been observed in the wild and is harder induce
	  than the current deadlock.

	  Thanks go to Filip Frank for suggesting a fix similar to this and
	  especially to IRC user hexanol for pointing out why this deadlock was
	  possible and testing this fix. And to Richard for catching my rookie
	  while loop mistake ;)

	  ASTERISK-25321 #close

	  Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def

2016-03-07 21:34 +0000 [fb28049de2]  gtjoseph <george.joseph@fairview5.com>

	* pjproject_bundled: Remove --with-external-pa from configure options.

	  Not sure why it was there in the first place as we already specify
	  --disable-sound.

	  Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9

2016-03-06 14:38 +0000 [d2eb65f71e]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Strip spaces from items parsed from comma-separated lists

	  Configurations like "aors = a, b, c" were either ignoring everything after "a"
	  or trying to look up " b".  Same for mailboxes,  ciphers, contacts and a few
	  others.

	  To fix, all the strsep(&copy, ",") calls have been wrapped in ast_strip.  To
	  facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were
	  updated to handle null pointers.

	  In some cases, an ast_strlen_zero() test was added to skip consecutive commas.

	  There was also an attempt to ast_free an ast_strdupa'd string in
	  ast_sip_for_each_aor which was causing a SEGV.  I removed it.

	  Although this issue was reported for realtime, the issue was in the res_pjsip
	  modules so all config mechanisms were affected.

	  ASTERISK-25829 #close
	  Reported-by: Mateusz Kowalski

	  Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2

2016-03-07 02:02 +0000 [f690c105f3]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* res_odbc_transaction: fix some format tab

	  Change-Id: I265e4ac47c629c9a63dd86b59df82a7ab3c64384

2016-02-17 22:58 +0000 [0ec9fe5421]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* main/cli.c: Refactor function to print seconds formatted

	  Refactor and created function ast_cli_print_timestr_fromseconds to print
	  seconds formatted:  year(s) week(s) day(s) hour(s) second(s)

	  This function now is used in addons/cdr_mysql.c,cdr_pgsql.c, main/cli.c,
	  res_config_ldap.c, res_config_pgsql.c.

	  Change-Id: Ibeb8634102cd11d3f8623398b279cb731bcde36c

2016-03-04 20:37 +0000 [471ff375fd]  gtjoseph <george.joseph@fairview5.com>

	* install_prereq: Add packages for bundled pjproject

	  RedHat/CentOS needs python-devel
	  Debian/Ubuntu needs automake, libsrtp-dev and python-dev

	  Ubuntu also needed libncurses5-dev for cmenuselect so while not
	  needed for pjproject, I adedd it anyway.

	  Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089

2016-02-24 17:25 +0000 [2b9849625c]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited

	  Per RFC3325, the 'From' header is now anonymized on outgoing calls when
	  caller id presentation is prohibited.

	  TID = trust_id_outbound
	  PRO = Set(CALLERID(pres)=prohib)
	  USR = endpoint/from_user
	  DOM = endpoint/from_domain
	  PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes)

	  Conditions          |Result
	  --------------------|----------------------------------------------------
	  TID PRO USR DOM     |PAI    FROM
	  --------------------|----------------------------------------------------
	  Y   Y   abc def.ghi |PRI    "Anonymous" <sip:abc@def.ghi>
	  Y   Y   abc         |PRI    "Anonymous" <sip:abc@anonymous.invalid>
	  Y   Y       def.ghi |PRI    "Anonymous" <sip:anonymous@def.ghi>
	  Y   Y               |PRI    "Anonymous" <sip:anonymous@anonymous.invalid>

	  Y   N   abc def.ghi |YES    <sip:abc@def.ghi>
	  Y   N   abc         |YES    <sip:abc@<ip_address>>
	  Y   N       def.ghi |YES    "Caller Name" <sip:<caller_exten>@def.ghi>
	  Y   N               |YES    "Caller Name" <sip:<caller_exten>@<ip_address>>

	  N   Y   abc def.ghi |NO     "Anonymous" <sip:abc@def.ghi>
	  N   Y   abc         |NO     "Anonymous" <sip:abc@anonymous.invalid>
	  N   Y       def.ghi |NO     "Anonymous" <sip:anonymous@def.ghi>
	  N   Y               |NO     "Anonymous" <sip:anonymous@anonymous.invalid>

	  N   N   abc def.ghi |YES    <sip:abc@def.ghi>
	  N   N   abc         |YES    <sip:abc@<ip_address>>
	  N   N       def.ghi |YES    "Caller Name" <sip:<caller_exten>@def.ghi>
	  N   N               |YES    "Caller Name" <sip:<caller_exten>@<ip_address>>

	  ASTERISK-25791 #close
	  Reported-by: Anthony Messina

	  Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9

2016-03-03 17:34 +0000 [37472f7398]  gtjoseph <george.joseph@fairview5.com>

	* third_party/Makefile.rules:  Replace unsupported != operator with $(shell ...)

	  Apparently the != operator is fairly new so I've replaced it with
	  the old $(shell ...) syntax.

	  Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479
	  Reported-by: Richard Mudgett

2016-01-23 15:50 +0000 [195100e770]  gtjoseph <george.joseph@fairview5.com>

	* loader: Retry dlopen when loading fails

	  Although we use the RTLD_LAZY flag when calling dlopen
	  the first time on a module, this only defers resolution
	  for function calls.  Pointer references to functions are
	  determined at link time so dlopen expects them to be there.
	  Since we don't cross-module link, pointers to functions
	  in other modules won't be available and dlopen will fail.

	  Doing a "hardened" build also causes problems because it
	  typically sets "-z now" on the ld command line which
	  overrides RTLD_LAZY at run time.

	  If the failing module isn't a GLOBAL_SYMBOLS module, then
	  dlopen will be called again after all the GLOBAL_SYMBOLS
	  modules have been loaded and they'll eventually resolve.

	  If the calling module IS a GLOBAL_SYMBOLS module itself
	  and a third module depends on it, then there's an issue
	  because the second time through the dlopen loop,
	  GLOBAL_SYMBOLS modules aren't given any special treatment
	  and since the order in which dlopen is called isn't
	  deterministic, the dependent may again be tried before the
	  module it needs is loaded.

	  Simple solution:  Save modules that fail load_resource
	  because of a dlopen error in a list and retry them
	  immediately after the first pass. Keep retrying until
	  the failed list is empty or we reach a #defined max
	  retries. Error messages are suppressed until the final
	  pass which also gets rid of those confusing error messages
	  about module failures that are later corrected.

	  Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb

2016-03-01 16:18 +0000 [15c5743ac1]  Kevin Harwell <kharwell@digium.com>

	* bridge.c: Crash during attended transfer when missing a local channel half

	  It's possible for the transferer channel to get hung up early during the
	  attended transfer process. For instance, a phone may send a "bye" immediately
	  upon receiving a sip notify that contains a sip frag 100 (I'm looking at you
	  Jitsi). When this occurs a race begins between the transferer being hung up
	  and completion of the transfer code.

	  If the channel hangs up too early during a transfer involving stasis bridging
	  for instance, then when the created local channel goes to look up its swap
	  channel (and associated datastore) it can't find it (since it is no longer in
	  the bridge) thus it fails to enter the stasis application. Consequently, the
	  created local channel(s) hang up as well. If the timing is just right then the
	  bridging code attempts to add the message link with missing local channel(s).
	  Hence the crash.

	  Unfortunately, there is no great way to solve the problem of the unexpected
	  "bye". While we can't guarantee we won't receive an early hangup, and in this
	  case still fail to enter the stasis application, we can make it so asterisk
	  does not crash.

	  This patch does just that by locking the local channel structure, checking
	  that the local channel's peer has not been lost, and then continuing. This
	  keeps the local channel's peer from being ripped out from underneath it by
	  the local/unreal hangup code while attempting to set the stasis message link.

	  ASTERISK-25771

	  Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880

2016-03-01 18:08 +0000 [0d2ccbca62]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100

	  During the transfer process, some phones (okay it was the Jitsi softphone,
	  but maybe others are out there) send a "bye" immediately after receiving a
	  SIP Notify. When a "bye" is received early for some types of transfers the
	  transferer channel may no longer be available during late stage transfer
	  processing.

	  For instance, during an attended transfer involving stasis bridging at one
	  point the created local channel looks for an associated swap channel in
	  order to retrieve the stasis application name. If the transferer has hung
	  up then the local channel will fail to find it. The local channel then has
	  no way to know which stasis app to enter, so it fails and hangs up as well.
	  Thus the transfer does not complete as expected.

	  This patch delays the sending of the initial notify in order to give the
	  transfer process enough time to gather the necessary data for a successful
	  transfer.

	  ASTERISK-25771

	  Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16

2016-03-03 08:26 +0000 [6af7fc4c37]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_dtmf_info: NULL terminate the message body.

	  PJSIP does not ensure that when printing the message body the
	  buffer will be NULL terminated. This is problematic when searching
	  for the signal and duration values of the DTMF.

	  This change ensures the buffer is always NULL terminated.

	  Change-Id: I52653a1a60c93092d06af31a27408d569cc98968

2016-03-01 20:03 +0000 [b8b7c2e428]  gtjoseph <george.joseph@fairview5.com>

	* alembic: Fix downgrade and tweak for sqlite

	  Downgrade had a few issues.  First there was an errant 'update' statement in
	  add_auto_dtmf_mode that looks like it was a copy/paste error.  Second, we
	  weren't cleaning up the ENUMs so subsequent upgrades on postgres failed
	  because the types already existed.

	  For sqlite...  sqlite doesn't support ALTER or DROP COLUMN directly.
	  Fortunately alembic batch_operations takes care of this for us if we
	  use it so the alter and drops were converted to use batch operations.

	  Here's an example downgrade:

	      with op.batch_alter_table('ps_endpoints') as batch_op:
	          batch_op.drop_column('tos_audio')
	          batch_op.drop_column('tos_video')
	          batch_op.add_column(sa.Column('tos_audio', yesno_values))
	          batch_op.add_column(sa.Column('tos_video', yesno_values))
	          batch_op.drop_column('cos_audio')
	          batch_op.drop_column('cos_video')
	          batch_op.add_column(sa.Column('cos_audio', yesno_values))
	          batch_op.add_column(sa.Column('cos_video', yesno_values))

	      with op.batch_alter_table('ps_transports') as batch_op:
	          batch_op.drop_column('tos')
	          batch_op.add_column(sa.Column('tos', yesno_values))
	      # Can't cast integers to YESNO_VALUES, so dropping and adding is required
	          batch_op.drop_column('cos')
	          batch_op.add_column(sa.Column('cos', yesno_values))

	  Upgrades from base to head and downgrades from head to base were tested
	  repeatedly for postgresql, mysql/mariadb, and sqlite3.

	  Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8

2016-03-02 15:55 +0000 [7b71bca8a4]  gtjoseph <george.joseph@fairview5.com>

	* config_transport:  Fix objects returned by ast_sip_get_transport_states

	  ast_sip_get_transport_states was returning a container of internal_state
	  objects instead of ast_sip_transport_state objects.  This was causing
	  transport lookups to fail, most noticably in res_pjsip_nat, which
	  couldn't find the correct external addresses.  This was causing contacts
	  to go out with internal ip addresses.

	  ASTERISK-25830 #close
	  Reported-by: Sean Bright

	  Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e

2016-03-02 11:17 +0000 [0a3f0e85ac]  Scott Griepentrog <scott@griepentrog.com>

	* CHAOS: cleanup possible null vars on msg alloc failure

	  In message.c, if msg_alloc fails to init the string field,
	  vars may be null, so use a null tolerant cleanup.

	  In res_pjsip_messaging.c, if msg_data_create fails, mdata
	  will be null, so use a null tolerant cleanup.

	  ASTERISK-25323

	  Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56

2016-03-02 09:34 +0000 [60aa871be3]  Scott Griepentrog <scott@griepentrog.com>

	* CHAOS: prevent crash on failed strdup

	  This patch avoids crashing on a null pointer
	  if the strdup() allocation fails.

	  ASTERISK-25323

	  Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5

2016-02-29 18:11 +0000 [0bdbf0d882]  Richard Mudgett <rmudgett@digium.com>

	* func_callerid.c: Update REDIRECTING reason documentation.

	  Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386

2016-02-26 18:57 +0000 [25de01f301]  Richard Mudgett <rmudgett@digium.com>

	* SIP diversion: Fix REDIRECTING(reason) value inconsistencies.

	  Previous chan_sip behavior:

	  Before this patch chan_sip would always strip any quotes from an incoming
	  reason and pass that value up as the REDIRECTING(reason).  For an outgoing
	  reason value, chan_sip would check the value against known values and
	  quote any it didn't recognize.  Incoming 480 response message reason text
	  was just assigned to the REDIRECTING(reason).

	  Previous chan_pjsip behavior:

	  Before this patch chan_pjsip would always pass the incoming reason value
	  up as the REDIRECTING(reason).  For an outgoing reason value, chan_pjsip
	  would send the reason value as passed down.

	  With this patch:

	  Both channel drivers match incoming reason values with values documented
	  by REDIRECTING(reason) and values documented by RFC5806 regardless of
	  whether they are quoted or not.  RFC5806 values are mapped to the
	  equivalent REDIRECTING(reason) documented value and is set in
	  REDIRECTING(reason).  e.g., an incoming RFC5806 'unconditional' value or a
	  quoted string version ('"unconditional"') is converted to
	  REDIRECTING(reason)'s 'cfu' value.  The user's dialplan only needs to deal
	  with 'cfu' instead of any of the aliases.

	  The incoming 480 response reason text supported by chan_sip checks for
	  known reason values and if not matched then puts quotes around the reason
	  string and assigns that to REDIRECTING(reason).

	  Both channel drivers send outgoing known REDIRECTING(reason) values as the
	  unquoted RFC5806 equivalent.  User custom values are either sent as is or
	  with added quotes if SIP doesn't allow a character within the value as
	  part of a RFC3261 Section 25.1 token.  Note that there are still
	  limitations on what characters can be put in a custom user value.  e.g.,
	  embedding quotes in the middle of the reason string is silly and just
	  going to cause you grief.

	  * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
	  e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
	  'cfu' value.

	  * Added missing malloc() NULL return check in res_pjsip_diversion.c
	  set_redirecting_reason().

	  * Fixed potential read from a stale pointer in res_pjsip_diversion.c
	  add_diversion_header().  The reason string needed to be copied into the
	  tdata memory pool to ensure that the string would always be available.
	  Otherwise, if the reason string returned by reason_code_to_str() was a
	  user's reason string then the string could be freed later by another
	  thread.

	  Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87

2016-02-26 18:54 +0000 [8c8ef4efb0]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason.

	  Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd

2016-02-29 20:41 +0000 [75ec137e91]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref.

	  * Fix double unref of other_party channel in off nominal path.

	  * This is unlikely to be a real problem.  However, for safety,
	  in handle_incoming_request() keep the datastore ref with the
	  other_party channel ref until we are finished with the other_party
	  channel.

	  Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821

2016-01-18 21:54 +0000 [3173e91bab]  gtjoseph <george.joseph@fairview5.com>

	* build-system: Allow building with static pjproject

	  Background here:
	  http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html

	  From CHANGES:
	   * To help insure that Asterisk is compiled and run with the same known
	     version of pjproject, a new option (--with-pjproject-bundled) has been
	     added to ./configure.  When specified, the version of pjproject specified
	     in third-party/versions.mak will be downloaded and configured.  When you
	     make Asterisk, the build process will also automatically build pjproject
	     and Asterisk will be statically linked to it.  Once a particular version
	     of pjproject is configured and built, it won't be configured or built
	     again unless you run a 'make distclean'.

	     To facilitate testing, when 'make install' is run, the pjsua and pjsystest
	     utilities and the pjproject python bindings will be installed in
	     ASTDATADIR/third-party/pjproject.

	     The default behavior remains building with the shared pjproject
	     installation, if any.

	  Building:

	     All you have to do is include the --with-pjproject-bundled option on
	     the ./configure command line (and remove any existing --with-pjproject
	     option if specified).  Everything else is automatic.

	  Behind the scenes:

	     The top-level Makefile was modified to include 'third-party' in the
	     list of MOD_SUBDIRS.

	     The third-party directory was created to contain any third party
	     packages that may be needed in the future.  Its Makefile automatically
	     iterates over any subdirectories passing on targets.

	     The third-party/pjproject directory was created to house the pjproject
	     source distribution.  Its Makefile contains targets to download, patch
	     configure, generate dependencies, compile libs, apps and python bindings,
	     sanitized build.mak and generate a symbols list.

	     When bootstrap.sh is run, it automatically includes the configure.m4
	     file in third-party/pjproject.  This file has a macro to download and
	     conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR
	     and PJPROJECT_BUNDLED.  It also tests for the capabilities like
	     PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to
	     trying to compile.  Of course, bootstrap.sh is only run once and the
	     configure file is incldued in the patch.

	     When configure is run with the new options, the macro in configure.m4
	     triggers the download, patch, conifgure and tests.  No compilation is
	     performed at this time.  The downloaded tarball is cached in /tmp so
	     it doesn't get downloaded again on a distclean.

	     When make is run in the top-level Asterisk source directory, it will
	     automatically descend all the subdirectories in third_party just as it
	     does for addons, apps, etc.  The top-level Makefile makes sure that
	     the 'third-party' is built before 'main' so that dependencies from the
	     other directories are built first.

	     When main does build, a new shared library (libasteriskpj) is created that
	     links statically to the pjproject .a files and exports all their symbols.
	     The asterisk binary links to that, just as it does with libasteriskssl.

	     When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject
	     python bindings are installed in ASTDATADIR/third-party/pjproject.  This
	     will facilitate testing, including running the testsuite which will be
	     updated to check that directory for the pjsua module ahead of the system
	     python library.

	  Modules should continue to depend on pjproject if they use pjproject APIs
	  directly.  They should not care about the implementation.  No changes to any
	  res_pjsip modules were made.

	  Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103

2016-02-22 16:59 +0000 [2dae4a1ccf]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Fix T.38 issues caused by leaving a bridge.

	  chan_sip could not handle AST_T38_TERMINATED frames being sent to it when
	  the channel left the bridge.  The action resulted in overlapping outgoing
	  reINVITEs.  The testsuite tests/fax/sip/directmedia_reinvite_t38 was not
	  happy.

	  * Force T.38 to be remembered as locally bridged.  Now when the channel
	  leaves the native RTP bridge after T.38, the channel remembers that it has
	  already reINVITEed the media back to Asterisk.  It just needs to terminate
	  T.38 when the AST_T38_TERMINATED arrives.

	  * Prevent redundant AST_T38_TERMINATED from causing problems.  Redundant
	  AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if
	  they happen before the T.38 state changes to disabled.  Now the T.38 state
	  is set to disabled before the reINVITE is sent.

	  ASTERISK-25582 #close

	  Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce

2016-02-18 18:27 +0000 [bf29a4e2e6]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_t38.c: Back out part of an earlier fix attempt.

	  This backs out item 4 of the 4875e5ac32f5ccad51add6a4216947bfb385245d
	  commit.  Item 4 added the t38_bye_supplement.  Unfortunately, the frame
	  that it puts into the bridge may or may not be processed by the time the
	  bridged peer is kicked out of the bridge.  If it is processed then all is
	  well.  However, if it is not processed then that channel is stuck in fax
	  mode until it hangs up or maybe if it joins another bridge for T.38
	  faxing.

	  ASTERISK-25582

	  Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7

2016-02-22 13:54 +0000 [c7d45b84f9]  Richard Mudgett <rmudgett@digium.com>

	* bridge core: Add owed T.38 terminate when channel leaves a bridge.

	  The channel is now going to get T.38 terminated when it leaves the
	  bridging system and the bridged peers are going to get T.38 terminated as
	  well.

	  ASTERISK-25582

	  Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7

2016-02-19 16:01 +0000 [0e296563d7]  Richard Mudgett <rmudgett@digium.com>

	* channel api: Create is_t38_active accessor functions.

	  ASTERISK-25582

	  Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b

2016-02-19 19:06 +0000 [86f7336c91]  Richard Mudgett <rmudgett@digium.com>

	* bridge_channel: Don't settle owed events on an optimization.

	  Local channel optimization could cause DTMF digits to be duplicated.
	  Pending DTMF end events would be posted to a bridge when the local channel
	  optimizes out and is replaced by the channel further down the chain.  When
	  the real digit ends, the channel would get another DTMF end posted to the
	  bridge.

	  A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B

	  1) LocalA has the /n flag to prevent optimization.
	  2) B is sending DTMF to A through the local channel chain.
	  3) When LocalB optimizes out it can move B to the position of LocalB;1
	  4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would
	  settle an owed DTMF end to the bridge toward LocalA;2.
	  5) When B finally ends its DTMF it sends the DTMF end down the chain.
	  6) Without this patch, A would hear the DTMF digit end when LocalB
	  optimizes out and when B ends the original digit.

	  ASTERISK-25582

	  Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251

2016-02-22 12:15 +0000 [128c96456c]  Richard Mudgett <rmudgett@digium.com>

	* channel.c: Route all control frames to a channel through the same code.

	  Frame hooks can conceivably return a control frame in exchange for an
	  audio frame inside ast_write().  Those returned control frames were not
	  handled quite the same as if they were sent to ast_indicate().  Now it
	  doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a
	  channel or ast_indicate().

	  ASTERISK-25582

	  Change-Id: I5775f41421aca2b510128198e9b827bf9169629b

2016-02-25 15:13 +0000 [4422905218]  gtjoseph <george.joseph@fairview5.com>

	* sorcery:  Refactor create, update and delete to better deal with caches

	  The ast_sorcery_create, update and delete function have been refactored
	  to better deal with caches and errors.

	  The action is now called on all non-caching wizards first. If ANY succeed,
	  the action is called on all caching wizards and the observers are notified.
	  This way we don't put something in the cache (or update or delete) before
	  knowing the action was performed in at least 1 backend and we only call the
	  observers once even if there were multiple writable backends.

	  ast_sorcery_create was never adding to caches in the first place which
	  was preventing contacts from getting added to a memory_cache when they
	  were created.  In turn this was causing memory_cache to emit errors if
	  the contact was deleted before being retrieved (which would have
	  populated the cache).

	  ASTERISK-25811 #close
	  Reported-by: Ross Beer

	  Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46
2016-02-25 15:39 +0000 [acf329a3c7]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_mwi:  Turn some NOTICEs and WARNINGs into debug 1s.

	  There are a few cases where we're emitting notices or warnings
	  for things that really need neither, like a client retrying to subscribe
	  to mwi when they're not conifgured for it.  They get a 404 so there's no
	  need for non-debug messages.

	  Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f
2016-02-25 14:17 +0000 [7e3e1ddf7e]  gtjoseph <george.joseph@fairview5.com>

	* res_sorcery_memory_cache:  Fix SEGV in some CLI commands

	  A few of the CLI commands weren't checking for enough arguments
	  and were SEGVing.

	  Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413

2016-02-22 19:31 +0000 [803a2fc2d5]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.h: Remove extraneous semicolons.

	  Change-Id: Ib462633d396fa941379dfef648dcd2245e350084

2016-02-23 14:57 +0000 [886ee09471]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Suppress T.38 SDP c= line if addr is the same.

	  Use the correct comparison function since we only care if the address
	  without the port is the same.

	  Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0

2016-02-16 08:14 +0000 [b7970cabfa]  Christof Lauber <christof.lauber@annax.ch>

	* res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables

	  Introduced realloaction of ast_str buf in sqlite3_escape functions in case
	  the returned buffer from threadstorage was actually too small.

	  Change-Id: I3c5eb43aaade93ee457943daddc651781954c445

2016-02-11 11:01 +0000 [ba8adb4ce3]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip/config_transport: Allow reloading transports.

	  The 'reload' mechanism actually involves closing the underlying
	  socket and calling the appropriate udp, tcp or tls start functions
	  again.  Only outbound_registration, pubsub and session needed work
	  to reset the transport before sending requests to insure that the
	  pjsip transport didn't get pulled out from under them.

	  In my testing, no calls were dropped when a transport was changed
	  for any of the 3 transport types even if ip addresses or ports were
	  changed. To be on the safe side however, a new transport option was
	  added (allow_reload) which defaults to 'no'.  Unless it's explicitly
	  set to 'yes' for a transport, changes to that transport will be ignored
	  on a reload of res_pjsip.  This should preserve the current behavior.

	  Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf

2016-02-19 04:30 +0000 [c00082329e]  Walter Doekes <walter+asterisk@wjd.nu>

	* chan_sip: Optionally supply fromuser/fromdomain in SIP dial string.

	  Previously you could add [!dnid] to the SIP dial string to alter the To:
	  header. This change allows you to alter the From header as well.

	  SIP dial string extra options now look like this:

	      [![touser[@todomain]][![fromuser][@fromdomain]]]

	  INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To:
	  header, that is no longer possible.

	  ASTERISK-25803 #close

	  Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7

2016-02-07 17:34 +0000 [f8767a8804]  gtjoseph <george.joseph@fairview5.com>

	* res_pjproject:  Add ability to map pjproject log levels to Asterisk log levels

	  Warnings and errors in the pjproject libraries are generally handled by
	  Asterisk.  In many cases, Asterisk wouldn't even consider them to be warnings
	  or errors so the messages emitted by pjproject directly are either superfluous
	  or misleading.  A good exampe of this are the level-0 errors pjproject emits
	  when it can't open a TCP/TLS socket to a client to send an OPTIONS.  We don't
	  consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS
	  client be treated any differently?

	  A config file for res_pjproject has bene added (pjproject.conf) and a new
	  log_mappings object allows mapping pjproject levels to Asterisk levels
	  (or nothing).  The defaults if no pjproject.conf file is found are the same
	  as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR,
	  2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level>

	  Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898

2016-02-18 10:55 +0000 [14886643c6]  Alexei Gradinari <alex2grad@gmail.com>

	* res_pjsip_outbound_publish: Fix processing 412 response

	  When Asterisk receives a 412 (Conditional Request Failed) response
	  it has to recreate publish session.
	  There is bug in res_pjsip_outbound_publish.c
	  The function sip_outbound_publish_client_alloc is called with wrong object
	  while processing 412 (Conditional Request Failed) response.
	  This patch fixes it.

	  ASTERISK-25229 #close

	  Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359

2016-02-18 11:15 +0000 [8055d080cd]  Mark Michelson <mmichelson@digium.com>

	* Fix failing threadpool_auto_increment test.

	  The threadpool_auto_increment test fails infrequently for a couple of
	  reasons
	  * The threadpool listener was notified of fewer tasks being pushed than
	    were actually pushed
	  * The "was_empty" flag was set to an unexpected value.

	  The problem is that the test pushes three tasks into the threadpool.
	  Test expects the threadpool to essentially gather those three tasks, and
	  then distribute those to the threadpool threads. It also expects that as
	  the tasks are pushed in, the threadpool listener is alerted immediately
	  that the tasks have been pushed. In reality, a task can be distributed
	  to the threadpool threads quicker than expected, meaning that the
	  threadpool has already emptied by the time each subsequent task is
	  pushed. In addition, the internal threadpool queue can be delayed so
	  that the threadpool listener is not alerted that a task has been pushed
	  even after the task has been executed.

	  From the test's point of view, there's no way to be able to predict
	  exactly the order that task execution/listener notifications will occur,
	  and there is no way to know which listener notifications will indicate
	  that the threadpool was previously empty.

	  For this reason, the test has been updated to only check the things it
	  can check. It ensures that all tasks get executed, that the threads go
	  idle after the tasks are executed, and that the listener is told the
	  proper number of tasks that were pushed.

	  Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c

2016-02-17 13:30 +0000 [30a49b8a6a]  Richard Mudgett <rmudgett@digium.com>

	* cel.c: Fix mismatch in ast_cel_track_event() return type.

	  The return type of ast_cel_track_event() is not large enough to return all
	  64 potential bits of the event enable mask.  Fortunately, the defined CEL
	  events do not really need all 64 bits and the return value is only used to
	  determine if the requested CEL event is enabled.

	  * Made the ast_cel_track_event() return 0 or 1 only so the return value
	  can fit inside an int type instead of zero or a truncated 64 bit non-zero
	  value.

	  Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c

2016-02-16 23:37 +0000 [15aeb78c66]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue: fix Calculate talktime when is first call answered

	  Fix calculate of average time for talktime is wrong when is completed the
	  first call beacuse the time for talked would be that call.

	  ASTERISK-25800 #close

	  Change-Id: I94f79028935913cd9174b090b52bb300b91b9492

2016-02-16 16:37 +0000 [62282bb8ce]  gtjoseph <george.joseph@fairview5.com>

	* res_odbc: Fix exports.in for missing symbols

	  res_odbc.exports.in was missing a few symbols.
	  Changed to wildcards.

	  Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c

2016-02-16 12:20 +0000 [49203628f9]  gtjoseph <george.joseph@fairview5.com>

	* res_statsd:  Fix exports.in for missing symbols

	  res_statsd.export.in was missing the _va variations of the log
	  functions causing Asterisk to crash in res_pjsip if OPTIONAL_API
	  wasn't enabled.

	  ASTERISK-25727 #close
	  Reported-by: Gergely Dömsödi

	  Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b

2016-02-15 21:31 +0000 [4f08e9fb64]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard:  Add command to export primitive objects

	  A new command (pjsip export config_wizard primitives) has been added that
	  will export all the pjsip objects it created to the console or a file
	  suitable for reuse in a pjsip.conf file.

	  ASTERISK-24919 #close
	  Reported-by: Ray Crumrine

	  Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b

2016-02-15 15:37 +0000 [be811c4be1]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_caller_id: Fix segfault when replacing rpid or pai header

	  If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid
	  or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify
	  the header added by the dialplan function.  Since the header added by the
	  dialplan function is generic string, there are no virtual functions to parse
	  the uri and we get a segfault when we try.  Since the modify, was really only
	  an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER
	  and recreate it.

	  This raises a question for another time though:  What should happen with
	  duplicate headers?  Right now res_pjsip_header_funcs doesn't check for dups
	  so if it's session supplement is loaded after res_pjsip_caller_id's (or any
	  other module that adds headers), there'll be dups in the message.

	  ASTERISK-25337 #close

	  Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa

2016-02-15 13:08 +0000 [13b6c02945]  Mark Michelson <mmichelson@digium.com>

	* Fix creation race of contact_status structures.

	  It is possible when processing a SIP REGISTER request to have two
	  threads end up creating contact_status structures in sorcery.
	  contact_status is created using a "find or create" function. If two
	  threads call into this at the same time, each thread will fail to find
	  an existing contact_status, and so both will end up creating a new
	  contact status.

	  During testing, we would see sporadic failures because the
	  PJSIP_CONTACT() dialplan function would operate on a different
	  contact_status than what had been updated by res_pjsip/pjsip_options.

	  The fix here is two-fold:
	  1) The "find or create" function for contact_status now has a lock
	  around the entire operation. This way, if two threads attempt the
	  operation simultaneously, the first to get there will create the object,
	  and the second will find the object created by the first thread.

	  2) res_sorcery_memory has had its create callback updated so that it
	  will not allow for objects with duplicate IDs to be created.

	  Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97

2016-02-15 12:52 +0000 [5c400a0fed]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_pubsub: Move where the subscription is stored to after initialized.

	  A problem arose when testing the AMI subscription listing actions where it
	  was possible for a subscription that had not been fully initialized to be
	  listed. This was problematic as the underlying listing code would crash.

	  This change makes it so the subscription tree is fully set up before it is
	  added to the list of subscriptions. This ensures that when the listing actions
	  get the subscription it is valid.

	  ASTERISK-25738 #close

	  Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48

2016-02-09 17:34 +0000 [b37555cc94]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Refactor load_module/unload_module

	  load_module was just too hairy with every step having to clean up all
	  previous steps on failure.

	  Some of the pjproject init calls have now been moved to a separate
	  load_pjsip function and the unload_pjsip function was enhanced to clean
	  up everything if an error happened at any stage of the load process.

	  In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns
	  and ast_threadpool_shutdowns were also corrected.

	  Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302

2016-02-09 22:42 +0000 [c4d9f46878]  Badalyan Vyacheslav <slavon.net@gmail.com>

	* Resources/res_phoneprov: fix memory leak and heap-use-after-free

	  * heap-use-after-free happens when we free "cfg"
	  but then use "value" which refers to it

	  * A memory leak occurs because in some cases
	  it is not released "defaults"

	  ASTERISK-25721 #close
	  Reported by: Badalyan Vyacheslav
	  Tested by: Badalyan Vyacheslav

	  Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469

2016-02-11 11:21 +0000 [e5fd972d24]  Etienne Lessard (license #6394)

	* func_iconv: Ensure output strings are properly terminated.

	  ASTERISK-25272 #close
	  Reported by: Etienne Lessard
	  patches:
	   AST-25272.patch submitted by Etienne Lessard (license #6394)

	  Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17

2016-02-10 16:16 +0000 [168c18737f]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Handle pjsip_dlg_create_uas deprecation

	  Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with
	  pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically
	  increments the lock on the returned dialog.  To account for this, configure.ac
	  now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c
	  has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use
	  the original call or the new one.  If the new one was used, the ref count is
	  decremented before returning.

	  ASTERISK-25751 #close
	  Reported-by Josh Colp

	  Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8

2016-02-09 20:13 +0000 [fd668670b5]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* res_config_pgsql: Show error message in reload if not connected.

	  Change-Id: I9290115a1aaadb589eb1d02eaeb502eec01b31fa

2016-02-09 23:40 +0000 [a23d01e943]  Badalyan Vyacheslav <slavon.net@gmail.com>

	* Build: Added testing compiler to support the system sanitizes

	  In older versions of the compiler was not sanitizes.
	  Compilers other than GCC can not support the Usan and TSAN
	  or have other options for *FLAGS.

	  ASTERISK-25767 #close
	  Reported by: Badalyan Vyacheslav
	  Tested by: Badalyan Vyacheslav

	  Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916

2016-02-09 20:57 +0000 [c7186c7f0a]  Badalyan Vyacheslav <v.badalyan@open-bs.ru>

	* Build: Fix menuselect USAN conflicts

	  USAN can be used together with other sanitizers.

	  Reported by: Badalyan Vyacheslav
	  Tested by: Badalyan Vyacheslav

	  Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f

2016-02-09 14:21 +0000 [68643f83cd]  Corey Farrell <git@cfware.com>

	* Simplify and fix conditional in FD_SET.

	  FD_SET contains a conditional statement to protect against buffer
	  overruns.  The statement was overly complicated and prevented use
	  of the last array element of ast_fdset.  We now just verify the fd
	  is less than ast_FDMAX.

	  Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40

2016-02-09 07:11 +0000 [e40fddbeb5]  Joshua Colp <jcolp@digium.com>

	* tests/test_sorcery_memory_cache_thrash: Improve termination process.

	  When terminating the threads thrashing a sorcery memory cache each
	  would be told to stop and then we would wait on them. During at
	  least one thrashing test this was problematic due to the specific
	  usage pattern in use. It would take some time for termination of the
	  thread to occur.

	  This would occur due to contention between the threads retrieving
	  and the threads updating the cache. As the retrieving threads are
	  given priority it may be some time before the updating threads
	  are able to proceed.

	  This change makes it so all threads are told to stop and then each
	  are joined to ensure they stop. This way all the threads should
	  stop at around the same time instead of waiting for one to stop,
	  the next to stop, then the next, and so on. As a result of this
	  the execution time for each thrash test is much closer to their
	  expected value than previously seen as well.

	  Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827
2016-01-29 17:56 +0000 [bbf3ace682]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Fix infinite recursion when loading transports from realtime

	  Attempting to load a transport from realtime was forcing asterisk into an
	  infinite recursion loop.  The first thing transport_apply did was to do a
	  sorcery retrieve by id for an existing transport of the same name. For files,
	  this just returns the previous object from res_sorcery_config's internal
	  container, if any.  For realtime, the res_sourcery_realtime driver looks in the
	  database and finds the existing row but now it has to rehydrate it into a
	  sorcery object which means calling... transport_apply.  And so it goes.

	  The main issue with loading from realtime (apart from the loop) was that
	  transport stores structures and pointers directly in the ast_sip_transport
	  structure instead of the separate ast_transport_state structure.  This patch
	  separates those items into the ast_sip_transport_state structure.  The pattern
	  is roughly the same as res_pjsip_outbound_registration.

	  Although all current usages of ast_sip_transport and ast_sip_transport_state
	  were modified to use the new ast_sip_get_transport_state API, the original
	  items are left in ast_sip_transport and kept updated to maintain ABI
	  compatability for third-party modules.  They are marked as deprecated and
	  noted that they're now in ast_sip_transport_state.

	  ASTERISK-25606 #close
	  Reported-by: Martin Moučka

	  Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19

2016-02-07 13:00 +0000 [72bf53eea5]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* res_config_pgsql: Add message on cli failed command status

	  In case failed of command "realtime show pgsql status" show a message the data
	  of connection to more clear information in error.

	  Change-Id: Ia8e9e2400466606e7118f52a46e05df0719b6a29

2016-02-05 10:29 +0000 [b69729dde5]  gtjoseph <george.joseph@fairview5.com>

	* chan_misdn: Fix a few issues causing compile errors

	  Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98

2016-01-25 17:36 +0000 [1bc54aee80]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge: Only use b_profile options from the conference.

	  A user cannot set new bridge options after the conference is created by
	  the first user.  Attempting to do so is documented as undefined behavior.

	  This patch ensures that the bridge profile options used are from the
	  conference and not what a subsequent user may have tried to set.

	  Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266

2016-02-04 16:17 +0000 [3b426a8b09]  Mark Michelson <mmichelson@digium.com>

	* Check for OpenSSL defines before trying to use them.

	  The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
	  to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
	  these options, which can cause problems on systems with older OpenSSL
	  installations.

	  This commit adds a configure script check for those defines and will not
	  attempt to make use of those if they do not exist. We will print a
	  warning urging the user to upgrade their OpenSSL installation if those
	  defines are not present.

	  Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d
2016-02-03 14:25 +0000 [9b13ab6a63]  gtjoseph <george.joseph@fairview5.com>

	* pjsip/alembic:  Add missing columns to system and registration

	  ps_systems needed disable_tcp_switch
	  ps_registrations needed line and endpoint

	  ASTERISK-25737 #close

	  Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19

2016-02-04 11:39 +0000 [82e2938fa8]  Mark Michelson <mmichelson@digium.com>

	* res_stasis_device_state: Fix refcounting error.

	  Device state subscription lifetimes were governed by when the
	  subscription was established and unsubscribed from. However, it is
	  possible that at the time of unsubscription, there could be device state
	  events still in flight. When those device state events occur, the device
	  state callback could attempt to dereference a freed pointer. Crash.

	  This change ensures that the lifetime of the device state subscription
	  does not end until the underlying stasis subscription has confirmed that
	  its final message has been sent.

	  Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2

2016-01-27 10:44 +0000 [d83dba7099]  Sean Bright <sean.bright@gmail.com>

	* res_rtp_asterisk: Allow ICE host candidates to be overriden

	  During ICE negotiation the IPs of the local interfaces are sent to the remote
	  peer as host candidates. In many cases Asterisk is behind a static one-to-one
	  NAT, so these host addresses will be internal IP addresses.

	  To help in hiding the topology of the internal network, this patch adds the
	  ability to override the host candidates by matching them against a
	  user-defined list of replacements.

	  Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f

2016-02-03 12:05 +0000 [0de74fad55]  Joshua Colp <jcolp@digium.com>

	* AST-2016-001 http: Provide greater control of TLS and set modern defaults.

	  This change exposes the configuration of various aspects of the TLS
	  support and sets the default to the modern standards.

	  The TLS cipher is now set to the best values according to the
	  Mozilla OpSec team, different TLS versions can now be disabled, and
	  the cipher order can be forced to be that of the server instead of
	  the client.

	  ASTERISK-24972 #close

	  Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8
2015-12-07 12:46 +0000 [e67b445e8d]  Richard Mudgett <rmudgett@digium.com>

	* AST-2016-003 udptl.c: Fix uninitialized values.

	  Sending UDPTL packets to Asterisk with the right amount of missing
	  sequence numbers and enough redundant 0-length IFP packets, can make
	  Asterisk crash.

	  ASTERISK-25603 #close
	  Reported by: Walter Doekes

	  ASTERISK-25742 #close
	  Reported by: Torrey Searle

	  Change-Id: I97df8375041be986f3f266ac1946a538023a5255
2015-09-28 17:07 +0000 [a877e0d94b]  Richard Mudgett <rmudgett@digium.com>

	* AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow.

	  Setting the sip.conf timert1 value to a value higher than 1245 can cause
	  an integer overflow and result in large retransmit timeout times.  These
	  large timeout times hold system file descriptors hostage and can cause the
	  system to run out of file descriptors.

	  NOTE: The default sip.conf timert1 value is 500 which does not expose the
	  vulnerability.

	  * The overflow is now detected and the previous timeout time is
	  calculated.

	  ASTERISK-25397 #close
	  Reported by: Alexander Traud

	  Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290
2016-02-03 14:07 +0000 [dcbedf9ab1]  gtjoseph <george.joseph@fairview5.com>

	* logging: Remove/fix some message annoyances

	  test_dlinklists doesn't need to NOTICE everyone that every macro worked.

	  res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or
	  provider was registered.

	  res_odbc was missing a newline at the end of one message.

	  Change-Id: I6c06361518ef3711821795e535acd439782a995e

2016-02-02 10:52 +0000 [6522361871]  Alexei Gradinari License #5691

	* res_sorcery_realtime: Fix regex regression.

	  A regression was introduced where searching for realtime PJSIP objects
	  by regex by starting the regex with a leading "^" would cause no items
	  to be returned.

	  This was due to a change which attempted to drop the requirement for a
	  leading "^" to be present due to how some CLI commands formulate their
	  regexes. However, the change, rather than simply eliminating the
	  requirement, caused any regexes that did begin with "^" to end up not
	  returning the expected results.

	  This change fixes the problem by inspecting the regex and formulating
	  the realtime query differently depending on if it begins with "^".

	  ASTERISK-25702 #close
	  Reported by Nic Colledge

	  Patches:
	      realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691

	  Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693

2016-02-02 04:05 +0000 [2a6f18cd55]  Karsten Wemheuer <kwe-digium@iptam.com>

	* res_xmpp: Does not connect in component mode

	  The module res_xmpp does not accept usernames in the form used in component
	  mode (XEP-0114). In component mode there is no @something in the name.
	  In component mode the connection is now not dropped anymore.

	  If the xmpp server sends out a "stream" tag before handshake is finished,
	  the connection gets dropped in res_xmpp. Now this tag will be ignored and
	  the connection will be established.

	  After connecting there will be an exchange of presence states. This does
	  not work as expected in component mode. The responsible function
	  "xmpp_pak_presence" is left before the states get sent out. Sending
	  presence states in component mode is now moved to the top of the function.

	  ASTERISK-25735 #close

	  Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c
2016-02-01 13:04 +0000 [40da6434c1]  gtjoseph <george.joseph@fairview5.com>

	* build_system:  Fix some warnings highlighted by clang

	  Fix some warnings found with clang.

	  Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd

2016-01-31 20:13 +0000 [52b29f9b4c]  gtjoseph <george.joseph@fairview5.com>

	* pjsip/alembic: Fix definition of qualify_timeout

	  A recent commit set qualify_timeout to Decimal which isn't supported.
	  This path corrects it to Float.

	  Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf

2016-01-29 07:39 +0000 [55a7367ad4]  Stefan Engström <stefanen@kth.se>

	* chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip.

	  When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a)
	  AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect
	  asterisk to include the same value for its own ip in both cases a) and b),
	  but it seems a) produces a contact header like Contact:
	  <sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like
	  <sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf

	  My guess is that manager_sipnotify should call
	  ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does,
	  because after applying this patch, both cases a) and b) produce
	  the contact header that I expect: <sip:asterisk@192.168.1.227:8060>

	  Reported by: Stefan Engström
	  Tested by: Stefan Engström

	  Change-Id: I86af5e209db64aab82c25417de6c768fb645f476
2016-01-28 12:44 +0000 [d2397f028f]  Richard Mudgett <rmudgett@digium.com>

	* config_options.c: Fix warning message wording.

	  Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4

2016-01-25 17:34 +0000 [af6b15976d]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge.c: Replace inlined code with existing function.

	  Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51

2016-01-25 16:05 +0000 [7932336a3d]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge: Add ability to get the muted conference state.

	  * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.

	  * Added Muted header to AMI ConfbridgeListRooms action response list
	  events to indicate the muted conference state.

	  * Added Muted column to CLI "confbridge list" output to indicate the muted
	  conference state and made the locked column a yes/no value instead of a
	  locked/unlocked value.

	  ASTERISK-20987
	  Reported by: hristo

	  Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1

2016-01-26 17:59 +0000 [894045e7cf]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.

	  Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7

2016-01-25 15:48 +0000 [12c93e8f81]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge: Make non-admin users join a muted conference muted.

	  ASTERISK-20987 #close
	  Reported by: hristo

	  Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38

2016-01-27 13:08 +0000 [f19bf7a321]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Add res_pjproject dependency to samples

	  Since res_pjsip now depends on res_pjproject, this has been added to
	  basic-pbx modules.conf.

	  Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d
2016-01-27 10:29 +0000 [c53903d447]  gtjoseph <george.joseph@fairview5.com>

	* build_system: Prevent goals needing makeopts from running when it's missing

	  The Makefile only optionally includes makeopts so when goals like uninstall that
	  dont depend on anything else are run after a distclean, rules like
	  'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts
	  to remove everything in the root directory.

	  Although there's a rule defined for makeopts which prints a message and does
	  an 'exit 1', since '-include makepopts' was specified (with the -), the exit
	  was ignored letting the rest of the rules run.

	  This patch makes makeopts required unless the goal has the string 'clean' in it.

	  ASTERISK-25730 #close
	  Reported-by: George Joseph

	  Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7

2016-01-25 09:35 +0000 [1dfd104a27]  Joshua Colp <jcolp@digium.com>

	* config: Allow options to register when documentation is unavailable.

	  The config options framework is strict in that configuration options must
	  be documented unless XML documentation support is not available. In
	  practice this is useful as it ensures documentation exists however in
	  off-nominal cases this can cause strange problems.

	  If it is expected that a config option has a non-zero or non-empty
	  default value but the config option documentation is unavailable
	  this reasonable expectation will not be met. This can cause obscure
	  crashes and weirdness depending on how the code handles it.

	  This change tweaks the behavior to ensure that the config option
	  is still allowed to register, apply default values, and be set when
	  devmode is not enabled. If devmode is enabled then the option can
	  NOT be set.

	  This also does not remove the initial documentation error message that
	  is output on load when registering the configuration option.

	  ASTERISK-25725 #close

	  Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8

2016-01-25 10:23 +0000 [a706ad44e6]  Mark Michelson <mmichelson@digium.com>

	* Stasis: Use custom structure when setting variables.

	  A recent change to queue channel variable setting to the Stasis control
	  queue caused a regression. When setting channel variables, it is
	  possible to give a NULL channel variable value in order to unset the
	  variable (i.e. remove it from the channel variable list). The change
	  introduced a call to ast_variable_new(), which is not tolerant of NULL
	  channel variable values.

	  This new change switches from using ast_variable to using a custom
	  channel variable struct that is lighter weight and NULL value-tolerant.

	  Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d

2016-01-25 16:56 +0000 [289daca9e8]  Rusty Newton <rnewton@digium.com>

	* sounds/Makefile: Incremented core and extra sounds versions to 1.5

	  Core and extra sounds 1.5 was recently released! The tarballs contain
	  change descriptions however I figure more people will see this one so
	  I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra
	  to Core for en, en_GB, fr and added for languages that didn't already
	  have Extra sound sets (it,ja,ru).

	  In addition all of the English and Russian sounds have been completely
	  re-recorded.

	  Sounds moved and added:
	  activated,added,all-circuits-busy-now,astcc-followed-by-pound
	  at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy
	  ,call-fwd-unconditional,calling,call-waiting,cancelled,
	  cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated
	  ,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist
	  ,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello
	  ,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to
	  ,location,number,number-not-answering,num-was-successfully,one-moment-please
	  ,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option
	  ,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached
	  ,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial
	  ,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension
	  ,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered
	  ,your

	  There were also a few random fixes here and there to file names for a few
	  of the languages.

	  ASTERISK-25068 #close

	  Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3
2016-01-25 16:51 +0000 [b073244c51]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Prevent crash from AMI command on freed subscription.

	  A test recently uncovered that running an ill-timed AMI command to show
	  inbound subscriptions could cause a crash since Asterisk will try to
	  operate on a freed subscription.

	  The fix for this is to remove the subscription tree from the list of
	  subscriptions at the time that we are sending our final NOTIFY request
	  out. This way, as the subscription is in the process of dying, it is
	  inaccessible from AMI.

	  Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23

2016-01-25 11:03 +0000 [830f8933c2]  Corey Farrell <git@cfware.com>

	* chan_sip: Fix buffer overrun in sip_sipredirect.

	  sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer
	  of 256 characters.  This patch reduces the copy to 255 characters to leave
	  room for the string null terminator.

	  ASTERISK-25722 #close

	  Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab

2016-01-23 16:45 +0000 [f299dc0d76]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue: Add  Lastpause field of queue member

	  Add time when started a the last pause for a queue member for
	  QueueMemberStatus ami event.

	  Also show accumulate time in seconds when started a pause for a queue
	  member to CLI command 'queue show'.

	  ASTERISK-16394 #close

	  Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c

2016-01-23 12:34 +0000 [8c664da0ff]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue: fix some tab format

	  Change-Id: I2734392b131f1fb0949515d538f83f30fbc15d8c

2016-01-23 11:41 +0000 [2fb45c7801]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* cdr_pgsql.cl: REFACTOR Macro LENGTHEN_BUF

	  Remove repeated code on macro of assigned buffer to SQL vars.

	  Add table and connection name to log error message when is not possible
	  allocate memory.

	  Change-Id: I1fbf37d286a032d38fdda72a9f736356956c9ffe

2016-01-22 15:08 +0000 [959f7436cc]  Mark Michelson <mmichelson@digium.com>

	* Stasis: Fix potential memory leak of control data.

	  When queuing tasks onto the Stasis control queue, you can pass an
	  arbitrary data pointer and a function to free that data. All ARI
	  commands that use the Stasis control queue made the assumption that the
	  destructor function would be called in all paths, whether the task was
	  queued successfully or not. However, this was not correct. If a task was
	  queued onto a control structure that was already completed, the
	  allocated data would not be freed properly.

	  This patch corrects this by making sure that all return paths call the
	  data destructor.

	  Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb

2016-01-21 10:58 +0000 [a45eacebf3]  Mark Michelson <mmichelson@digium.com>

	* Stasis: Use control queue to prevent crash.

	  A crash occurred when attempting to set a channel variable on a channel
	  that had already been hung up. This is because there is a small window
	  between when a control is grabbed and when the channel variable is set
	  that the channel can be hung up.

	  The fix here is to queue the setting of the channel variable onto the
	  control queue. This way, the manipulation of the channel happens in a
	  thread where it is safe to be done.

	  In this change, I also noticed that the setting of bridge roles on
	  channels was being done outside of the control queue, so I also changed
	  those operations to be done in the control queue.

	  ASTERISK-25709 #close
	  Reported by Mark Michelson

	  Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78

2016-01-22 11:48 +0000 [7866806fc3]  Richard Mudgett <rmudgett@digium.com>

	* logger.c: Fix buffer overrun found by address sanitizer.

	  The null terminator of the tail struct member was not being allocated
	  when no logger.conf config file is installed.

	  ASTERISK-25714 #close
	  Reported by: Badalian Vyacheslav

	  Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30

2015-12-23 15:07 +0000 [9714da7aa4]  Mark Michelson <mmichelson@digium.com>

	* res_odbc: Remove connection management

	  Asterisk by default will create a single database connection and share
	  it among all threads that attempt to access the database. In previous
	  versions of Asterisk, this was tolerable, because the most used channel
	  driver, chan_sip, mostly accessed the database from a single thread.
	  With PJSIP, however, many threads may be attempting to perform database
	  operations, and there is the potential for many more database accesses,
	  meaning the concurrency is a horrible bottleneck if only one connection
	  is shared.

	  Asterisk has a connection pooling facility built into it, but the
	  implementation has flaws. For one, there is a strict limit on the number
	  of simultaneous connections that could be made to the database. Anything
	  beyond the maximum would result in a failed operation. Attempting to
	  predict what the maximum should be is nearly impossible even for someone
	  intimately familiar with Asterisk's threading model. In addition, use of
	  transactions in the dialplan can cause some severe bugs if connection
	  pooling is enabled.

	  This commit seeks to fix the concurrency problem by removing all
	  connection management code from Asterisk and leaving that to the
	  underlying unixODBC code instead. Now, Asterisk does not share a single
	  connection, nor does it try to maintain a connection pool. Instead, all
	  Asterisk ever does is request a connection from unixODBC and allow
	  unixODBC to either allocate those connections or retrieve them from a
	  pool.

	  Doing this has a bit of a ripple effect. For one, since connections are
	  not long-lived objects, several of the safeguards that previously
	  existed have been removed. We don't have to worry about trying to use a
	  connection that has gone stale. In every case, when we request a
	  connection, it has just been made and we don't need to perform any
	  sanity checks to be sure it's still active.

	  Another major player affected by this change is transactions.
	  Transactions and their respective connections were so tightly coupled
	  that it was almost pornographic. This code change moves
	  transaction-related code to its own file separate from the core ODBC
	  functionality. This way, the core of ODBC does not even have to know
	  that transactions exist.

	  In making this large change, I had to look at a lot of code and
	  understand it. When making this change, I discovered several places
	  where the behavior is definitely not ideal, but it seemed outside the
	  scope of this change to be fixing it. Instead, any place where I saw
	  some sort of room for improvement has had a XXX comment added explaining
	  what could be altered to improve it.

	  Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf

2016-01-22 11:18 +0000 [d3969d09ae]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue.c: remove include for core_unreal.h not used in code.

	  Change-Id: Idc2ae8a6bd869a66544916906744a5678622262d

2016-01-21 16:40 +0000 [5dde111719]  Corey Farrell <git@cfware.com>

	* Build System: Add support for checking alembic branches.

	  * Add 'check-alembic' target to root Makefile.
	  * Create build_tools/make_check_alembic to do the actual checks.

	  ASTERISK-25685

	  Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004

2016-01-19 18:20 +0000 [04078f43b5]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case.

	  ASTERISK-25712 #close
	  Reported by: Richard Mudgett

	  Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f

2016-01-13 16:49 +0000 [5615db3714]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Add CLI "pjsip dump endpt [details]"

	  Dump the res_pjsip endpt internals.

	  In non-developer mode we will not document or make easily accessible the
	  "details" option even though it is still available.  The user has to know
	  it exists to use it.  Presumably they would also be aware of the potential
	  crash warning below.

	  Warning: PJPROJECT documents that the function used by this CLI command
	  may cause a crash when asking for details because it tries to access all
	  active memory pools.

	  Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb

2016-01-18 03:49 +0000 [b259ac95ac]  Diederik de Groot <ddegroot@talon.nl>

	* main/asterisk.c: ast_el_read_char

	  Make sure buf[res] is not accessed at res=-1 (buffer underrun).
	  Address Sanitizer will complain about this quite loudly.

	  ASTERISK-24801 #close

	  Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9

2016-01-18 19:27 +0000 [dd5c063934]  gtjoseph <george.joseph@fairview5.com>

	* res_pjproject:  Add module providing pjproject logging and utils

	  res_pjsip_log_forwarder has been renamed to res_pjproject
	  and enhanced as follows:

	  As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch,
	  a new ast_pjproject_get_buildopt function has been added.  It
	  allows the caller to get the value of one of the buildopts.

	  The initial use case is retrieving the runtime value of
	  PJ_MAX_HOSTNAME to insure we don't send a hostname greater
	  than pjproject can handle.  Since it can differ between
	  the version of pjproject that Asterisk was compiled against
	  and the version of pjproject that Asterisk is running against,
	  we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk
	  source code.

	  Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e

2016-01-18 17:16 +0000 [3b9cba4294]  Matt Jordan <mjordan@digium.com>

	* funcs/func_cdr: Correctly report high precision values for duration and billsec

	  When CDRs were refactored, func_cdr's ability to report high precision values
	  for duration and billsec (the 'f' option) was broken. This was due to func_cdr
	  incorrectly interpreting the duration/billsec values provided by the CDR engine
	  in milliseconds, as opposed to seconds. Since the CDR engine only provides
	  duration and billsec in seconds, and does not expose either attribute with
	  sufficient precision to merely pass back the underlying value, this patch fixes
	  the bug by re-calculating duration and billsec with microsecond precision based
	  on the start/answer/end times on the CDR.

	  ASTERISK-25179 #close

	  Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841

2016-01-20 07:52 +0000 [479cc99acd]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* README: Update year in copyright

	  Change-Id: I56240f537fb3205672cdb2a74f0591ae7bb73dbc

2016-01-19 17:15 +0000 [9fa76ba215]  Joshua Colp <jcolp@digium.com>

	* test_threadpool: Wait for each task to complete and fix memory leak.

	  This change makes the thread_timeout_thrash unit test wait for
	  each task to complete. This fixes the problem where the test would
	  prematurely end when all threads were gone and a new one had to be
	  started to handle the last task. It also increases the thrasing as
	  it is now more likely for each task to encounter the above scenario.

	  This also fixes a memory leak where the data for each task was not
	  being freed.

	  ASTERISK-25611 #close

	  Change-Id: I5017d621a4dc911f509074c16229b86bff2fb3c6

2016-01-18 19:44 +0000 [c9f7269b2e]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Increase CLI "core ping taskprocessor" timeout.

	  Change-Id: I4892d6acbb580d6c207d006341eaf5e0f8f2a029

2016-01-18 19:43 +0000 [6e2a867716]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Fix some taskprocessor unrefs.

	  You have to call ast_taskprocessor_unref() outside of the taskprocessor
	  implementation code.  Taskprocessor use since v12 has become more
	  transient than just the singleton uses in earlier versions.

	  Change-Id: If7675299924c0cc65f2a43a85254e6f06f2d61bb

2016-01-19 14:16 +0000 [a4dcbdf50f]  Richard Mudgett <rmudgett@digium.com>

	* Fix alembic branches on master.

	  Change-Id: I64ed21fec50eb833641ca49d92184f6aaabd86e8

2016-01-05 17:12 +0000 [35a3e8cc7f]  Corey Farrell <git@cfware.com>

	* Refactor init_logger_chain locking.

	  This removes logchannels locking from init_logger_chain, puts the
	  responsibility on the caller.  Adds locking around the one call that was
	  missing it.

	  ASTERISK-24833

	  Change-Id: I6cc42117338bf9575650a67bcb78ab1a33d7bad8

2016-01-18 22:10 +0000 [378fed4900]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue: Fix preserved reason of pause when Asterisk is restared

	  When the Asterisk is restared is not preseved reason paused of members.
	  This patch fixed this cases, retain data on astdb and set when Asterisk
	  is started.

	  ASTERISK-25732 #close

	  Report by: Rodrigo Ramírez Norambuena

	  Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5

2016-01-18 19:01 +0000 [130aa1427e]  gtjoseph <george.joseph@fairview5.com>

	* pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject

	  Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b

2016-01-16 13:18 +0000 [eaf2b5052e]  Daniel Journo <dan@keshercommunications.com>

	* Update version number in features.conf.sample

	  Update the version number in the comments from Asterisk 12 to Asterisk 12+

	  Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b

2016-01-13 15:58 +0000 [c60d6c0162]  Daniel Journo <dan@keshercommunications.com>

	* pjsip/alembic:  Fix qualify_timeout column definition

	  Corrects the qualify_timeout column type from Integer to Decimal

	  ASTERISK-25686 #close
	  Reported-by: Marcelo Terres

	  Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8

2016-01-15 19:52 +0000 [480ccfcc97]  Corey Farrell <git@cfware.com>

	* main/config: Clean config maps on shutdown.

	  ASTERISK-25700 #close

	  Change-Id: I096da84f9c62c6095f68bcf98eac4b7c7868e808

2016-01-14 14:42 +0000 [a5b38b604c]  Kevin Harwell <kharwell@digium.com>

	* bridge_basic: don't cache xferfailsound during an attended transfer

	  The xferfailsound was read from the channel at the beginning of the transfer,
	  and that value is "cached" for the duration of the transfer. Therefore, changing
	  the xferfailsound on the channel using the FEATURE() dialplan function does
	  nothing once the transfer is under way.

	  This makes it so the transfer code instead gets the xferfailsound configuration
	  options from the channel when it is actually going to be used.

	  This patch also fixes a potential memory leak of the props object as well as
	  making sure the condition variable gets initialized before being destroyed.

	  ASTERISK-25696 #close

	  Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4

2015-07-10 10:37 +0000 [d36c4d0b01]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Simplify ast_taskprocessor_get() return code.

	  Change-Id: Id5bd18ef1f60ef8be453e677e98478298358a9d1

2016-01-13 18:20 +0000 [0a878020dc]  Richard Mudgett <rmudgett@digium.com>

	* astmm.c: Add more stats to CLI "memory show" commands.

	  * Add freed regions totals to allocations and summary.

	  * Add totals for all allocations and not just the selected allocations.

	  Change-Id: I61d5a5112617b0733097f2545a3006a344b4032a

2016-01-14 16:00 +0000 [84b30c5e18]  Kevin Harwell <kharwell@digium.com>

	* bridge_basic: don't play an attended transfer fail sound after target hangs up

	  If the attended transfer destination answers (picks call up or goes to
	  voicemail) and then hangs up on the transferer then transferer hears the
	  fail sound.

	  This patch makes it so the fail sound is not played when the transfer
	  destination/target hangs up after answering.

	  ASTERISK-25697 #close

	  Change-Id: I97f142fe4fc2805d1a24b7c16143069dc03d9ded

2016-01-14 14:36 +0000 [c7caee6c4b]  Corey Farrell <git@cfware.com>

	* Remove *.gcna / *.gcno files from added module sources.

	  Asterisk uses a Makefile macro to associate additional sources with a
	  module.  This macro is responsible for creating clean targets but
	  previously left behind *.gcna and *.gcno files.

	  ASTERISK-25683 #close
	  Reported by yaron nahum

	  Change-Id: Idc0823fe80a25c42cefae901fde875e9fc38d8ea

2016-01-14 09:26 +0000 [68cad96ffd]  Rusty Newton <rnewton@digium.com>

	* func_channel: Add help text for undocumented CHANNEL function arguments

	  Adding help text documentation for:
	  * hangupsource
	  * appname
	  * appdata
	  * exten
	  * context
	  * channame
	  * uniqueid
	  * linkedid

	  ASTERISK-24097 #close
	  Reported by: Steven T. Wheeler
	  Tested by: Rusty Newton

	  Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d

2016-01-10 16:22 +0000 [8182146e85]  Daniel Journo <dan@keshercommunications.com>

	* pjsip:  Add option global/regcontext

	  Added new global option (regcontext) to pjsip. When set, Asterisk will
	  dynamically create and destroy a NoOp priority 1 extension
	  for a given endpoint who registers or unregisters with us.

	  ASTERISK-25670 #close
	  Reported-by: Daniel Journo

	  Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62

2016-01-12 11:14 +0000 [022423b98b]  Joshua Colp <jcolp@digium.com>

	* app: Queue hangup if channel is hung up during sub or macro execution.

	  This issue was exposed when executing a connected line subroutine.
	  When connected or redirected subroutines or macros are executed it is
	  expected that the underlying applications and logic invoked are fast
	  and do not consume frames. In practice this constraint is not enforced
	  and if not adhered to will cause channels to continue when they shouldn't.
	  This is because each caller of the connected or redirected logic does not
	  check whether the channel has been hung up on return. As a result the
	  the hung up channel continues.

	  This change makes it so when the API to execute a subroutine or
	  macro is invoked the channel is checked to determine if it has hung up.
	  If it has then a hangup is queued again so the caller will see it
	  and stop.

	  ASTERISK-25690 #close

	  Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea

2016-01-13 07:20 +0000 [79a7321a47]  Sean Bright <sean.bright@gmail.com>

	* res_musiconhold: Prevent multiple simultaneous reloads.

	  There are two ways in which the reload() function in res_musiconhold can be
	  called from the CLI:

	    * module reload res_musiconhold.so
	    * moh reload

	  In the former case, the module loader holds a lock that prevents multiple
	  concurrent calls, but in the latter there is no such protection.

	  This patch changes the 'moh reload' CLI command to invoke the module loader
	  directly, rather than call reload() explicitly.

	  ASTERISK-25687 #close

	  Change-Id: I408968b4c8932864411b7f9ad88cfdc7b9ba711c
2016-01-12 14:25 +0000 [1fffe71f77]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts".

	  PJPROJECT has a function available to dump the compile time
	  options used when building the library.

	  * Add CLI "pjsip show buildopts" command.

	  * Update contrib/scripts/autosupport to get pjproject information.

	  Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748

2016-01-12 10:36 +0000 [01c5e2a07e]  Mark Michelson <mmichelson@digium.com>

	* res_sorcery_realtime: Remove leading ^ requirement.

	  res_sorcery_realtime's search-by-regex callback performed a check to
	  ensure that the passed-in regex began with a caret (^). If it did not,
	  then no results would be returned.

	  This callback only started to become used when "like" support was added
	  to PJSIP CLI commands. The CLI command for listing objects would pass an
	  empty regex ("") to the sorcery backend if no "like" statement was
	  present. For most sorcery backends, this resulted in returning all
	  objects. However, for realtime, this resulted in returning no objects.

	  This commit seeks to fix the regression by removing the requirement from
	  res_sorcery_realtime for the passed-in-regex to begin with a caret.

	  ASTERISK-25689 #close
	  Reported by Marcelo Terres

	  Change-Id: I22b4dc5d7f3f11bb29ac2e42ef94682e9bab3b20

2016-01-07 11:57 +0000 [a41aab477a]  gtjoseph <george.joseph@fairview5.com>

	* pjsip_sdp_rtp:  Add option endpoint/bind_rtp_to_media_address

	  On a system with multiple ip addresses in the same subnet, if a
	  transport is bound to a specific ip address and endpoint/media_address
	   is set, the SIP/SDP will have the correct address in all fields but
	  the rtp stream MAY still originate from one of the other ip addresses,
	  most probably the "primary" ip address.  This happens because
	   res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
	  the "all" ip address (0.0.0.0 or ::).

	  The new option causes res_pjsip_sdp_rtp/create_rtp to call
	  ast_rtp_instance_new with the endpoint's media_address (if specified)
	  instead of the "all" address.  This causes the packets to originate from
	  the specified address.

	  ASTERISK-25632
	  ASTERISK-25637
	  Reported-by: Olivier Krief
	  Reported-by: Dan Journo

	  Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88

2016-01-08 16:59 +0000 [7760029f19]  Kevin Harwell <kharwell@digium.com>

	* pbx: Deadlock between contexts container and context_merge locks

	  Recent changes (ASTERISK-25394 commit 2bd27d12223fe33b58c453965ed5c6ed3af7c4f5)
	  introduced the possibility of a deadlock. Due to the mentioned modifications
	  ast_change_hints now needs to keep both merge/delete and state callbacks from
	  occurring while it executes. Unfortunately, sometimes ast_change_hints can be
	  called with the contexts container locked. When this happens it's possible for
	  another thread to grab the context_merge_lock before the thread calling into
	  ast_change_hints does and then try to obtain the contexts container lock. This
	  of course causes a deadlock between the two threads. The thread calling into
	  ast_change_hints waits for the other thread to release context_merge_lock and
	  the other thread is waiting on that one to release the contexts container lock.

	  Unfortunately, there is not a great way to fix this problem. When hints change,
	  the subsequent state callbacks cannot run at the same time as a merge/delete,
	  nor when the usual state callbacks do. This patch alleviates the problem by
	  having those particular callbacks (the ones run after a hint change) occur in a
	  serialized task. By moving the context_merge_lock to a task it can now safely be
	  attempted or held without a deadlock occurring.

	  ASTERISK-25640 #close
	  Reported by: Krzysztof Trempala

	  Change-Id: If2210ea241afd1585dc2594c16faff84579bf302

2016-01-10 17:08 +0000 [e9c2c1dc67]  Corey Farrell <git@cfware.com>

	* devicestate: Cleanup engine thread during graceful shutdown.

	  ASTERISK-25681 #close

	  Change-Id: I64337c70f0ebd8c77f70792042684607c950c8f1

2016-01-10 13:51 +0000 [90c0dcaee4]  Corey Farrell <git@cfware.com>

	* manager: Cleanup manager_channelvars during shutdown.

	  ASTERISK-25680 #close

	  Change-Id: I3251d781cbc3f48a6a7e1b969ac4983f552b2446

2016-01-10 13:27 +0000 [a868a381f0]  Corey Farrell <git@cfware.com>

	* res_calendar: Cleanup scheduler context at unload.

	  ASTERISK-25679 #close

	  Change-Id: I839159bf6882cccc1b23494c7aa2bc2a2624613f

2016-01-08 11:49 +0000 [a1c43022d2]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Revert DTLS negotiation changes.

	  Due to locking issues within pjnath these changes are being
	  reverted until pjnath can be changed.

	  ASTERISK-25645

	  Revert "res_rtp_asterisk.c: Fix DTLS negotiation delays."

	  This reverts commit 24ae124e4f7310cfa64c187b944b2ffc060da28d.

	  Change-Id: I2986cfb2c43dc14455c1bcaf92c3804f9da49705

	  Revert "res_rtp_asterisk: Resolve further timing issues with DTLS negotiation"

	  This reverts commit 965a0eee46d24321f74c244e23c5a5f45e67e12b.

	  Change-Id: Ie68fafde27dad4b03cb7a1e27ce2a8502c3f7bbe

2016-01-09 17:57 +0000 [220ba979cf]  gtjoseph <george.joseph@fairview5.com>

	* Revert "pjsip_location: Delete contact_status object when contact is deleted"

	  This reverts commit 0a9941de9d24093b5ff44096d1d7406f29d11e45.

	  Matt,

	  This patch causes another problem and should not have been needed.
	  Before this patch, persistent_endpoint_contact_deleted_observer WAS
	  deleting the contact_status when ast_sip_location_delete_contact was
	  called.  By deleting it yourself in ast_sip_location_delete_contact
	  it was gone before the observer could run and the observer therefore
	  was throwing an error and not sending stasis/AMI/statsd messages.

	  So, I don't think this was the cause of your original issue.  I also
	  had verified the contact AMI and statsd lifecycle and it was working.
	  I'll double check now though.

	  ASTERISK-25675
	  Reported-by: Daniel Journo

	  Change-Id: Ib586a6b7f90acb641b0c410f659743ab90e84f1a

2016-01-09 18:04 +0000 [26e0e113dc]  Corey Farrell <git@cfware.com>

	* pbx_dundi: Run cleanup on failed load.

	  During failed startup of pbx_dundi no cleanup was performed.  Add a call
	  to unload_module before returning AST_MODULE_LOAD_DECLINE.

	  ASTERISK-25677 #close

	  Change-Id: I8ffa226fda4365ee7068ac1f464473f1a4ebbb29

2016-01-09 13:28 +0000 [dc2c000fd5]  Corey Farrell <git@cfware.com>

	* res_crypto: Perform cleanup at shutdown.

	  This change causes res_crypto to unregister CLI at shutdown while still
	  preventing the module from being unloaded.

	  ASTERISK-25673 #close

	  Change-Id: Ie5d57338dc2752abfc0dd05d0eec86413f2304fc

2016-01-06 19:10 +0000 [0bca2a5c26]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Create human friendly serializer names.

	  PJSIP name formats:
	  pjsip/aor/<aor>-<seq> -- registrar thread pool serializer
	  pjsip/default-<seq> -- default thread pool serializer
	  pjsip/messaging -- messaging thread pool serializer
	  pjsip/outreg/<registration>-<seq> -- outbound registration thread pool
	  serializer
	  pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer
	  pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer
	  pjsip/session/<endpoint>-<seq> -- session thread pool serializer
	  pjsip/websocket-<seq> -- websocket thread pool serializer

	  Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084

2016-01-06 19:09 +0000 [f0f5fbbc01]  Richard Mudgett <rmudgett@digium.com>

	* Sorcery: Create human friendly serializer names.

	  Sorcery name formats:
	  sorcery/<type>-<seq> -- Sorcery thread pool serializer

	  Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47

2016-01-06 19:09 +0000 [b1c7ae9afc]  Richard Mudgett <rmudgett@digium.com>

	* Stasis: Create human friendly taskprocessor/serializer names.

	  Stasis name formats:
	  subm:<topic>-<seq> -- Stasis subscription mailbox task processor
	  subp:<topic>-<seq> -- Stasis subscription thread pool serializer

	  Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd

2016-01-07 16:15 +0000 [3e857bb347]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: New API for human friendly taskprocessor names.

	  * Add new API call to get a sequence number for use in human friendly
	  taskprocessor names.

	  * Add new API call to create a taskprocessor name in a given buffer and
	  append a sequence number.

	  Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9

2016-01-06 17:19 +0000 [84c245d38c]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Fix CLI "core show taskprocessors" output format.

	  Update the CLI "core show taskprocessors" output format to not be
	  distorted because UUID names are longer than previously used taskprocessor
	  names.

	  Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601

2016-01-07 21:07 +0000 [7d86979ea0]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Fix CLI "core show taskprocessors" unref.

	  Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5

2016-01-06 19:00 +0000 [1fb39aa8a0]  Richard Mudgett <rmudgett@digium.com>

	* ccss.c: Replace space in taskprocessor name.

	  The CLI "core ping taskprocessor" command does not work very
	  well with taskprocessor names that have spaces in them.  You
	  have to put quotes around the name so using tab completion
	  becomes awkward.

	  Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0

2016-01-07 20:44 +0000 [71bb7b9c40]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Sort CLI "core show taskprocessors" output.

	  Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e

2016-01-05 16:54 +0000 [b025e1982f]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock.

	  Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b

2015-12-16 11:25 +0000 [c5e16fe33a]  Mark Michelson <mmichelson@digium.com>

	* Alembic: Add PJSIP global keep_alive_interval.

	  The keep_alive_interval option was added about a year ago, but no
	  alembic revision was created to add the appropriate column to the
	  database.

	  This commit fixes the problem and adds the column. This was discovered
	  by running the testsuite with automatic conversion to realtime enabled.

	  Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a

2016-01-07 03:21 +0000 [6745cd6529]  Diederik de Groot <ddegroot@talon.nl>

	* include/asterisk/time.h: Renamed global declaration:tv

	  Renamed global declaration:tv to dummy_tv_var_for_types,
	  which would oltherwise cause 'shadow' warnings when 'tv'
	  was declared as a local variable elsewhere.

	  Added comment to note that dummy_tv_var_for_types is never
	  really exported and only used as a place holder.

	  ASTERISK-25627 #close

	  Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28

2016-01-07 15:37 +0000 [1afc8432dc]  Mark Michelson <mmichelson@digium.com>

	* PJSIP: Prevent deadlock due to dialog/transaction lock inversion.

	  A deadlock was observed where the monitor thread was stuck, therefore
	  resulting in no incoming SIP traffic being processed.

	  The problem occurred when two 200 OK responses arrived in response to a
	  terminating NOTIFY request sent from Asterisk. The first 200 OK was
	  dispatched to a threadpool worker, who locked the corresponding
	  transaction. The second 200 OK arrived, resulting in the monitor thread
	  locking the dialog. At this point, the two threads are at odds, because
	  the monitor thread attempts to lock the transaction, and the threadpool
	  thread loops attempting to try to lock the dialog.

	  In this case, the fix is to not have the monitor thread attempt to hold
	  both the dialog and transaction locks at the same time. Instead, we
	  release the dialog lock before attempting to lock the transaction.

	  There have also been some debug messages added to the process in an
	  attempt to make it more clear what is going on in the process.

	  ASTERISK-25668 #close
	  Reported by Mark Michelson

	  Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a

2016-01-07 09:39 +0000 [5d8c42c6d3]  Corey Farrell <git@cfware.com>

	* ast_format_cap_append_by_type: Resolve codec reference leak.

	  This resolves a reference leak caused by ASTERISK-25535.  The pointer
	  returned by ast_format_get_codec is saved so it can be released.

	  ASTERISK-25664 #close

	  Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec

2016-01-07 03:33 +0000 [7856762f2f]  Diederik de Groot <ddegroot@talon.nl>

	* main: Use ast_strdup instead of strdup

	  Fix compile error in main/utils.c because strdup was used in dummy_start

	  Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793

2016-01-06 07:12 +0000 [64b2046f3d]  Walter Doekes <walter+asterisk@wjd.nu>

	* Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts.

	  The spandspflow2pcap.py creates pcap files from fax.log files, generated
	  through 'fax set debug on' when receiving a fax. An example fax.log is
	  included as spandspflow2pcap.log.

	  The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a
	  recent version of SIPp.

	  ASTERISK-25660 #close

	  Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526

2016-01-04 04:26 +0000 [084563e136]  Aaron An <anjb@ti-net.com.cn>

	* cel/cel_radius: Fix wrong pointer.

	  The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter
	  y not the address of y.

	  I capture the radius UDP packet via tcpdump, and the AV pairs are not correct,
	  then i review the source code and compare it with cdr/cdr_radius.c. Fix it and
	   it works.

	  ASTERISK-25647 #close
	  Reported by: Aaron An
	  Tested by: Aaron An

	  Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0

2016-01-04 20:23 +0000 [36f1eaf0b5]  Corey Farrell <git@cfware.com>

	* main/pbx: Move hangup handler routines to pbx_hangup_handler.c.

	  This is the sixth patch in a series meant to reduce the bulk of pbx.c.
	  This moves hangup handler management functions to their own source.

	  Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104

2015-12-21 11:07 +0000 [90b06d1a3c]  Martin Tomec <tomec.martin@gmail.com>

	* app_queue: Add member flag "in_call" to prevent reading wrong lastcall time

	  Member lastcall time is updated later than member status. There was chance to
	  check wrapuptime for available member with wrong (old) lastcall time.
	  New boolean flag "in_call" is set to true right before connecting call, and
	  reset to false after update of lastcall time. Members with "in_call" set to true
	  are treat as unavailable.

	  ASTERISK-19820 #close

	  Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500

2016-01-04 19:46 +0000 [3507494b8a]  Corey Farrell <git@cfware.com>

	* main/pbx: Move dialplan application management routines to pbx_app.c.

	  This is the sixth patch in a series meant to reduce the bulk of pbx.c.
	  This moves dialplan application management functions to their own source.

	  Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c

2016-01-04 18:20 +0000 [54a8f1a396]  Corey Farrell <git@cfware.com>

	* main/pbx: Move switch routines to pbx_switch.c.

	  This is the fifth patch in a series meant to reduce the bulk of pbx.c.
	  This moves ast_switch functions to their own source.

	  Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e

2016-01-04 18:00 +0000 [c3c8b8e41d]  Corey Farrell <git@cfware.com>

	* main/pbx: Move timing routines to pbx_timing.c.

	  This is the fourth patch in a series meant to reduce the bulk of pbx.c.
	  This moves pbx timing functions to their own source.

	  Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6

2015-12-30 10:49 +0000 [6d18fe151c]  gtjoseph <george.joseph@fairview5.com>

	* voicemail: Move app_voicemail / res_mwi_external conflict to runtime

	  The menuselect conflict between app_voicemail and res_mwi_external
	  makes it hard to package 1 version of Asterisk.  There no actual
	  build dependencies between the 2 so moving this check to runtime
	  seems like a better solution.

	  The ast_vm_register and ast_vm_greeter_register functions in app.c
	  were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there
	  is already a voicemail module registered. The modules' load_module
	  functions were then modified to return DECLINE instead of -1 to the
	  loader.  Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE,
	  the modules were incorrectly causing Asterisk to stop so this needed
	  to be cleaned up anyway.

	  Now you can build both and use modules.conf to decide which voicemail
	  implementation to load.

	  The default menuselect options still build app_voicemail and not
	  res_mwi_external but if both ARE built, res_mwi_external will load
	  first and become the voicemail provider unless modules.conf rules
	  prevent it.  This is noted in CHANGES.

	  Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247

2016-01-04 16:15 +0000 [5ee5c3739e]  Corey Farrell <git@cfware.com>

	* main/pbx: Move variable routines to pbx_variables.c.

	  This is the third patch in a series meant to reduce the bulk of pbx.c.
	  This moves channel and global variable routines to their own source.

	  Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6

2015-12-04 17:22 +0000 [f88b952093]  Richard Mudgett <rmudgett@digium.com>

	* app_dial: Immediately exit dial if the caller is already hung up.

	  If a caller hangs up before dial is executed within an AGI then the AGI
	  has likely eaten all queued frames before executing the dial in DeadAGI
	  mode.  With the caller hung up and no pending frames from the caller's
	  read queue, dial would not know that the call has hung up until a called
	  channel answers.  It is rather annoying to whoever just answered the
	  non-existent call.

	  Dial should not continue execution in DeadAGI mode, hangup handlers, or
	  the h exten.

	  * Added a check early in dial to abort dialing if the caller has hungup.

	  ASTERISK-25307 #close
	  Reported by: David Cunningham

	  Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418

2016-01-02 10:26 +0000 [e9dd16364e]  Matt Jordan <mjordan@digium.com>

	* main/cdr: Allow setting properties on a finalized CDR if it is the last one

	  Prior to this patch, we explicitly disallowed setting any properties on a
	  finalized CDR. This seemed like a good idea at the time; in practice, it was
	  more restrictive.

	  There are weird and strange scenarios where setting a property on a finalized
	  CDR is definitely wrong. For example, we may Fork a CDR, finalizing the
	  previous one, then change a property. In said case, the old CDR is supposed
	  to now be 'immutable' (so to speak), and should not be updated. From the
	  perspective of the code, a forked CDR that is finalized is just finalized.
	  Hence why we decided these should not be updated.

	  In practice, it is much more common to want to set a property on a CDR in
	  the h extension or in a hangup handler. Disallowing a common scenario to make
	  an esoteric behaviour work isn't good. This patch fixes this by allowing
	  callers to set a property IF we are the last CDR in the chain. This preserves
	  the finalized CDR if it was forked, while allowing the more common case to
	  function.

	  ASTERISK-25458 #close

	  Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9

2016-01-02 10:23 +0000 [153547a9b1]  Matt Jordan <mjordan@digium.com>

	* main/cdr: Set the end time on a CDR if endbeforehexten is Yes

	  Prior to this patch, the CDR engine attempted to set the end time on a CDR
	  that was executing hangup logic and with endbeforehexten set to Yes by
	  calling a function that inspects the properties on the Party A snapshot to
	  determine if we are ready to set the end time. That always failed. This is
	  because a Party A snapshot is not updated for CDRs that are executing hangup
	  logic with endbeforehexten=Yes.

	  Instead of calling a function that looks at the Party A snapshot, we just
	  simply set the end time on the CDR. This is safe to call multiple times, and is
	  safe to call at this point as we know that (a) we are executing hangup logic,
	  and (b) we are supposed to set the end time at this point.

	  ASTERISK-25458

	  Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3

2015-12-30 20:51 +0000 [f9bfc2450e]  Corey Farrell <git@cfware.com>

	* main/pbx: Move custom function routines to pbx_functions.c.

	  This is the second patch in a series meant to reduce the bulk of pbx.c.
	  This moves custom function management routines to their own source.

	  Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177

2016-01-01 05:25 +0000 [3fd528dddf]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* Happy new year 2016.

	  Change-Id: I22d3c90f6f27df82e915bbf81c1d91221f7a945e

2015-12-13 13:09 +0000 [9cdf3ec19d]  Matt Jordan <mjordan@digium.com>

	* res_pjsip_history: Add a module that provides PJSIP history for debugging

	  This patch adds a new module, res_pjsip_history, that provides a slightly
	  better way of debugging SIP message traffic on a busy Asterisk system. The
	  existing mechanisms all rely on passively dumping a SIP message to the CLI.
	  While this is perfectly fine for logging purposes and well controlled
	  environments, on many installations, the amount of SIP messages Asterisk
	  receives will quickly swamp the CLI. This makes it difficult to view/capture
	  those messages that you want to diagnose in real time.

	  This patch provides another way of handling this. When enabled, the module
	  will store SIP message traffic in memory. This traffic can then be queried
	  at leisure.

	  In order to make the querying useful, a CLI command has been implemented,
	  'pjsip show history', that supports a basic expression syntax similar to
	  SQL or other query languages. A small number of useful fields have been
	  added in this initial patch; additional fields can easily be added in
	  later improvements. Those fields are:
	   - number: The entry index in the history
	   - timestamp: The time the message was recieved
	   - addr: The source/destination address of the message
	   - sip.msg.request.method: The request method
	   - sip.msg.call-id: The Call-ID header

	  Note - this is a resurrection of the module initially proposed on Review Board
	  here: https://reviewboard.asterisk.org/r/4053/

	  Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36

2015-12-28 19:18 +0000 [5e67e51c6a]  gtjoseph <george.joseph@fairview5.com>

	* main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c

	  We joked about splitting pbx.c into multiple files but this first step was
	  fairly easy.  All of the pbx_builtin dialplan applications have been moved
	  into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins()
	  is called by asterisk.c just after load_pbx().

	  A few functions were renamed and are cross-exposed between the 2 source files.

	  Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a

2015-12-28 14:02 +0000 [a05bb258b1]  Joshua Colp <jcolp@digium.com>

	* test_time: Provide a timeout when waiting.

	  The test_timezone_watch unit test is written to expect a
	  condition to be signaled when the inotify daemon thread runs.
	  There exists a small window where the test_timezone_watch
	  thread can signal the inotify daemon thread while it is not
	  reading on the underlying file descriptor. If this occurs
	  the test_timezone_watch thread will wait indefinitely for a
	  signal that will never arrive.

	  This change adds a timeout to the condition so it will return
	  regardless after a period of time.

	  Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390

2015-12-24 20:26 +0000 [96b32e0321]  Matt Jordan <mjordan@digium.com>

	* tests/test_stasis_endpoints: Remove expected duplicate events

	  The cache_clear test was written to expect duplicate Stasis messages
	  sent from the technology endpoint to the all caching topic. This patch
	  fixes the test to no longer expect these duplicate messages.

	  ASTERISK-25137

	  Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981

2015-12-24 22:19 +0000 [3bddcc0219]  Dade Brandon <dade@xencall.com>

	* res_http_websocket.c: prevent avoidable disconnections caused by write errors

	  Updated ast_websocket_write to encode the entire frame in to one
	  write operation, to ensure that we don't end up with a situation
	  where the websocket header has been sent, while the body can not
	  be written.

	  Previous to August's patch in commit b9bd3c14, certain network
	  conditions could cause the header to be written, and then the
	  sub-sequent body to fail - which would cause the next successful
	  write to contain a new header, and a new body (resulting in
	  the peer receiving two headers - the second of which would be
	  read as part of the body for the first header).

	  This was patched to have both write operations individually fail
	  by closing the websocket.

	  In a case available to the submitter of this patch, the same
	  body which would consistently fail to write, would succeed
	  if written at the same time as the header.

	  This update merges the two operations in to one, adds debug messages
	  indicating the reason for a websocket connection being closed during
	  a write operation, and clarifies some variable names for code legibility.

	  Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598

2015-05-27 13:22 +0000 [22db16fa81]  gtjoseph <george.joseph@fairview5.com>

	* endpoint/stasis: Eliminate duplicate events on endpoint status change

	  When an endpoint is created, its messages are forwarded to both the tech
	  endpoint topic and the all endpoints topic. This is done so that various
	  parties interested in endpoint messages can subscribe to just the tech
	  endpoint and receive all messages associated with that particular technology,
	  as opposed to subscribing to the all endpoints topic. Unfortunately, when the
	  tech endpoint is created, it also forwards all of its messages to the all
	  topic. This results in duplicate messages whenever an endpoint publishes its
	  messages.

	  This patch resolves the duplicate message issue by creating a new function
	  for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts
	  as a normal caching topic, save that it no longer forwards messages it receives
	  to the all endpoints topic. This allows it to act as an aggregation "sink",
	  while preserving the necessary caching behaviour.

	  ASTERISK-25137 #close
	  Reported-by: Vitezslav Novy

	  ASTERISK-25116 #close
	  Reported-by: George Joseph <george.joseph@fairview5.com>
	  Tested-by: George Joseph <george.joseph@fairview5.com>

	  Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b

2015-12-27 22:38 +0000 [6b08f01c60]  Corey Farrell <git@cfware.com>

	* Remove res_jabber file that was left behind.

	  Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b

2015-12-26 09:24 +0000 [d4b10cfb3e]  Ward van Wanrooij <ward@ward.nu>

	* chan_sip: option 'notifyringing' change and doc fix

	  In the sample sip.conf this is written with regard to notifyringing:
	  ;notifyringing = no ; Control whether subscriptions already INUSE get sent
	  RINGING when another call is sent (default: yes)

	  However, this setting changes whether or not any RINGING indications are sent
	  to subscriptions. There is no separate configurable setting that allows
	  to control whether INUSE subscriptions also get sent RINGING. This is however
	  a useful option, to see (using BLF) if somebody else is able to handle an
	  incoming call or if everybody is busy.

	  This patch corrects the documentation for notifyringing (so the documentation
	  matches the functionality) and make notifyringing a tri-state option, by adding
	  the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing =
	  notinuse, only subscriptions that are not INUSE are sent the RINGING signal.

	  The default setting for notifyringing remains set to yes, so the default
	  behaviour is not affected.

	  ASTERISK-25558

	  Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa

2015-12-25 09:56 +0000 [6dc21bbf00]  Dade Brandon <dade@xencall.com>

	* chan_sip.c: fix websocket_write_timeout default value

	  websocket_write_timeout was not being set to its default value
	  during sip config reload, which meant that prior to this commit,
	  1) the default value of 100 was not used, unless an invalid value
	  (or 1) was specified in sip.conf for websocket_write_timeout, and
	  2) if the websocket_write_timeout directive was removed from sip.conf
	  without a full restart of asterisk, then the previous value would
	  continue to be used indefinitely.

	  This essentially lead to a 0ms write timeout (the first write attempt
	  in ast_careful_fwrite must have succeeded) in websocket write requests
	  from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf.

	  Changes to websocket_write_timeout still only apply to new websocket
	  sessions, after the sip reload -- timeouts on existing sessions are
	  not adjusted during sip reload.

	  Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953

2015-12-23 17:40 +0000 [8eb5da0679]  Richard Mudgett <rmudgett@digium.com>

	* bridge_basic.c: Fix GOTO_ON_BLINDXFR

	  Use of GOTO_ON_BLINDXFR would not work at all.  The target location would
	  never be executed by the transferring channel.

	  * Made feature_blind_transfer() call ast_bridge_set_after_go_on() with
	  valid context, exten, and priority parameters from the transferring
	  channel.

	  * Renamed some feature_blind_transfer() local variables for clarity.

	  ASTERISK-25641 #close
	  Reported by Dmitry Melekhov

	  Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac

2015-12-24 12:19 +0000 [2df4ad647c]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_location: Delete contact_status object when contact is deleted

	  In 450579e908, a change was made that removed the deletion of the
	  'contact_status' object when a 'contact' object is deleted in sorcery.
	  This unfortunately means that the 'contact_status' object persists, even when
	  something has explicitly removed a contact. The result is that the state of
	  the contact will not be regenerated if that contact is re-created, and the
	  stale state will be reported/used for that contact. It also results in
	  no ContactStatusChanged events being generated for either ARI or AMI.

	  This patch restores the deletion logic that was removed. Doing so now
	  results in the expected events being generated again.

	  Change-Id: I28789a112e845072308b5b34522690e3faf58f07

2015-12-24 10:18 +0000 [b8876711f3]  Kevin Harwell <kharwell@digium.com>

	* res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef

	  Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151

2015-12-20 21:33 +0000 [ca394161cf]  Dade Brandon <dade@xencall.com>

	* app_amd: Correct maximum_number_of_words functionality & documentation

	  - The maximum_number_of_words was previously documented as being
	  the number of words that when exceeded, would result in the AMD
	  application returning that the audio represents a machine.

	  This was inconsistent with its actual functionality - it was
	  a number of words that when REACHED, would result in determination
	  as a machine.

	  This update corrects the functionality to match the previously
	  documented functionality.  This is a backwards incompatible change
	  in configuration file, and has been added to UPGRADE.txt as a result.

	  The sample configuration file and application defaults have been updated
	  so that the default value is now 2, which reflects the same default
	  functionality as previous versions.

	  - Update documentation for silence_threshold, which previously implied
	  that it was measuring time, rather than noise averages in the sample.

	  - Update the comments in amd.conf.sample.

	  ASTERISK-25639 #close
	  Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093

2015-12-17 19:05 +0000 [648ca2b1b8]  Dade Brandon <dade@xencall.com>

	* res_rtp_asterisk: Resolve further timing issues with DTLS negotiation

	  Resolves an edge case dtls negotiation delay for certain networks which
	  somehow manage to drop the rtcp side's packet when these are both sent
	  ast_rtp_remote_address_set, causing it to have to time-out and restart
	  the handshake.

	  Move dtls pending bio flush in to it's own function, and call it from
	  ast_rtp_on_ice_complete, when we're rtp->ice, rather than when
	  ast_rtp_remote_address_set.

	  Keep the existing flush from the recent change to res_rtp_remote_address_set
	  if ice is not being used.

	  ASTERISK-25614 #close
	  Reported-by: XenCALL
	  Tested by: XenCALL

	  Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5

2015-12-05 10:01 +0000 [902309fd04]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Add support for a full backend cache.

	  This change introduces the configuration option 'full_backend_cache'
	  which changes the cache to be a full mirror of the backend instead
	  of a per-object cache. This allows all sorcery retrieval operations
	  to be carried out against it and is useful for object types which
	  are used in a "retrieve all" or "retrieve some" pattern.

	  ASTERISK-25625 #close

	  Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5

2015-12-17 10:25 +0000 [a2431f83ef]  Joshua Colp <jcolp@digium.com>

	* rtp_engine: Ignore empty filenames in DTLS configuration.

	  When applying an empty DTLS configuration the filenames in the
	  configuration will be empty. This is actually valid to do and
	  each filename should simply be ignored.

	  Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539

2015-12-17 08:10 +0000 [d2c8614122]  Joshua Colp <jcolp@digium.com>

	* chan_sip: Enable WebSocket support by default.

	  Per the documentation the WebSocket support in chan_sip is
	  supposed to be enabled by default but is not. This change
	  corrects that.

	  Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423

2015-12-14 12:04 +0000 [d17d9a9288]  Joshua Colp <jcolp@digium.com>

	* json: Audit ast_json_* usage for thread safety.

	  The JSON library Asterisk uses, jansson, is not thread
	  safe for us in a few ways. To help with this wrappers for JSON
	  object reference count increasing and decreasing were added
	  which use a global lock to ensure they don't clobber over
	  each other. This does not extend to reference count manipulation
	  within the jansson library itself. This means you can't safely
	  use the object borrowing specifier (O) in ast_json_pack and
	  you can't share JSON instances between objects.

	  This change removes uses of the O specifier and replaces them
	  with the o specifier and an explicit ast_json_ref. Some cases
	  of instance sharing have also been removed.

	  ASTERISK-25601 #close

	  Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1

2015-12-16 11:28 +0000 [cfb34adb83]  Mark Michelson <mmichelson@digium.com>

	* Alembic: Increase column size of PJSIP AOR "contact".

	  When running the PJSIP AMI "show_endpoint" test with automatic
	  conversion to realtime, the test would fail. This was because the AOR
	  "contact" column was sized at 40, and the configured contact was larger
	  than that.

	  This commit increases the size of the contact column to 255 characters.

	  Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1

2015-12-14 13:53 +0000 [32ec83f37f]  server-pandora <server-pandora@xencall.com>

	* res_rtp_asterisk.c: Fix DTLS negotiation delays.

	  - Trigger pending DTLS packets to send out, once the RTP instance's remote
	    address is set.
	  - Avoids locking the DTLS structure unnecessarily by only doing this if
	    DTLS is passive.
	  - Add DTLS locks around the structurally sensitive calls in the SSL
	    portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock
	    inside of itself, and we're dealing with the SSL BIO in at least two
	    threads.

	  WebRTC channels may receive a DTLS handshake before
	  ast_rtp_remote_address_set is called, which causes there to be a pending
	  response to send out.   Previous to 1ad827, this was handled by calling
	  dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP
	  packet could trigger the pending handshake response.  Since that was
	  rightfully removed, whenever the DTLS handshake is received before the
	  remote address is set, we would have to wait until another SSL packet
	  arrives.

	  As of Chrome M47's optimizations to their handshake process, WebRTC
	  conversations between Chrome M47+ and Asterisk, where Asterisk is passive,
	  experience a 1 second delay without this patch, because the SSL handshake
	  is received before ICE negotation stores the remote_address, and the next
	  SSL packet isn't received until after a 1 second timeout in Chrome, which
	  causes a new handshake request.

	  ASTERISK-25614 #close

	  Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908

2015-12-08 13:04 +0000 [52ca6fb94a]  sungtae kim <pchero21@gmail.com>

	* AMI: Fixed OriginateResponse message

	  When the asterisk sending OriginateResponse message,
	  it doesn't set the "Uniqueid".
	  And it didn't support correct response message for
	  Application originate.

	  ASTERISK-25624 #close

	  Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d

2015-12-14 15:25 +0000 [eccdf2250b]  Richard Mudgett <rmudgett@digium.com>

	* Fix sscanf() format string type mismatch.

	  ASTERISK-25615
	  Reported by: George Joseph

	  Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b

2015-12-14 06:26 +0000 [3e7522533c]  Carlos Oliva <carlos.oliva@invoxcontact.com>

	* app_queue: update RT members when the 1st call joins a queue with no agents

	  If a call enters on a queue and the members on that queue are updated in
	  realtime (ex: using mysql inserting a new agent) the queue members are
	  never refreshed and the call will stay in the queue until other event occurs.
	  This happens only if this is the first call of the queue and there is no
	  agents servicing.
	  This patch prevent this issue, ensuring realtime members are updated if
	  there is one call in the queue and no available agents

	  ASTERISK-25442 #close

	  Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682

2015-12-13 13:13 +0000 [9a96a86e2d]  Matt Jordan <mjordan@digium.com>

	* main/utils: Don't emit an ERROR message if the read end of a pipe closes

	  An ERROR or WARNING message should generally indicate that something has gone
	  wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not
	  in control of when the far end closes its reading on a file descriptor. If the
	  far end does close the file descriptor in an unclean fashion, this isn't a bug
	  or error in Asterisk, particularly when the situation can be gracefully
	  handled in Asterisk.

	  Currently, when this happens, a user would see the following somewhat cryptic
	  ERROR message:

	    "utils.c: write() returned error: Broken pipe"

	  There's a few problems with this:
	  (1) It doesn't provide any context, other than 'something broke a pipe'
	  (2) As noted, it isn't actually an error in Asterisk
	  (3) It can get rather spammy if the thing breaking the pipe occurs often, such
	      as a FastAGI server
	  (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even
	      mask legitimate issues

	  This patch changes ast_carefulwrite to only log an ERROR if we actually had one
	  that was reasonably under our control. For debugging purposes, we still emit
	  a debug message if we detect that the far side has stopped reading.

	  Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566

2015-12-12 11:08 +0000 [3e6637feb5]  gtjoseph <george.joseph@fairview5.com>

	* pjsip/config_transport: Check pjproject version at runtime for async ops

	  pjproject < 2.5.0 will segfault on a tls transport if async_operations
	  is greater than 1.  A runtime version check has been added to throw
	  an error if the version is < 2.5.0 and async_operations > 1.

	  To assist in the check, a new api "ast_compare_versions" was added
	  to utils which compares 2 major.minor.patch.extra version strings.

	  ASTERISK-25615 #close

	  Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98
	  Reported-by: George Joseph
	  Tested-by: George Joseph

2015-12-10 11:44 +0000 [ceebdfce40]  Jonathan Rose <jrose@digium.com>

	* chan_sip: Add TCP/TLS keepalive to TCP/TLS server

	  Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously
	  this option was only being set on session sockets.
	  http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/
	  According to the link above, the SO_KEEPALIVE option is useful for knowing
	  when a TCP connected endpoint has severed communication without indicating
	  it or has become unreachable for some reason. Without this patch, keep
	  alive is not set on the socket listening for incoming TCP sessions and
	  in Komatsu's report this resulted in the thread listening for TCP becoming
	  stuck in a waiting state.

	  ASTERISK-25364 #close
	  Reported by: Hiroaki Komatsu

	  Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36
2015-12-07 13:07 +0000 [fcaebb0e43]  Corey Farrell <git@cfware.com>

	* app_meetme: Set default value for audio_buffers.

	  The default value was never set for audio_buffers, causing bad
	  audio quality.  This ensures the default is always set.

	  ASTERISK-25569 #close

	  Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44
2015-12-09 09:48 +0000 [5790700497]  Tyler Cambron <tcambron@digium.com>

	* res_chan_stats: Fix bug to send correct statistics to StatsD

	  Fixed a bug that originally would show a negative number of
	  active calls occuring in Asterisk. A gauge is persistent so
	  incrementing and decrementing it results in a more consistent
	  performance. Also changed to the call to StatsD to use
	  ast_statsd_log_string() so that a "+" could be sent to StatsD.

	  ASTERISK-25619 #close

	  Change-Id: Iaaeff5c4c6a46535366b4d16ea0ed0ee75ab2ee7
2015-12-08 17:49 +0000 [a987434564]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Add existence and readablity checks for tls related files

	  Both transport and endpoint now check for the existence and readability
	  of tls certificate and key files before passing them on to pjproject.
	  This will cause the object to not load rather than waiting for pjproject
	  to discover that there's a problem when a session is attempted.

	  NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located
	  in build_peer which is gigantic and I didn't want to disturb it.
	  Error messages will emit but it won't interrupt chan_sip loading.

	  ASTERISK-25618 #close

	  Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9
	  Reported-by: George Joseph
	  Tested-by: George Joseph

2015-12-02 12:42 +0000 [be693539c3]  Eugene Voityuk <eugene@thirdlane.com>

	* chan_sip.c: Start ICE negotiation when response is sent or received.

	  The current logic for ICE negotiation starts it
	  when receiving an SDP with ICE candidates. This is
	  incorrect as ICE negotiation can only start when each 
	  call party have at least one pair of local and remote 
	  candidate. Starting ICE negotiation early would result 
	  in negotiation failure and ultimately no audio.

	  This change makes it so ICE negotiation is only started
	  when a response with SDP is received or when a response
	  with SDP is sent.

	  ASTERISK-24146

	  Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca

2015-12-08 01:57 +0000 [59a91c350a]  Filip Jenicek <phill@janevim.cz>

	* chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c)

	  Asterisk may crash when calling ast_channel_get_t38_state(c)
	  on a locked channel which is being hung up.

	  ASTERISK-25609 #close

	  Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b
2015-12-08 11:03 +0000 [28ab03fbf7]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls

	  See ASTERISK-25615.
	  If the transport protocol is tls and async_operations > 1, pjproject
	  will segfault if more than one operation is attempted on the same socket.
	  Until this is fixed upstream, a check has been added to throw an error
	  if a tls transport config has async_operations set to > 1.

	  ASTERISK-25615

	  Change-Id: I76b9a5b2a5a0054fe71ca5851e635f2dca7685a6
	  Reported-by: George Joseph
	  Tested-by: George Joseph

2015-12-08 08:39 +0000 [55dd7125b3]  Alexander Traud <pabstraud@compuserve.com>

	* codec_resample: Increase buffer for Opus Codec with FEC.

	  ASTERISK-25599 #close

	  Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e

2015-12-08 03:46 +0000 [64f899e5f3]  Alexander Traud <pabstraud@compuserve.com>

	* translate: Avoid a warning message when doing FEC within Opus Codec.

	  ASTERISK-25616 #close

	  Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319

2015-12-04 15:36 +0000 [65c8147952]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip: Fix crash involving the bogus peer during sip reload.

	  A crash happens sometimes when performing a CLI "sip reload".  The bogus
	  peer gets refreshed while it is in use by a new call which can cause the
	  crash.

	  * Protected the global bogus peer object with an ao2 global object
	  container.

	  ASTERISK-25610 #close

	  Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed

2015-11-13 07:58 +0000 [48c065e46d]  Christof Lauber <christof.lauber@annax.ch>

	* chan_sip: Support parsing of Q.850 reason header in SIP BYE and CANCEL requests.

	  Current support for reason header did work only in SIP responses.
	  According to RFC3336 the reason header might appear in any SIP request.
	  But it seems to make most sence in BYE and CANCEL so parasing is done
	  there too (if use_q850_reason=yes).

	  Change-Id: Ib6be7b34c23a76d0e98dfd0816c89931000ac790

2015-12-06 16:35 +0000 [75c800eb28]  Matt Jordan <mjordan@digium.com>

	* Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state"

	  This reverts commit f42d22d3a1ca5c8ea73df99a50c6a28caa8f8749.

	  Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks
	  in core_unreal/chan_local. Local channels attempt to reach across both their
	  peer and the peer's bridge to inspect T.38 state. Given the propensity of
	  Local channel chains, managing the locking situation in such a scenario is
	  practically infeasible.

	  Change-Id: I932107387c13aad2c75a7a4c1e94197a9d6d8a51

2015-12-04 16:23 +0000 [4be231e82f]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip/contacts/statsd:  Make contact lifecycle events more consistent

	  It will never be perfect or even pretty, mostly because of the differences
	  between static and dynamic contacts.

	  Created:

	  Can't use the contact or contact_status alloc functions
	  because the objects come and go regardless of the actual state.

	  Can't use the contact_apply_handler, ast_sip_location_add_contact or
	  a sorcery created handler because they only get called for dynamic
	  contacts.  Similarly, permanent_uri_handler only gets called for
	  static contacts.

	  So, Matt had it right. :)  ast_res_pjsip_find_or_create_contact_status is
	  the only place it can go and not have duplicated code.  Both
	  permanent_uri_handler and contact_apply_handler call find_or_create.

	  Removed:

	  Can't use the destructors for the same reason as above.  The only
	  place to put this is in persistent_endpoint_contact_deleted_observer
	  which I believe is the "correct" place but even that will handle only
	  dynamic contacts.  This doesn't called on shutdown however.  There is
	  no hook to use for static contacts that may be removed because of a
	  config change while asterisk is in operation.

	  I moved the cleanup of contact_status from ast_sip_location_delete_contact
	  to the handler as well.

	  Status Change and RTT:

	  Although they worked fine where they were (in update_contact_status) I
	  moved them to persistent_endpoint_contact_status_observer to make it
	  more consistent with removed.  There was logic there already to detect
	  a state change.

	  Finally, fixed a nit in permanent_uri_handler rmudgett reported
	  eralier.

	  ASTERISK-25608 #close

	  Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d
	  Reported-by: George Joseph
	  Tested-by: George Joseph

2015-11-21 06:08 +0000 [63c6d39a3e]  Alexander Traud <pabstraud@compuserve.com>

	* res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8.

	  ASTERISK-25584 #close

	  Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91

2015-11-28 08:46 +0000 [f42d22d3a1]  Matt Jordan <mjordan@digium.com>

	* bridges/bridge_t38: Add a bridging module for managing T.38 state

	  When 4875e5ac32 was merged, it fixed several issues with a direct media bridge
	  transitioning to handling a T.38 fax. However, it uncovered a race condition
	  caused by the bridging core. When a channel involved in a T.38 fax leaves a
	  bridge, the frame queued by the channel driver that should inform the far side
	  that it is no longer in a T.38 fax may not make it across the bridge. The
	  bridging framework is *extremely* aggressive in tearing down the bridge, and
	  control frames that are currently in flight *may* get dropped.

	  This patch adds a new module to the bridging framework, bridge_t38. This module
	  maintains some notion of the T.38 state for the two channels in a bridge. When
	  the bridge detects that it is being torn down or when one of the two channels
	  leaves, it informs the respective channel(s) that they should stop faxing. This
	  ensures that channels switch back to audio if they survive and are ejected out
	  of a bridge while faxing.

	  ASTERISK-25582

	  Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0

2015-11-21 05:35 +0000 [dcc01bc0a7]  Alexander Traud <pabstraud@compuserve.com>

	* res_format_attr_opus: Update to latest RFC 7587.

	  Beside that, the format-attribute module sends only non-default values in the
	  line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore,
	  previously the parameter stereo was not parsed when being the first parameter.

	  ASTERISK-25583 #close

	  Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73
2015-12-02 14:11 +0000 [69457b8d61]  Jonathan Rose <jrose@digium.com>

	* Fix crash in audiohook translate to slin

	  This patch fixes a crash which would occur when an audiohook was
	  applied to a channel using an audio codec that could not be translated
	  to signed linear (such as when using pass-through codecs like OPUS or
	  when the codec translator module for the format in use is not loaded).

	  ASTERISK-25498 #close
	  Reported by: Ben Langfeld

	  Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384
2015-12-03 12:07 +0000 [5959186017]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip: Use a MD5 hash for static Contact IDs

	  When 90d9a70789 was merged, it mostly tested dynamic contacts created as
	  a result of registering a PJSIP endpoint. Contacts generated in this
	  fashion typically have a long alphanumeric string as their object identifier,
	  which maps reasonably well for StatsD. Unfortunately, this doesn't work in the
	  general case. StatsD treats both '.' and ':' characters as special characters.
	  In particular, having a ':' appear in the middle of a StatsD metric will
	  result in the metric being rejected.

	  This causes some obvious issues with SIP URIs.

	  The StatsD API should not be responsible for escaping the metric name passed
	  to it. The metric is treated as a single long string, and it would be
	  challenging to know what to escape in the string passed to the function.
	  Likewise, we don't want to escape the metric in PJSIP, as that involves
	  overhead that is wasted when either res_statsd isn't loaded or enabled.

	  This patch takes an alternative approach. The Contact ID has been changed
	  to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the
	  aforementioned special characters, (b) can be done on Contact creation,
	  which has minimal impact on run-time performance, and (c) also conforms to an
	  earlier commit that changed the ID for dynamic contacts.

	  The downside of this is that StatsD users will have to map SHA1 hashes back to
	  the Contacts that are emitting the statistics. To that end, the CLI commands
	  have been updated to include the first 10 characters of the MD5 hash, which
	  should be enough to match what is shown in Graphite (or some other StatsD
	  backend).

	  ASTERISK-25595 #close

	  Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2
	  Reported-by: Matt Jordan
	  Tested-by: George Joseph

2015-11-30 22:19 +0000 [bd265a90be]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Update logging to show contact->uri in messages

	  An earlier commit changed the id of dynamic contacts to contain
	  a hash instead of the uri.  This patch updates status change
	  logging to show the aor/uri instead of the id.  This required
	  adding the aor id to contact and contact_status and adding
	  uri to contact_status.  The aor id gets added to contact and
	  contact_status in their allocators and the uri gets added to
	  contact_status in pjsip_options when the contact_status is
	  created or updated.

	  ASTERISK-25598 #close

	  Reported-by: George Joseph
	  Tested-by: George Joseph

	  Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511

2015-12-01 16:11 +0000 [b5281b74e0]  Jonathan Rose <jrose@digium.com>

	* Unset BRIDGEPEER when leaving a bridge

	  Currently if a channel is transferred out of a bridge, the BRIDGEPEER
	  variable (also BRIDGEPVTCALLID) remain set even once the channel is
	  out of the bridge. This patch removes these variables when leaving
	  the bridge.

	  ASTERISK-25600 #close
	  Reported by: Mark Michelson

	  Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da

2015-11-30 14:22 +0000 [59ba84e5cd]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_memory_cache.c: Fix off nominal ref leak.

	  Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49

2015-11-30 16:42 +0000 [ef77439e39]  Richard Mudgett <rmudgett@digium.com>

	* sched.c: Make not return a sched id of 0.

	  According to the API doxygen a sched ID of 0 is valid.  Unfortunately, 0
	  was never returned historically and several users incorrectly coded usage
	  of the returned sched ID assuming that 0 was invalid.

	  ASTERISK-25476

	  Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20

2015-11-25 12:23 +0000 [145d10a5d0]  Richard Mudgett <rmudgett@digium.com>

	* Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions)

	  chan_sip.c:
	  * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to
	  ao2 conversion.

	  * Initialize register scheduler ids earlier because of ASTOBJ to ao2
	  conversion.

	  chan_skinny.c:
	  * Fix more scheduler usage for the valid 0 id value.

	  ASTERISK-25476

	  Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95

2015-11-24 12:44 +0000 [fa20729032]  Richard Mudgett <rmudgett@digium.com>

	* Audit improper usage of scheduler exposed by 5c713fdf18f.

	  channels/chan_iax2.c:
	  * Initialize struct chan_iax2_pvt scheduler ids earlier because of
	  iax2_destroy_helper().

	  channels/chan_sip.c:
	  channels/sip/config_parser.c:
	  * Fix initialization of scheduler id struct members.  Some off nominal
	  paths had 0 as a scheduler id to be destroyed when it was never started.

	  chan_skinny.c:
	  * Fix some scheduler id comparisons that excluded the valid 0 id.

	  channel.c:
	  * Fix channel initialization of the video stream scheduler id.

	  pbx_dundi.c:
	  * Fix channel initialization of the packet retransmission scheduler id.

	  ASTERISK-25476

	  Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8

2015-12-01 07:55 +0000 [b24f2f4c2e]  Alexander Traud <pabstraud@compuserve.com>

	* codec_resample: Increase buffer for Opus Codec.

	  ASTERISK-25599 #close

	  Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10

2015-11-30 11:13 +0000 [e5723d2776]  gtjoseph <george.joseph@fairview5.com>

	* dns: Change lookup failures from LOG_ERROR to debug 1.

	  dns.c and dns_system_resolver.c were spitting out errors for lookup
	  failures for things like not finding a SRV record even though
	  there was an A record.  Those have been changed to debug messages.
	  Logging not finding ANY record is left to the higher level caller.

	  Also, dns_system_resolver was using Windows line endings so I
	  converted them to Unix style.  The actual log changes are on lines
	  156 and 159.

	  Change-Id: I65be16ea15304b96f9dcb4d289dbd3e2286fc094

2015-11-25 10:42 +0000 [270f7be54f]  Alexander Traud <pabstraud@compuserve.com>

	* Build System: Support include-what-you-use.

	  ASTERISK-25591 #close

	  Change-Id: I8d3efa0826142ece9cbed2fd0d46f3b607fee6ae

2015-11-08 23:49 +0000 [f2a84b500d]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue: Show reason of pause on CLI

	  Add value of pause reason when is paused on CLI command "queue show"

	  ASTERISK-25581 #close

	  Report by: Rodrigo Ramírez Norambuena

	  Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864

2015-11-27 07:39 +0000 [7cb8f2f33e]  Niklas Larsson <niklas@tese.se>

	* CHANGES: Fix a typo

	  Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7

2015-11-25 15:26 +0000 [9014f1f4a5]  Kevin Harwell <kharwell@digium.com>

	* fastagi: record file closed after sending result

	  The fastagi record-file testsuite test sometimes fails reporting an empty
	  recorded file. This was happening because Asterisk was sending the agi result
	  notification prior to actually closing the file and the data, being buffered,
	  had not been written to the file yet when the test attempts to check the file
	  size.

	  This patch makes it so the record file stream is closed prior to sending the
	  agi result notification.

	  ASTERISK-25593 #close

	  Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde

2015-11-25 13:29 +0000 [03759c5587]  Walter Doekes <walter+asterisk@wjd.nu>

	* main: Slight refactor of main. Improve color situation.

	  Several issues are addressed here:
	  - main() is large, and half of it is only used if we're not rasterisk;
	    fixed by spliting up the daemon part into a separate function.
	  - Call ast_term_init from rasterisk as well.
	  - Remove duplicate code reading/writing asterisk history file.
	  - Attempt to tackle background color issues and color changes that
	    occur. Tested by starting asterisk -c until the colors stopped
	    changing at odd locations.
	  - Remove unused term_prep() and term_prompt() functions.

	  ASTERISK-25585 #close

	  Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f

2015-11-24 13:54 +0000 [91346b9fb7]  David M. Lee <dlee@respoke.io>

	* Fixed some typos

	  Fixes some minor typos in the CHANGES file, plus an embarrasing typo in
	  the StatsD API.

	  Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7

2015-11-24 13:07 +0000 [fb45130476]  Corey Farrell <git@cfware.com>

	* res_pjsip_notify: Fix CLI usage info

	  The usage info for 'pjsip send notify' previously referenced the
	  chan_sip configuration sip_notify.conf.  Fix this to reference
	  the correct configuration pjsip_notify.conf.

	  ASTERISK-25590 #close

	  Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea

2015-11-18 09:43 +0000 [ee9c114747]  Matt Jordan <mjordan@digium.com>

	* res/res_endpoint_stats: Add module to emit endpoint StatsD statistics

	  This patch adds a module that emits StatsD statistics about Asterisk
	  endpoints. This includes:
	   * A GAUGE statistic for endpoint states, tracking how many endpoints are in
	     a particular state.
	   * A GAUGE statistic for each endpoint, counting the number of channels
	     currently associated with an endpoint.

	  ASTERISK-25572

	  Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305
2015-11-23 14:27 +0000 [9ca652f1b9]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_realtime.c: Fix crash from NULL sorcery object type.

	  If the sorcery object type is not found a NULL is returned.
	  Unfortunately, sorcery_realtime_filter_objectset() will crash after
	  complaining about not finding the object type and saying to expect errors.

	  * Use ao2_cleanup() instead of ao2_ref() to prevent the crash.

	  ASTERISK-25165
	  Reported by Corey Farrell

	  Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97

2015-11-18 10:07 +0000 [75d90a9951]  Matt Jordan <mjordan@digium.com>

	* res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts

	  This patch adds the ability to send StatsD statistics related to the
	  state of PJSIP contacts. This includes:
	   * A GUAGE statistic measuring the count of contacts in a particular state.
	     This measures how many contacts are reachable, unreachable, etc.
	   * The RTT time for each contact, if those contacts are qualified. This
	     provides StatsD engines useful time-based data about each contact.

	  ASTERISK-25571

	  Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c

2015-11-13 10:34 +0000 [482f2fc5ff]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_outbound_registration: Add registration statistics for StatsD

	  This patch adds outbound registration statistics for StatsD. This includes
	  the following:
	   * A GUAGE metric for the overall count of outbound registrations.
	   * A GUAGE metric for each state an outbound registration can be in. As the
	     outbound registrations change state, the overall count of how many
	     outbound registrations are in the particular state is changed.

	  These statistics are particularly useful for systems with a large number of
	  SIP trunks, and where measuring the change in state of the trunks is useful
	  for monitoring.

	  ASTERISK-25571

	  Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37

2015-11-18 10:05 +0000 [97d7b344de]  Matt Jordan <mjordan@digium.com>

	* res_statsd: Add functions that support variable arguments

	  Often, the metric names of statistics we are generating for StatsD have some
	  dynamic component to them. This can be the name of a particular resource, or
	  some internal status label in Asterisk. With the current set of functions,
	  callers of the statsd API must first build the metric name themselves, then
	  pass this to the API functions. This results in a large amount of boilerplate
	  code and usage of either fixed length static buffers or dynamic memory
	  allocation, neither of which is desireable.

	  This patch adds two new functions to the StatsD API that support a printf
	  style format specifier for constructing the metric name. A dynamic string,
	  allocated in threadstorage, is used to build the metric name. This eases
	  the burden on users of the StatsD API.

	  Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea

2015-11-20 21:08 +0000 [726ee873a6]  Matt Jordan <mjordan@digium.com>

	* chan_pjsip: Handle T.38 faxes with direct media bridges

	  When a channel is in a direct media bridge, a re-INVITE may arrive that forces
	  Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge
	  must change its technology to a simple bridge, and re-INVITE the media back
	  to Asterisk.

	  Generally, this logic mostly already exists in Asterisk. However, prior to this
	  patch, there were a few bugs:
	  (1) The T.38 framehook currently prevents a channel capable of T.38 faxes from
	      ever entering into a direct media bridge. This applies even when the only
	      media being passed over the channel is audio. This patch fixes this bug
	      by having the framehook specify that it defers caring about any frame type.
	      This allows the channels to enter into a direct media bridge, which will
	      be broken when a re-INVITE is received.
	  (2) When a re-INVITE is received, nothing instructed the bridging layer to
	      re-inspect the allowed bridging technology. This now occurs when either
	      a re-INVITE is received from a peer, or when a response is received from
	      the far end (that is, when the T.38 state changes to either
	      T38_PEER_REINVITE or T38_LOCAL_REINVITE).
	  (3) chan_pjsip needs to do a small amount of work to prevent a direct media
	      bridge from being chosen when a T.38 session is in progress. When a T.38
	      session supplement has a t38 datastore - which is added when we detect
	      we should start thinking about T.38 on a channel - we now refuse a native
	      RTP bridge.
	  (4) When a BYE request is received, we don't terminate the T.38 session. If
	      the other side of a T.38 fax survives the hangup (due to the 'g' flag
	      in Dial, for example), we don't currently re-INVITE the media on the
	      other channel back to audio. This patch now has res_pjsip_t38 intercept
	      BYE requests and inform the far side that the T.38 session is terminated.
	      This naturally causes the correct re-INVITEs to be sent.

	  ASTERISK-25582

	  Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb

2015-10-22 09:44 +0000 [9315a93757]  Matt Jordan <mjordan@digium.com>

	* main/cli: Use proper string methods to check existence of context/exten/app

	  Because the context, extension, and application are stored in stringfields,
	  checking for them being NULL doesn't work so well. This patch uses the
	  appropriate string library call, ast_strlen_zero, to see if there is a value
	  in the context/exten/app values.

	  Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23

2015-11-20 21:07 +0000 [d2b141c79f]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_t38: Add debug statements

	  This patch adds some debug statements to res_pjsip_t38. These statements help
	  to determine which SDP negotiation callbacks are being executed, and, when
	  a particular callback exits, why a callback may not have applied its logic
	  to the local or remote SDP.

	  Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77

2015-11-19 09:40 +0000 [1bca90fcbe]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_outbound_registration: Apply configuration on object type load

	  When Asterisk is configured to use a dynamic sorcery backend (such as
	  res_sorcery_astdb) with 'registration' objects, it will fail to create the
	  internal state objects associated with the registration objects on module
	  load. This is due to nothing actually querying for the specific objects
	  and calling their sorcery apply handler during module load.

	  This patch fixes that by calling get_registrations in the sorcery observer's
	  object_type_loaded handler. Doing this causes the sorcery backends to be
	  asked for the current state of all registration objects, which causes the
	  apply handler to be called and the internal run-time state to be created.

	  ASTERISK-25575 #close

	  Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23

2015-11-11 06:29 +0000 [8ccb1d2bed]  Alexander Traud <pabstraud@compuserve.com>

	* translate: Provide translation modules the result of SDP negotiation.

	  Previously, a trancoding module did not have access to the joint but cached
	  format. Therefore, the module did not have access to the attributes negotiated
	  via SDP (line fmtp). Now, a translation module receives the joint format.

	  ASTERISK-25545 #close

	  Change-Id: Id6878a989b50573298dab115d3371ea369e1a718

2015-11-19 01:03 +0000 [92ea46ba94]  Alexander Traud <pabstraud@compuserve.com>

	* res_format_attr_h264: Do not reset string buffer.

	  When no parameter is present, Asterisk does not generate the line fmtp, as
	  expected. However, because a buffer was reset, even rtpmap and fmtp of previous
	  media codecs got removed. Now, Asterisk does not reset other codecs in case of
	  no parameter for H.264.

	  ASTERISK-25573 #close

	  Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286

2015-11-18 02:25 +0000 [8c14b91651]  Alec Davis <sivad.a@paradise.net.nz>

	* app_bridgeaddchan: ability to barge into existing call

	  To be able to barge into a call by dialling a prefix+extension that maps
	  to the extensions device.

	  Senario is that DECT headset users may be away from their desks and need
	  to transfer the call, the goal is that from any phone they dial a prefix
	  then their extension and are added to the bridge that they are in, from
	  there they can drop the headset call, as it's also on the handset,
	  and transfer the caller.

	  The dialplan would look like, where prefix=73, extension = 8512;
	  exten => _738512,1,BridgeAdd(SIP/cisco0001)

	  ASTERISK-25551 #close
	  Reported By: Alec Davis

	  Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540

2015-11-05 15:37 +0000 [05addf3d8f]  Tyler Cambron <tcambron@digium.com>

	* StatsD: Add sample rate compatibility

	  Implemented support for the StatsD sample rate parameter,
	  which is a parameter for determining when to send computed
	  statistics to a client.

	  Valid sample rate values are:
	  Less than or equal to 0.0 will never be sent.
	  Between 0.0 and 1.0 will randomly be sent.
	  Greater than or equal to 1.0 will always be sent.

	  ASTERISK-25419
	  Reported By: Ashley Sanders

	  Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52

2015-11-17 14:53 +0000 [3dbaf696e9]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts.

	  Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d

2015-11-17 14:53 +0000 [eaf898ac88]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Fix 423 response handling.

	  Receiving a 423 Interval Too Brief response after authentication for an
	  outbound registration attempt results in assuming that the registrar has
	  rejected the registration permanently.  If there are no configured retries
	  for fatal responses then the outbound registration is stopped for that
	  endpoint.

	  For registrations, PJSIP/PJPROJECT intercepts the handling of 423
	  responses and does not include any authentication in the updated
	  registration request.  When the updated request is challenged then the
	  Asterisk code assumes that we were challenged again because the peer
	  rejected the authentication we sent earlier.

	  * Made registration challenges keep track of the CSeq number to determine
	  if the received challenge response was for the request we thought we sent.
	  If the response's CSeq number differs from the CSeq number we last sent
	  with authentication then authenticate again because it is a challenge to a
	  different request.

	  Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09

2015-11-18 00:20 +0000 [4013f9d577]  Alec Davis <sivad.a@paradise.net.nz>

	* app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked!

	  commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525)
	  refer ASTERISK-24958

	  above commit removed ast_channel_lock(qe->chan);
	  but failed to remove corresponding ast_channel_unlock(qe->chan);

	  ASTERISK-25561 #close
	  Reported Alec Davis

	  Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a

2015-11-16 16:10 +0000 [6919daab61]  gtjoseph <george.joseph@fairview5.com>

	* dns: Fix pointer increment in dns_parse_answer_ex

	  When dns_parse_answer_ex was iterating over the answers it
	  wasn't incrementing the answer pointer correctly after the first
	  answer.  The result was that no answers after the first
	  were being returned.  For results where multiple records should
	  have been sorted by priority, weight, etc., there was nothing
	  to sort so the only the first record was returned even if it
	  wouldn't have been the correct record based on the sort.

	  ASTERISK-25565 #close
	  Reported-by: Daniel Tryba
	  Tested-by George Joseph

	  Change-Id: I8622604fefdcd3c11e2c5609a6382e53b1467b0b

2015-11-13 14:03 +0000 [ed13732188]  Mark Michelson <mmichelson@digium.com>

	* Confbridge: Add a user timeout option

	  This option adds the ability to specify a timeout, in seconds, for a
	  participant in a ConfBridge. When the user's timeout has been reached,
	  the user is ejected from the conference with the CONFBRIDGE_RESULT
	  channel variable set to "TIMEOUT".

	  The rationale for this change is that there have been times where we
	  have seen channels get "stuck" in ConfBridge because a network issue
	  results in a SIP BYE not being received by Asterisk. While these
	  channels can be hung up manually via CLI/AMI/ARI, adding some sort of
	  automatic cleanup of the channels is a nice feature to have.

	  ASTERISK-25549 #close
	  Reported by Mark Michelson

	  Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98

2015-11-16 13:56 +0000 [a83e426e91]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip: Fix off nominal crash with requests that fail and have a timer

	  When a request is sent using pjsip_endpt_send_request and fails, a condition
	  exists where the request wrapper, which is an AO2 object, may be de-ref'd
	  more times than it should. This occurs when the request's callback is called,
	  and, in the callback, the timer on the PJSIP heap is cancelled. When that
	  occurs, the request wrapper's lifetime is decremented. When
	  pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of
	  the request wrapper again, even though we've already cancelled the reference
	  associated with the timer.

	  This patch checks the return result of pj_timer_heap_cancel_if_active before
	  removing the reference associated with the timer. We now only decrement it
	  in this case if a timer is cancelled as a result of the function call.

	  Change-Id: I21332343a1a019c1117076f9bf2df27be2850102

2015-11-14 07:02 +0000 [a1fcf6f7b2]  Joshua Colp <jcolp@digium.com>

	* hashtab: Add NULL check when destroying iterator.

	  The hashtab API is pretty NULL tolerant which has resulted
	  in remaining callers not doing much checks themselves.
	  Unfortunately the function to destroy an iterator does not
	  do a NULL check and will result in a crash if passed NULL.
	  This change fixes that.

	  ASTERISK-25552 #close

	  Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619

2015-11-13 14:32 +0000 [436023a322]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_rfc3326.c: Fix crash when channel goes away.

	  If an authenticated incoming caller does not respond to our 200 OK INVITE
	  response with an ACK then PJSIP will hangup the call.  Unfortunately,
	  there is a chance that the session's channel will go away between one use
	  of the channel pointer and another when building the BYE request because
	  the BYE is being built by the monitor thread and not the call's serializer
	  thread.

	  * Added a check to ensure that the thread trying to add the Reason header
	  is the call's serializer thread.  This ensures that the channel will not
	  go away on us.

	  Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89

2015-11-13 14:19 +0000 [e8881e1770]  Mark Michelson <mmichelson@digium.com>

	* Taskprocessors: Increase high-water mark

	  In practical tests, we have seen certain taskprocessors, specifically
	  Stasis subscription taskprocessors, cross the recently-added high-water
	  mark and emit a warning. This high-water mark warning is only intended
	  to be emitted when things have tanked on the system and things are
	  heading south quickly. In the practical tests, the Stasis taskprocessors
	  sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in
	  any danger at all.

	  As such, this ups the high-water mark to 500 tasks instead. It also
	  redefines the SIP threadpool request denial number to be a multiple of
	  the taskprocessor high-water mark.

	  Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce

2015-11-11 07:00 +0000 [fd23d423d8]  Alexander Traud <pabstraud@compuserve.com>

	* format: Register format-attribute module with cached formats.

	  In Asterisk 13, cached formats are created before their corresponding format-
	  attribute module is registered. Cached formats are involved when a local
	  extension is called. Therefore, ast_format_generate_sdp_fmtp did not work
	  on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264,
	  and format-attribute modules provided externally.

	  ASTERISK-25160 #close

	  Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354

2015-11-12 11:17 +0000 [40b58a5d2b]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip distributor: Don't send 503 response to responses.

	  When the SIP threadpool is backed up with tasks, we send 503 responses
	  to ensure that we don't try to overload ourselves. The problem is that
	  we were not insuring that we were not trying to send a 503 to an
	  incoming SIP response.

	  This change makes it so that we only send the 503 on incoming requests.

	  Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404

2015-11-11 17:11 +0000 [264c74aa22]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Deny requests when threadpool queue is backed up.

	  We have observed situations where the SIP threadpool may become
	  deadlocked. However, because incoming traffic is still arriving, the SIP
	  threadpool's queue can continue to grow, eventually running the system
	  out of memory.

	  This change makes it so that incoming traffic gets rejected with a 503
	  response if the queue is backed up too much.

	  Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816

2015-11-12 06:24 +0000 [a159747660]  Joshua Colp <jcolp@digium.com>

	* format_cap: Don't append the 'none' format when appending all.

	  When appending all formats of a type all the codecs are iterated
	  and added. This operation was incorrectly adding the ast_format_none
	  format which is special in that it is supposed to be used when no
	  format is present. It shouldn't be appended.

	  ASTERISK-25535

	  Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c

2015-11-11 04:16 +0000 [d982b99e71]  Steve Davies <steve@one47.co.uk>

	* Further fixes to improper usage of scheduler

	  When ASTERISK-25449 was closed, a number of scheduler issues mentioned in
	  the comments were missed. These have since beed raised in ASTERISK-25476
	  and elsewhere.

	  This patch attempts to collect all of the scheduler issues discovered so
	  far and address them sensibly.

	  ASTERISK-25476 #close

	  Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b

2015-11-11 11:04 +0000 [2954354404]  Joshua Colp <jcolp@digium.com>

	* threadpool: Handle worker thread transitioning to dead when going active.

	  This change adds handling of dead worker threads when moving them
	  to be active. When this happens the worker thread is removed from
	  both the active and idle threads container. If no threads are able
	  to be moved to active then the pool grows as configured.

	  A unit test has also been added which thrashes the idle timeout
	  and thread activation to exploit any race conditions between the
	  two.

	  ASTERISK-25546 #close

	  Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143

2015-11-10 09:24 +0000 [525c7ab780]  Alexander Traud <pabstraud@compuserve.com>

	* rtp_engine: Init a format-attribute module to its RFC defaults.

	  Previously, format-attribute modules relied on an existing fmtp line in SDP
	  negotiation. However, fmtp is optional for several formats like the Opus Codec.
	  Now, the format-attribute module is called with an empty fmtp, which allows the
	  module to initialise itself to RFC defaults. Furthermore now, Asterisk is able
	  to differentiate between internally and externally created formats.

	  ASTERISK-25537 #close

	  Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52

2015-11-09 18:19 +0000 [be93036a4e]  Corey Farrell <git@cfware.com>

	* Remove ABI compatibility stub functions.

	  ABI compatibility stubs existed for ast_app_separate_args and ast_verbose,
	  this is not needed in master.

	  Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453

2015-11-10 07:51 +0000 [02a124eda5]  Corey Farrell <git@cfware.com>

	* Remove execute permission from dns_system_resolver.c

	  Change-Id: I3185735db42064bab00d3e073aed703385a00bf4

2015-11-09 03:01 +0000 [cf79b62778]  Alexander Traud <pabstraud@compuserve.com>

	* ast_format_cap_get_names: To display all formats, the buffer was increased.

	  ASTERISK-25533 #close

	  Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a

2015-11-09 07:04 +0000 [e85f0c81af]  Alexander Traud <pabstraud@compuserve.com>

	* ast_format_cap: Avoid format creation on module load, use cache instead.

	  Since Asterisk 13, formats are immutable and cached. However while loading a
	  module like chan_sip, some formats were created instead using cached ones.

	  ASTERISK-25535 #close

	  Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b

2015-11-06 07:54 +0000 [7dd8f89a50]  Walter Doekes <walter+asterisk@wjd.nu>

	* func_callerid: Document that CALLERID(pres) is available.

	  CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres)
	  and CALLERID(name-pres).  But for channel driver that don't make a
	  distinction between the two (e.g. SIP), it makes more sense to get/set
	  both at once.  This change reveals the availability of CALLERID(pres),
	  CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and
	  REDIRECTING(from-pres).

	  ASTERISK-25373 #close

	  Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a
2015-11-06 07:52 +0000 [39daf9f066]  Walter Doekes <walter+asterisk@wjd.nu>

	* docs: Fix a few typo's in app docs (more then, resourse).

	  Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7

2015-11-06 14:19 +0000 [d82a4b098f]  gtjoseph <george.joseph@fairview5.com>

	* dns: Use ntohl for ans->ttl in dns_parse_answer_ex

	  dns_parse_answer_ex was not converting ans->ttl from network
	  by order to host byte order which was causing certain ttls
	  it to go negative. In turn this was causing answer edit checks
	  to fail.

	  ASTERISK-25528 #close
	  Reported-by: Daniel Tryba
	  Tested-by: George Joseph

	  Change-Id: I31505132d6321c46d2f39fd06c20ee808a864037

2015-11-06 07:36 +0000 [74e7333317]  Walter Doekes <walter+asterisk@wjd.nu>

	* xmldoc: Improve xmldoc wrapping of 'core show ...' output.

	  Previously, the wrapping did both lookahead and lookback, which,
	  together with color escape sequences, caused some lines to be wrapped
	  way earlier than other lines.  This led to inconsistent output.

	  This simplifies the wrapping code and makes it more sane: if maxcolumns
	  is hit, we simply jump back to the last space and wrap there.

	  ASTERISK-25527 #close

	  Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957

2015-11-06 06:57 +0000 [9d6e917349]  Sean Bright (license #5060)

	* res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.

	  In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual
	  amount of channels is negotiated in-band. Therefore now, the Opus codec and its
	  attribute rtpmap are registered with two channels.

	  ASTERISK-24779 #close
	  Reported by: PowerPBX
	  Tested by: Alexander Traud
	  patches:
	    asterisk-24779.patch submitted by Sean Bright (license #5060)

	  Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b

2015-11-03 16:19 +0000 [a2c2a8e1bb]  Jonathan Rose <jrose@digium.com>

	* taskprocessor: Add high water mark warnings

	  If a taskprocessor's queue grows large, this can indicate that there
	  may be a problem with tasks not leaving the processor or else that
	  the number of available task processors for a given type of task is
	  too low. This patch makes it so that if a taskprocessor's task queue
	  grows above 100 queued tasks that it will emit a warning message.
	  Warning messages are emitted only once per task processor.

	  ASTERISK-25518 #close
	  Reported by: Jonathan Rose

	  Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c

2015-11-02 20:11 +0000 [cd5ae02812]  Corey Farrell <git@cfware.com>

	* Increase account code maximum length to 80.

	  This increases the maximum length of account code's to match
	  extensions.  This ensures it is always possible to set an
	  accountcode to ${EXTEN} without truncation.

	  ASTERISK-23904
	  Reported by: Ben Merrills

	  Change-Id: If122602304ce03362722eb213a3111b32da5eeb9

2015-11-03 14:36 +0000 [379c041038]  Tyler Cambron <tcambron@digium.com>

	* StatsD: Add res_statsd compatibility

	  Added a new api to res_statsd.c to allow it to receive a
	  character pointer for the value argument. This allows for a
	  '+' and a '-' to easily be sent with the value.

	  ASTERISK-25419
	  Reported By: Ashley Sanders

	  Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611

2015-11-04 14:31 +0000 [9c293b5104]  Matt Jordan <mjordan@digium.com>

	* main/dial: Protect access to the format_cap structure of the requesting channel

	  When a dial attempt is made that involves a requesting channel, we previously
	  were not:
	  a) Protecting access to the native format capabilities structure on the
	     requesting channel. That is inherently unsafe.
	  b) Reference bumping the lifetime of the format capabilities structure.

	  In both cases, something else could sneak in, blow away the format
	  capabilities, and we'd be holding onto an invalid format_cap structure. When
	  the newly created channel attempts to construct its format capabilities, things
	  go poorly.

	  This patch:
	  a) Ensures that we get a reference to the native format capabilities while
	     the requesting channel is locked
	  b) Holds a reference to the native format capabilities during the creation
	     of the new channel.

	  ASTERISK-25522 #close

	  Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f

2015-10-30 22:57 +0000 [b0bf189908]  Corey Farrell <git@cfware.com>

	* Fix cli display of build options.

	  A previous commit reduced the AST_BUILDOPTS compiler define to
	  only include options that affected ABI.  This included some options
	  that were previously displayed by cli "core show settings".  This
	  change corrects the CLI display while still restricting buildopts.h
	  to ABI effecting options only.

	  ASTERISK-25434 #close
	  Reported by: Rusty Newton

	  Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325

2015-11-03 10:58 +0000 [63e02b45c6]  Matt Jordan <mjordan@digium.com>

	* pjsip_configuration: On delete, remove the persistent version of an endpoint

	  When an endpoint is deleted (such as through an API), the persistent endpoint
	  currently continues to lurk around. While this isn't harmful from a memory
	  consumption perspective - as all persistent endpoints are reclaimed on
	  shutdown - it does cause Stasis endpoint related operations to continue
	  to believe that the endpoint may or may not exist.

	  This patch causes the persistent endpoint related to a PJSIP endpoint to be
	  destroyed if the PJSIP endpoint is deleted.

	  Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb
2015-11-03 11:15 +0000 [d33a1682e3]  Matt Jordan <mjordan@digium.com>

	* res_pjsip/location: Destroy contact_status objects on contact deletion

	  The contact_status Sorcery objects are currently not destroyed when a contact
	  is deleted. This causes the contact's last known RTT/status to be 'sticky'
	  when the contact itself may no longer exist. This patch causes the
	  contact_status objects associated with both dynamic and static contacts to
	  be destroyed if the AoR holding those contacts is also destroyed (or via
	  other paths where a contact may be deleted.)

	  Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e

2015-11-03 08:15 +0000 [e26a06c1da]  Matt Jordan <mjordan@digium.com>

	* main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field

	  The JSON packing for the ContactStatusChange event forgot to include the
	  roundtrip_usec field. As a result, the field never showed up in any event,
	  even when the data was available. This patch corrects that error by properly
	  packing the JSON blob with the data.

	  Change-Id: I8df80da659a44010afbd48f645967518ff5daa17

2015-11-02 20:24 +0000 [40574a2ea3]  Corey Farrell <git@cfware.com>

	* chan_sip: Allow websockets to be disabled.

	  This patch adds a new setting "websockets_enabled" to sip.conf.
	  Setting this to false allows chan_sip to be used without causing
	  conflicts with res_pjsip_transport_websocket.

	  ASTERISK-24106 #close
	  Reported by: Andrew Nagy

	  Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7

2015-11-02 17:19 +0000 [f80a0ae49b]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Set threadpool max size default to 50.

	  During a stress test of subscriptions, a huge blast of
	  subscription-related traffic resulted in the threadpool expanding to a
	  ridiculous number of threads. The balooning of threads resulted in an
	  increase of memory, which led to a crash due to being out of memory.

	  An easy fix for the particular test was to limit the size of the
	  threadpool, thus reining in the amount of memory that would be used. It
	  was decided that there really is no downside to having a non-infinite
	  default value for the maximum size of the threadpool, so this change
	  introduces 50 threads as the maximum threadpool size for the SIP
	  threadpool.

	  ASTERISK-25513 #close
	  Reported by John Bigelow

	  Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be

2015-10-29 15:25 +0000 [c5093b21ad]  Tyler Cambron <tcambron@digium.com>

	* StatsD: Send stuff to the StatsD server and test

	  Added code to allow the StatsD dialplan application to
	  send data to the server specified in statsd.conf.

	  ASTERISK-25419

	  Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922

2015-11-02 06:57 +0000 [014e3d426b]  Matt Jordan <mjordan@digium.com>

	* pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction

	  When an AoR is created or destroyed dynamically, the scheduled OPTIONS
	  requests that qualify the contacts on the AoR are not necessarily started
	  or destroyed, particularly for persistent contacts created for that AoR.
	  This patch adds create/update/delete sorcery observers for an AoR, which
	  schedule/unschedule the qualifies as expected.

	  Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d

2015-10-30 13:22 +0000 [80cf4960ff]  Matt Jordan <mjordan@digium.com>

	* Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs

	  This patch adds a rule for installing the Super Awesome Company based 'Basic
	  PBX' configuration files. As part of adding this rule, a bit of the content
	  that makes up installing the configuration files under the 'samples' target
	  was refactored into a make subroutine for usage by additional later config
	  make targets.

	  Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404
2015-10-29 08:28 +0000 [b522a5e30f]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_pubsub: Fix assertion when UAS dialog creation fails.

	  When compiled with assertions enabled one will occur when destroying
	  the subscription tree when UAS dialog creation fails. This is because
	  the code assumes that a dialog will always exist on a subscription
	  tree when in reality during this specific scenario it won't.

	  This change makes it so a dialog is not removed from the subscription
	  tree if it is not present.

	  ASTERISK-25505 #close

	  Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee

2015-10-08 11:50 +0000 [fdfd0fb488]  Tyler Cambron <tcambron@digium.com>

	* StatsD: Add user input validation to the application

	  Added code to accept user input and validate it before
	  allowing it to be sent to the StatsD server.

	  ASTERISK-25419
	  Reported By: Ashley Sanders

	  Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3

2015-10-26 11:42 +0000 [d343a25173]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: Do not send all codecs on INVITE.

	  Since version 13, Asterisk sent all allowed codecs as callee, even when the
	  caller did not request/support them. In case of dynamic RTP payloads, this led
	  to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the
	  intersection between the requested and the supported codecs is send again.

	  ASTERISK-24543 #close

	  Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287

2015-10-19 07:11 +0000 [88f3dbaec9]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* install_prereq: Update repositories before install on Debian systems

	  When to install packages the indexed local is more old of the
	  version of software on the repository they have been upgraded by security
	  update then get the package will give 404 not found.

	  The patch prevent by update local index to repository for aptitude before
	  install.

	  ASTERISK-25495 #close

	  Reporte by: Rodrigo Ramírez Norambuena

	  Change-Id: I645959e553aac542805ced394cac2dca964051fa

2015-10-24 13:08 +0000 [4328d320c2]  gtjoseph <george.joseph@fairview5.com>

	* build: GCC 5.1.x catches some new const, array bounds and missing paren issues

	  Fixed 1 issue in each of the affected files.

	  ASTERISK-25494 #close
	  Reported-by: George Joseph
	  Tested-by: George Joseph

	  Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77

2015-10-20 16:02 +0000 [a8aee0bbdb]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Add "like" processing to pjsip list and show commands

	  Add the ability to filter output from pjsip list and show commands
	  using the "like" predicate like chan_sip.

	  For endpoints, aors, auths, registrations, identifyies and transports,
	  the modification was a simple change of an ast_sorcery_retrieve_by_fields
	  call to ast_sorcery_retrieve_by_regex.  For channels and contacts a
	  little more work had to be done because neither of those objects are
	  true sorcery objects.  That was just removing the non-matching object
	  from the final container.  Of course, a little extra plumbing in the
	  common pjsip_cli code was needed to parse the "like" and pass the regex
	  to the get_container callbacks.

	  Some of the get_container code in res_pjsip_endpoint_identifier was also
	  refactored for simplicity.

	  ASTERISK-25477 #close
	  Reported by: Bryant Zimmerman
	  Tested by: George Joseph

	  Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1

2015-10-21 12:22 +0000 [691c0e0b31]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_registration: registration stops due to fatal 4xx response

	  During outbound registration it is possible to receive a fatal (any permanent/
	  non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due
	  to a problem with the registrar itself. Upon receiving the failure response
	  Asterisk terminates outbound registration for the given endpoint.

	  This patch adds an option, 'fatal_retry_interval', that when set continues
	  outbound registration at the given interval up to 'max_retries' upon receiving
	  a fatal response.

	  ASTERISK-25485 #close

	  Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2

2015-10-22 17:07 +0000 [5dd9e1938a]  Mark Michelson <mmichelson@digium.com>

	* format_cap: Detect vector allocation failures.

	  A crash was seen on a system that ran out of memory due to Asterisk not
	  checking for vector allocation failures in format_cap.c. With this
	  change, if either of the AST_VECTOR_INIT calls fail, we will return a
	  value indicating failure.

	  Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8

2015-10-02 15:32 +0000 [7f9823ff57]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog.

	  A certain situation can result in our attempting to send a NOTIFY on a
	  destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but
	  that subscriber has dropped off the network. We end up retransmitting
	  that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY
	  transaction. When the pjsip evsub code is told that the transaction has
	  been terminated, it responds in kind by alerting us that the
	  subscription has been terminated, destroying the subscription, and then
	  removing its reference to the dialog, thus destroying the dialog.

	  The problem is that when we get told that the subscription is being
	  terminated, we detect that we have not sent a terminating NOTIFY
	  request, so we queue up such a NOTIFY to be sent out. By the time that
	  queued NOTIFY gets sent, the dialog has been destroyed, so attempting to
	  send that NOTIFY can result in a crash.

	  The fix being introduced here is actually a reintroduction of something
	  the pubsub code used to employ. We hold a reference to the dialog and
	  wait to decrement our reference to the dialog until our subscription
	  tree object is destroyed. This way, we can send messages on the dialog
	  even if the PJSIP evsub code wants to terminate earlier than we would
	  like.

	  In doing this, some NULL checks for subscription tree dialogs have been
	  removed since NULL dialogs are no longer actually possible.

	  Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5

2015-09-29 14:53 +0000 [e9e4bc9ece]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Ensure dialog lock balance.

	  When sending a NOTIFY, we lock the dialog and then unlock the dialog
	  when finished. A recent change made it so that the subscription tree's
	  dialog pointer will be set NULL when sending the final NOTIFY request
	  out. This means that when we attempt to unlock the dialog, we pass a
	  NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog
	  remains locked after we think we have unlocked it. When a response to
	  the NOTIFY arrives, the monitor thread attempts to lock the dialog, but
	  it cannot because we never released the dialog lock. This results in
	  Asterisk being unable to process incoming SIP traffic any longer.

	  The fix in this patch is to use a local pointer to save off the pointer
	  value of the subscription tree's dialog when locking and unlocking the
	  dialog. This way, if the subscription tree's dialog pointer is NULLed
	  out, the local pointer will still have point to the proper place and the
	  dialog lock will be unlocked as we expect.

	  Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a

2015-09-28 16:36 +0000 [b96267f7a3]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Prevent crashes on final NOTIFY.

	  The SIP dialog is removed from the subscription tree when the final
	  NOTIFY is sent. However, after the final NOTIFY is sent, the persistence
	  update function still attempts to access the cseq from the dialog,
	  resulting in a crash.

	  This fix removes the subscription persistence at the same time that the
	  dialog is removed from the subscription tree. This way, there is no
	  attempt to update persistence when the subscription is being destroyed.

	  Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb

2015-09-17 17:28 +0000 [386cd7b2b0]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Remove serializer when sending final NOTIFY.

	  There have been crashes seen where a taskprocessor's listener is NULL
	  unexpectedly.

	  Looking at backtraces, the problem was specifically seen in PJSIP
	  serializers.

	  Subscriptions make the mistake of removing a serializer from a dialog
	  during subscription tree destruction. Since subscription trees are
	  reference-counted, guaranteeing the circumstances behind the destruction
	  are not possible. This makes it so that the dialog serializer can be
	  removed while not holding the dialog lock. This makes it possible for
	  the distributor to get a pointer to the dialog serializer and have that
	  serializer get freed out from under it.

	  The fix for this is to remove the serializer from a subscription dialog
	  when sending the final NOTIFY. This guarantees that the serializer is
	  removed with the dialog lock held. By doing this, we guarantee that if
	  the distributor gains access to the dialog's serializer, it will not be
	  possible for the serializer to get freed by another thread.

	  Change-Id: I21f5dac33529f65cec45679bdace60670800ff66

2015-09-02 09:14 +0000 [0b63d011c9]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Fix crash on destruction of empty subscription tree.

	  If an old persistent subscription is recreated but then immediately
	  destroyed because it is out of date, the subscription tree will have no
	  leaf subscriptions on it. This was resulting in a crash when attempting
	  to destroy the subscription tree.

	  A simple NULL check fixes this problem.

	  Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac

2015-09-01 15:47 +0000 [ac0194dad6]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Solidify lifetime and ownership of objects.

	  There have been crashes and general instability seen in the pubsub code,
	  so this patch introduces three changes to increase the stability.

	  First, the ownership model for subscriptions has been modified. Due to
	  RLS, subscriptions are stored in memory as a tree structure. Prior to my
	  patch, the PJSIP subscription was the owner of the subscription tree.
	  When the PJSIP subscription told us that it was terminating, we started
	  destroying the subscription tree along with all of the individual leaf
	  subscriptions that belong to the tree. The problem with this model is
	  that the two actors in play here, the PJSIP subscription and the
	  individual leaf subscriptions, need to have joint ownership of the
	  subscription tree. So now, the PJSIP subscription and the individual
	  leaf subscriptions each have a reference to the subscription tree. This
	  way, we will not actually free memory until no players are left that
	  care. The PJSIP subscription is a bigger stakeholder, in that if the
	  PJSIP subscription's reference to the subscription tree is removed, the
	  subscription tree instructs the leaf subscriptions to shut down and drop
	  their references to the subscription tree when possible. The individual
	  leaf subscriptions, upon being told to shut down, can drop their stasis
	  subscriptions or whatever they use to learn of new state, and then drop
	  their reference to the subscription tree once they are ready to die.

	  Second, the lifetime of a PJSIP subscription's reference to our
	  subscription tree has been altered. As I learned from doing a deep dive,
	  the PJSIP evsub code can tell Asterisk multiple times that the
	  subscription has been terminated, and not all of these times
	  are especially helpful. I have altered the message flow that we use for
	  SIP subscriptions such that we will always drop the PJSIP subscription's
	  reference to the subscription tree when we send the NOTIFY that
	  terminates a SIP subscription. This also means that we will now queue
	  NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so
	  that we can have predictable state changes from the PJSIP evsub code.

	  Third, the synchronization of operations has been improved. PJSIP can
	  call into our code from a serializer thread (e.g. upon receiving an
	  incoming request) or from the monitor thread (e.g. when a subscription
	  times out). Because of this, there is the possibility of competing
	  threads stepping on each other. PJSIP attempts to do some
	  synchronization on its own by always keeping the dialog lock held when
	  it calls into us. However, since we end up pushing tasks into the
	  serializer, the result was that serialized operations were not grabbing
	  the dialog lock and could, as a result, step on something that was being
	  attempted by a different thread. Now we ensure that serialized
	  operations grab the dialog lock, then check for extenuating
	  circumstances, then proceed with their operation if they can.

	  Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5

2015-10-19 15:28 +0000 [1ce62b2545]  Richard Mudgett <rmudgett@digium.com>

	* strings.c: Fix __ast_str_helper() to always return a terminated string.

	  Users of functions which call __ast_str_helper() such as the ones listed
	  below are likely to not check the return value for failure so ensuring
	  that the string is always nil terminated is a good safety measure.

	  ast_str_set_va()
	  ast_str_append_va()
	  ast_str_set()
	  ast_str_append()

	  Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07

2015-10-19 15:27 +0000 [a04d946eaa]  Richard Mudgett <rmudgett@digium.com>

	* Add missing failure checks to ast_str_set_va() callers.

	  Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f

2015-10-21 11:44 +0000 [64c172deba]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Move URI validation to use time.

	  In a realtime based system with a limited number of threadpool threads
	  it is possible for a deadlock to occur. This happens when permanent
	  endpoint state is updated, which will cause database queries to be done.
	  These queries may result in URI validation being done which is done
	  synchronously using a PJSIP thread. If all PJSIP threads are in use
	  processing traffic they themselves may be blocked waiting to get the
	  permanent endpoint container lock when identifying an endpoint.

	  This change moves URI validation to occur at use time instead of
	  configuration time. While this comes at a cost of not seeing a problem
	  until you use it it does solve the underlying deadlock problem.

	  ASTERISK-25486 #close

	  Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a

2015-10-21 08:08 +0000 [f9cbac7321]  Alexander Traud <pabstraud@compuserve.com>

	* format: Update the maximum packetization time for iLBC 30.

	  In September 2006, the maximum packetization time (ptime) were set to such a
	  low value, packetization was disabled for many codecs actually. This was fixed
	  for many codecs but not for iLBC 30. This enables packetization for iLBC which
	  can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf.

	  ASTERISK-7803

	  Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12
2015-10-21 09:51 +0000 [f3b2b3d1b3]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: Fix autoframing=yes.

	  With Asterisk 13, the structures ast_format and ast_codec changed. Because of
	  that, the paketization timing (framing) of the RTP channel moved away from the
	  formats/codecs. In the course of that change, the ptime of the callee was not
	  honored anymore, when the optional autoframing was enabled.

	  ASTERISK-25484 #close

	  Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4

2015-10-20 22:24 +0000 [b425850f8b]  Matt Jordan <mjordan@digium.com>

	* rest-api-templates: Wikify error code response reasons

	  Error response code descriptions may contain wiki markup that need to be
	  escaped. Without this patch, Confluence will reject the document being sent
	  and the responsible script will raise an exception.

	  Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5

2015-10-20 12:06 +0000 [7be6194d6f]  Matt Jordan <mjordan@digium.com>

	* funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function

	  When ab803ec342 was committed, it accidentally forgot to actually *add* the
	  HOLD_INTERCEPT function. This highlights two interesting points:
	  * Gerrit forces you to put the patch as it is going to into the repo up for
	    review, which Review Board did not. Yay Gerrit.
	  * No one apparently bothered to use this feature, or else they don't know about
	    it. I'm going to go with the latter explanation.

	  ASTERISK-24922

	  Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396

2015-10-19 14:14 +0000 [77780790e0]  Jonh Wendell <jonh.wendell@gmail.com>

	* main/cdr: Allow modules to modify CDR fields before dispatching them

	  This patch adds the functions

	  	ast_cdr_modifier_register()
	  	ast_cdr_modifier_unregister()

	  That work much like ast_cdr_register() and ast_cdr_unregister().

	  Modules registered will be given a chance to modify (or to do whatever
	  they want) CDR fields just before they are passed to registered engines.

	  Thus, for instance, if a module change the "userfield" field of a CDR,
	  the modified value will be passed to every registered CDR backend for
	  logging.

	  ASTERISK-25479 #close

	  Change-Id: If11d8fd19ef89b1a66ecacf1201e10fcf86ccd56
2015-10-19 19:59 +0000 [b9bd249a85]  Matt Jordan <mjordan@digium.com>

	* contrib/scripts/autosupport: Update for Asterisk 13

	  This patch adds some minor tweaks for autosupport to update it for Asterisk 13.
	  This includes:
	  * Finally removing most references to Zaptel
	  * Adding support for some additional 'core' commands, and fixing nomenclature
	    that generally hasn't been used for some time
	  * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels

	  Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1
	  (cherry picked from commit 9fc9777fa34753fb38991d42d8dbed516e907ca2)

2015-10-18 18:22 +0000 [92fa8d1e0e]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue: Added reason pause of member

	  In app_queue added value Paused Reason on QueueMemberStatus when a member
	  on queue is paused and the reason was set.

	  ASTERISK-25480 #close
	  Reporte by: Rodrigo Ramírez Norambuena

	  Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e

2015-10-16 22:01 +0000 [b19860c03a]  Corey Farrell <git@cfware.com>

	* res_ari_events: Fix memory leak in mustache template.

	  ASTERISK-25308 fixed a memory leak in res_ari_events.c, but
	  this file is regenerated by a template and the template was
	  not fixed.

	  Change-Id: Ied4c6deae89d21f87f9cf99676b1d055aa83b38b

2015-10-14 14:15 +0000 [d799bcf361]  mdu113 <mulitskiy@acedsl.com>

	* res_config_pgsql.c: Fix deadlock loading realtime configuration.

	  On v13, loading several thousand PJSIP endpoints on Asterisk start causes
	  a deadlock most of the time.

	  Thanks to mdu113 for discovering that there was a call to pgsql_exec() not
	  protected by the pgsql_lock reentrancy lock.

	  {quote}
	  I believe a code path exists that attempts to use pgsql connection without
	  locking pgsql_lock.  I believe what happens during that deadlock that I
	  see is two concurrent threads are both attempting to send query to pgsql,
	  one of the thread is using a code path without locking pgsql_lock.  If
	  they managed to send queries at the same time, it seems postgres ignores
	  one of the queries and replies only to the one of them.  If it happens so
	  that the thread holding the lock didn't receive the reply it will wait for
	  it (and hold the lock) forever (or at least for very long time), thus
	  completely blocking all access to db.
	  {quote}

	  * Added missing reentrancy locking around pgsql_exec() in find_table().

	  * Moved unlock of pgsql_lock in unload_module() to avoid locking inversion
	  between the psql_tables list lock and the pgsql_lock.

	  ASTERISK-25455 #close
	  Reported by:  mdu113
	  Patches:
	        res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113

	  Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2

2015-10-13 14:13 +0000 [13229037d1]  Olle Johansson (License 5267)

	* channels/chan_sip: Set cause code to 44 on RTP timeout

	  To quote Olle:

	  "When issuing a hangup due to RTP timeouts the cause code is not set. I have
	  selected 44 based on Cisco's implementation..."

	  ASTERISK-25135 #close
	  Reported by: Olle Johansson
	  patches:
	    rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267)

	  Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc

2015-10-12 11:21 +0000 [984f100dab]  Richard Mudgett <rmudgett@digium.com>

	* config.c: Fix off-nominal memory leak.

	  Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0

2015-10-12 11:20 +0000 [9951255775]  Richard Mudgett <rmudgett@digium.com>

	* config.c: Fix potential memory corruption after [section](+).

	  The memory corruption could happen if the [section](+) is the last section
	  in the file with trailing comments.  In this case process_text_line() has
	  left *last_cat is set to newcat and newcat is destroyed.

	  Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93

2015-10-12 11:21 +0000 [c1ed11ee31]  Richard Mudgett <rmudgett@digium.com>

	* config.c: Fix #include after [section](+).

	  An #include right after a [section](+) would associate any variable
	  assignments before a new section in the #include with the wrong section.

	  * Fix section association by setting the current section to the appended
	  section.

	  * Fix '+' and '!' section flag interaction corner case depending upon
	  which flag came first.  If the '!' came first then it would be ignored.
	  If the '!' came after then it would affect the appended section.  The '!'
	  will now no longer be ignored.

	  ASTERISK-25461 #close
	  Reported by: Sean Pimental

	  Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3

2015-10-10 15:20 +0000 [a12eb89ea4]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* Build: Add menuselect options for using compiler sanitizers

	  This patch adds menuselect options for building Asterisk with
	  various sanitizers provided by gcc and clang.

	  When one of *SANITIZER flags is set in menuselect, the appropriate
	  option is added to CFLAGS ad LDFLAGS for the build.

	  Information on sanitizers in the project wiki:
	  https://github.com/google/sanitizers/wiki

	  GCC Manual:
	  https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html

	  Clang Compiler User's Manual:
	  http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation

	  ASTERISK-24718 #close
	  Reported by: Badalian Vyacheslav

	  Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0

2015-10-08 16:43 +0000 [ca030845ff]  Richard Mudgett <rmudgett@digium.com>

	* configure: Fix check for libunbound to require v1.5.0 as minimum.

	  Versions of libunbound before v1.4.21 do not compile with Asterisk.
	  However, since v1.4.21 has a configure script bug that fails to detect the
	  ldns library (which is fixed in v1.4.22) and v1.4.22 is not an easily
	  detectable version we will require v1.5.0 as a minimum version of the
	  library to work with Asterisk.

	  ASTERISK-25108 #close
	  Reported by: Richard Mudgett

	  Change-Id: Ieb228bfb01467573fc121c7356a9dde27128894d

2015-10-08 11:50 +0000 [2fe9f09705]  Tyler Cambron <tcambron@digium.com>

	* StatsD: Write skeleton Asterisk application

	  Wrote the skeleton framework for the Asterisk StatsD dialplan
	  application. This includes a load function, unload function, a
	  callback for execution, and XML documentation.

	  ASTERISK-25419
	  Reported By: Ashley Sanders

	  Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363

2015-10-06 18:01 +0000 [34d7fa6c4a]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Fix deadlock when sending out-of-dialog requests.

	  The struct send_request_wrapper has a pjsip lock associated with it that
	  is created non-recursive.  There is a code path for the struct
	  send_request_wrapper lock that will attempt to lock it recursively.  The
	  reporter's deadlock showed that the thread calling endpt_send_request()
	  deadlocked itself right after the wrapper object got created.

	  Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited
	  MWI NOTIFY messages can hit this deadlock.

	  * Replaced the struct send_request_wrapper pjsip lock with the mutex lock
	  that can come with an ao2 object since all of Asterisk's mutexes are
	  recursive.  Benefits include removal of code maintaining the pjsip
	  non-recursive lock since ao2 objects already know how to maintain their
	  own lock and the lock will show up in the CLI "core show locks" output.

	  ASTERISK-25435 #close
	  Reported by: Dmitriy Serov

	  Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d

2015-10-06 11:05 +0000 [cc131832aa]  Stefan Engström <stefanen@kth.se>

	* res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending

	  In ast_rtp_read, the value of the variable 'mark' which we try to assign to a
	  frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate
	  it to 0 or 1.

	  ASTERISK-25451 #close
	  Change-Id: I53bdf5c026041730184a6a809009c028549ce626

2015-10-07 01:24 +0000 [c944263e36]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* func_presencestate: Return "not_set" when no data is set in AstDB

	  Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not
	  exist, i.e. when a new CustomPresence is added in the dialplan.

	  ASTERISK-25400 #close
	  Reported by: Andrew Nagy

	  Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a

2015-10-06 20:43 +0000 [4bf395e81e]  Matt Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk: Fix assignment after ao2 decrement

	  When we decide we will no longer schedule an RTCP write, we remove the
	  reference to the RTP instance, then assign -1 to the stored scheduler ID
	  in case something else comes along and wants to see if anything is scheduled.

	  That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to
	  fix the regression introduced by 3cf0f29310, this improper assignment on a
	  potentially destroyed object started getting tripped on the build agents.

	  Frankly, this should have been crashing a lot more often earlier. I can only
	  assume that the timing was changed just enough by both changes to start
	  actually hitting this problem.

	  As it is, simply moving the assignment prior to the ao2 deference is sufficient
	  to keep the RTP instance from being referenced when it is very, truly,
	  aboslutely dead.

	  (Note that it is still good practice to assign -1 to the scheduler ID when we
	  know we won't be scheduling it again, as the ao2 deref *may* not always destroy
	  the ao2 object.)

	  ASTERISK-25449

	  Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7

2015-10-06 12:40 +0000 [3ec9cf7d6a]  Florian Sauerteig <ffs@ccn.net>

	* chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers.

	  If a Via header containes an IPv6 address and a port number is ommitted,
	  as it is the standard port, we now leave the port empty and to not set it
	  to the value after the first colon of the IPv6 address.

	  ASTERISK-25443 #close

	  Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70

2015-10-05 16:53 +0000 [8fe9350b68]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip: Fix crash on reINVITE before initial INVITE completes.

	  Apparently some endpoints attempt to send a reINVITE before completing the
	  initial INVITE transaction.  In this case PJSIP responds appropriately to
	  the reINVITE with a 491 INVITE request pending.  Unfortunately chan_pjsip
	  is using the initial INVITE transaction state to determine if an INVITE is
	  the initial INVITE or a reINVITE.  Since the initial INVITE transaction
	  has not been confirmed yet chan_pjsip thinks the reINVITE is an initial
	  INVITE and starts another PBX thread on the channel.  The extra PBX thread
	  ensures that hilarity ensues.

	  * Fix checks for a reINVITE on incoming requests to look for the presence
	  of a to-tag instead of the initial INVITE transaction state.

	  * Made caller_id_incoming_request() determine what to do if there is a
	  channel on the session or not.  After a channel is created it is too late
	  to just store the new party id on the session because the session's party
	  id has already been copied to the channel's caller id.

	  ASTERISK-25404 #close
	  Reported by: Chet Stevens

	  Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be

2015-10-05 21:34 +0000 [8cb614fe20]  Matt Jordan <mjordan@digium.com>

	* Fix improper usage of scheduler exposed by 5c713fdf18f

	  When 5c713fdf18f was merged, it allowed for scheduled items to have an ID of
	  '0' returned. While this was valid per the documentation for the API, it was
	  apparently never returned previously. As a result, several users of the
	  scheduler API viewed the result as being invalid, causing them to reschedule
	  already scheduled items or otherwise fail in interesting ways.

	  This patch corrects the users such that they view '0' as valid, and a returned
	  ID of -1 as being invalid.

	  Note that the failing HEP RTCP tests now pass with this patch. These tests
	  failed due to a duplicate scheduling of the RTCP transmissions.

	  ASTERISK-25449 #close

	  Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39
2015-08-26 16:58 +0000 [c6b0d60264]  Debian Amtelco <dan@amtelco.com>

	* chan_pjsip: Add Referred-By header to the PJSIP REFER packet.

	  Some systems require the REFER packet to include a Referred-By header.
	  If the channel variable SIPREFERREDBYHDR is set, it passes that value as the
	  Referred-By header value.  Otherwise, it adds the current dialog’s local info.

	  Reported by: Dan Cropp
	  Tested by: Dan Cropp

	  Change-Id: I3d17912ce548667edf53cb549e88a25475eda245

2015-10-03 06:27 +0000 [89dec7675d]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* manager: Fix GetConfigJSON returning invalid JSON

	  When GetConfigJSON was introduced back in 1.6, it returned each
	  section as an array of strings: ["key=value", "key2=value2"].
	  Afterwards, it was changed a few times and became
	  ["key": "value", "key2": "value2"], which is not a correct JSON.
	  This patch fixes that by constructing a JSON object {} instead of
	  an array [].

	  Also, the keys "istemplate" and "tempates" that are used to
	  indicate templates and their inherited categories are now wrapped in
	  quotes.

	  ASTERISK-25391 #close
	  Reported by: Bojan Nemčić

	  Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8

2015-09-30 17:28 +0000 [1b80dbeb60]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_memory_cache.c: Fix deadlock with scheduler.

	  A deadlock can happen when a sorcery object is being expired from the
	  memory cache when at the same time another object is being placed into the
	  memory cache.  There are a couple other variations on this theme that
	  could cause the deadlock.  Basically if an object is being expired from
	  the sorcery memory cache at the same time as another thread tries to
	  update the next object expiration timer the deadlock can happen.

	  * Add a deadlock avoidance loop in expire_objects_from_cache() to check if
	  someone is trying to remove the scheduler callback from the scheduler.

	  ASTERISK-25441 #close

	  Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc

2015-10-01 14:30 +0000 [9c1ca287a4]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_memory_cache.c: Replace inline code with function.

	  Make sorcery_memory_cache_close() call remove_all_from_cache() instead of
	  partially inlining it.

	  ASTERISK-25441

	  Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c

2015-10-01 14:27 +0000 [6554a3b25e]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_memory_cache.c: Shutdown in a less crash potential order.

	  Basically you should shutdown in the opposite order of how you setup since
	  later setup pieces likely depend on earlier setup pieces.  e.g.,
	  Registering your external API with the rest of the system should be the
	  last thing setup and the first thing unregistered during shutdown.

	  Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e

2015-09-30 17:27 +0000 [359394cc29]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_memory_cache.c: Misc tweaks.

	  Change-Id: I8cd32dffbb4f33bb0c39518d6e4c991e73573160

2015-09-30 17:27 +0000 [7942d1c2ff]  Richard Mudgett <rmudgett@digium.com>

	* res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK.

	  Change-Id: Ibca6574dc3c213b29cc93486e01ccd51f5caa46c

2015-09-30 13:42 +0000 [9f229d6a49]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Move "Set role" warning to be debug.

	  In practice the set_role API callback can be invoked even
	  when no ICE is present on an RTP instance. This can occur
	  if ICE has not been enabled on it.

	  ASTERISK-25438 #close

	  Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69

2015-09-28 15:31 +0000 [9bc7386b7c]  Richard Mudgett <rmudgett@digium.com>

	* sched.c: Add warning about negative time interval request.

	  Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc

2015-09-25 18:37 +0000 [12feec0bf7]  Richard Mudgett <rmudgett@digium.com>

	* res/ari/config.c: Fix user sort compare function.

	  Made use the ao2 sort compare template function and OBJ_SEARCH_xxx
	  identifiers.

	  Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c

2015-09-25 17:26 +0000 [3f4fa245e5]  Richard Mudgett <rmudgett@digium.com>

	* res/ari/config.c: Optimize conf_alloc() object init.

	  * Now conf_alloc() has more off nominal error checking.

	  * Eliminated RAII_VAR() use in conf_alloc().

	  * Eliminated a dubius shortcut when destroying cfg->general in
	  conf_destructor() that would cause a crash if cfg->general failed to get
	  allocated.

	  * Add some ACO registration section comments.

	  Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a

2015-09-25 16:48 +0000 [aa00df62ee]  Richard Mudgett <rmudgett@digium.com>

	* res/ari/config.c: Fix conf_alloc() object init.

	  Need to finish initializing the string fields in the ao2 object before
	  putting any default strings into them.

	  ASTERISK-25383 #close
	  Reported by:  yaron nahum

	  Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84

2015-09-21 07:26 +0000 [2d7a4a3357]  Matt Jordan <mjordan@digium.com>

	* main/logger: Add log formatters and JSON structured logs

	  When Asterisk is part of a larger distributed system, log files are often
	  gathered using tools (such as logstash) that prefer to consume information
	  and have it rendered using other tools (such as Kibana) that prefer a
	  structured format, e.g., JSON. This patch adds support for JSON formatted
	  logs by adding support for an optional log format specifier in Asterisk's
	  logging subsystem. By adding a format specifier of '[json]':

	  full => [json]debug,verbose,notice,warning,error

	  Log messages will be output to the 'full' channel in the following
	  format:

	  {
	    "hostname": Hostname or name specified in asterisk.conf
	    "timestamp": Date/Time
	    "identifiers": {
	      "lwp": Thread ID,
	      "callid": Call Identifier
	    }
	    "logmsg": {
	      "location": {
	        "filename": Name of the file that generated the log statement
	        "function": Function that generated the log statement
	        "line": Line number that called the logging function
	      }
	      "level": Log level, e.g., DEBUG, VERBOSE, etc.
	      "message": Actual text of the log message
	    }
	  }

	  ASTERISK-25425 #close

	  Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238

2015-09-27 20:45 +0000 [9402f80726]  Matt Jordan <mjordan@digium.com>

	* res/res_stasis: Fix accidental subscription to 'all' bridge topic

	  When b99a7052621700a1aa641a1c24308f5873275fc8 was merged, subscribing to a
	  NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to
	  all bridges. Unfortunately, the res_stasis control loop did not check that
	  a bridge changing on a channel's control object was actually also non-NULL.
	  As a result, app_subscribe_bridge will be called with a NULL bridge when a
	  channel leaves a bridge. This causes a new subscription to be made to the
	  bridge. If an application has also subscribed to the bridge, the application
	  will now have two subscriptions:
	  (1) The explicit one created by the app
	  (2) The implicit one accidentally created by the control structure

	  As a result, the 'BridgeDestroyed' event can be sent multiple times. This
	  patch corrects the control loop such that it only subscribes an application
	  to a new bridge if the bridge pointer is non-NULL.

	  ASTERISK-24870

	  Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f

2015-09-04 13:51 +0000 [d6472d96b3]  Scott Griepentrog <scott@griepentrog.com>

	* Scripts: check file versions of Asterisk and dependencies

	  To help in diagnosing mismatched modules and libraries, this
	  script scans for version, repository, and source information
	  and reports what is found.

	  ASTERISK-25376 #close
	  Reported by: Ashley Sanders

	  Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6

2015-09-24 14:56 +0000 [7c7a7ddd27]  Richard Mudgett <rmudgett@digium.com>

	* app_queue.c: Force COLP update if outgoing channel name changed.

	  * When a call is answered and the outgoing channel name has changed then
	  force a connected line update because the channel is no longer the same.
	  The channel was masqueraded into by another channel.  This is usually
	  because of a call pickup.

	  Note: Forwarded calls are handled in a controlled manner so the original
	  channel name is replaced with the forwarded channel.

	  ASTERISK-25423 #close
	  Reported by: John Hardin

	  Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172

2015-09-24 14:20 +0000 [145608bd81]  Richard Mudgett <rmudgett@digium.com>

	* app_queue.c: Factor out a connected line update routine.

	  Replace inlined code with update_connected_line_from_peer().

	  ASTERISK-25423
	  Reported by: John Hardin

	  Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3

2015-09-24 13:27 +0000 [1d394774b2]  Richard Mudgett <rmudgett@digium.com>

	* app_dial.c: Make 'A' option pass COLP updates.

	  While the 'A' option is playing the announcement file allow the caller and
	  peer to exchange COLP update frames.

	  ASTERISK-25423
	  Reported by: John Hardin

	  Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9

2015-09-24 12:59 +0000 [680b76eb25]  Richard Mudgett <rmudgett@digium.com>

	* app_dial.c: Force COLP update if outgoing channel name changed.

	  * When a call is answered and the outgoing channel name has changed then
	  force a connected line update because the channel is no longer the same.
	  The channel was masqueraded into by another channel.  This is usually
	  because of a call pickup.

	  Note: Forwarded calls are handled in a controlled manner so the original
	  channel name is replaced with the forwarded channel.

	  ASTERISK-25423
	  Reported by: John Hardin

	  Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c

2015-09-24 12:37 +0000 [fdf0bcb04a]  Richard Mudgett <rmudgett@digium.com>

	* app_dial.c: Factor out a connected line update routine.

	  Replace inlined code with update_connected_line_from_peer().

	  ASTERISK-25423
	  Reported by: John Hardin

	  Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091

2015-09-23 17:41 +0000 [c285879845]  Richard Mudgett <rmudgett@digium.com>

	* app_dial.c: Remove some no-op code.

	  Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54

2015-09-23 14:02 +0000 [3eefa07a39]  Mark Michelson <mmichelson@digium.com>

	* logger: Prevent duplicate dynamic channels from being added.

	  There was a problem observed where the "logger add channel" CLI command
	  would allow for a channel with the same name to be added multiple times.
	  This would result in each message being written out to the same file
	  multiple times.

	  The problem was due to the difference in how logger channel filenames
	  are stored versus the format they are allowed to be presented when they
	  are added. For instance, if adding the logger channel "foo" through the
	  CLI, the result would be a logger channel with the file name
	  /var/log/asterisk/foo being stored. So when trying to add another "foo"
	  channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily
	  add the duplicate channel.

	  The fix presented here is to introduce two new methods in the logger
	  code:
	   * make_filename(): given a logger channel name, this creates the
	     filename for that logger channel.
	   * find_logchannel(): given a logger channel name, this calls
	     make_filename() and then traverses the list of logchannels in order
	     to find a match.

	  This change has made use of make_filename() and find_logchannel()
	  throughout to more consistently behave.

	  ASTERISK-25305 #close
	  Reported by Mark Michelson

	  Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36

2015-09-24 14:49 +0000 [f42084be09]  Mark Michelson <mmichelson@digium.com>

	* Do not swallow frames on channels leaving bridges.

	  When leaving a bridge, indications on a channel could be swallowed by
	  the internal indication logic because it appears that the channel is on
	  its way to be hung up anyway. One such situation where this is
	  detrimental is when channels on hold are redirected out of a bridge. The
	  AST_CONTROL_UNHOLD indication from the bridging code is swallowed,
	  leaving the channel in question to still appear to be on hold.

	  The fix here is to modify the logic inside ast_indicate_data() to not
	  drop the indication if the channel is simply leaving a bridge. This way,
	  channels on hold redirected out of a bridge revert to their expected "in
	  use" state after the redirection.

	  ASTERISK-25418 #close
	  Reported by Mark Michelson

	  Change-Id: If6115204dfa0551c050974ee138fabd15f978949

2015-09-22 17:08 +0000 [06f4f80a63]  Richard Mudgett <rmudgett@digium.com>

	* app_page.c: Fix crash when forwarding with a predial handler.

	  Page uses the async method of dialing with the dial API.  When a call gets
	  forwarded there is no calling channel available.  If the predial handler
	  was set then the calling channel could not be put into auto-service
	  for the forwarded call because it doesn't exist.  A crash is the result.

	  * Moved the callee predial parameter string processing to before the
	  string is passed to the dial API rather than having the dial API do it.
	  There are a few benefits do doing this.  The first is the predial
	  parameter string processing doesn't need to be done for each channel
	  called by the dial API.  The second is in async mode and the forwarded
	  channel is to have the predial handler executed on it then the
	  non-existent calling channel does not need to be present to process the
	  predial parameter string.

	  * Don't start auto-service on a non-existent calling channel to execute
	  the predial handler when the dial API is in async mode and forwarding a
	  call.

	  ASTERISK-25384 #close
	  Reported by: Chet Stevens

	  Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981

2015-09-04 12:25 +0000 [b99a705262]  Matt Jordan <mjordan@digium.com>

	* ARI: Add the ability to subscribe to all events

	  This patch adds the ability to subscribe to all events. There are two possible
	  ways to accomplish this:
	  (1) On initial WebSocket connection. This patch adds a new query parameter,
	      'subscribeAll'. If present and True, Asterisk will subscribe the
	      applications to all ARI events.
	  (2) Via the applications resource. When subscribing in this manner, an ARI
	      client should merely specify a blank resource name, i.e., 'channels:'
	      instead of 'channels:12354'. This will subscribe the application to all
	      resources of the 'channels' type.

	  ASTERISK-24870 #close

	  Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6

2015-09-21 18:06 +0000 [c74101509d]  Kevin Harwell <kharwell@digium.com>

	* app_record: RECORDED_FILE variable not being populated

	  The RECORDED_FILE variable is empty unless a '%d' is specified in the filename.
	  This patch makes it so the variable is always set to the filename.

	  ASTERISK-25410 #close

	  Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653

2015-09-21 08:16 +0000 [a29cf45c76]  Elazar Broad <elazar@thebroadfamily.com>

	* core/logging: Fix logging to more than one syslog channel

	  Currently, Asterisk will log to the last configured syslog
	  channel in logger.conf. This is due to the fact that the
	  final call to openlog() supersedes all of the previous calls.
	  This commit removes the call to openlog() and passes the
	  facility to ast_log_vsyslog(), along with utilizing the
	  LOG_MAKEPRI macro to ensure that the message is routed to
	  the correct facility and with the correct priority.

	  ASTERISK-25407 #close
	  Reported by: Elazar Broad
	  Tested by: Elazar Broad

	  Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2
2015-09-04 12:24 +0000 [47813cc51c]  Matt Jordan <mjordan@digium.com>

	* res/res_stasis_device_state: Allow for subscribing to 'all' device state

	  This patch adds support for subscribing to all device state changes. This is
	  done either by subscribing to an empty device, e.g., 'eventSource=deviceState:',
	  or by the WebSocket connection specifying that it wants all state in the
	  system.

	  ASTERISK-24870

	  Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32

2015-09-03 21:19 +0000 [5206aa9d30]  Matt Jordan <mjordan@digium.com>

	* ARI: Add events for Contact and Peer Status changes

	  This patch adds support for receiving events regarding Peer status changes
	  and Contact status changes. This is particularly useful in scenarios where
	  we are subscribed to all endpoints and channels, where we often want to know
	  more about the state of channel technology specific items than a single
	  endpoint's state.

	  ASTERISK-24870

	  Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9

2015-09-19 12:49 +0000 [9200ad03a3]  Alexander Traud <pabstraud@compuserve.com>

	* astfd: Adds a timestamp for each entry.

	  Now with menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", a timestamp is
	  shown with each file descriptor. This helps to debug leaked UDP/TCP ports on
	  long-lived servers, for example.

	  ASTERISK-25405 #close

	  Change-Id: I968339e5155a512eba1032a5263f1ec8b5e1f80b

2015-09-16 08:22 +0000 [42a897c4c3]  Joshua Colp <jcolp@digium.com>

	* pbx: Update device and presence state when changing a hint extension.

	  When changing a hint extension without removing the hint first the
	  device state and presence state is not updated. This causes the state
	  of the hint to be that of the previous extension and not the current
	  one. This state is kept until a state change occurs as a result of
	  something (presence state change, device state change).

	  This change updates the hint with the current device and presence
	  state of the new extension when it is changed. Any state callbacks
	  which may have been added before the hint extension is changed are
	  also informed of the new device and presence state if either have
	  changed.

	  ASTERISK-25394 #close

	  Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f

2015-09-17 16:34 +0000 [d9723d242a]  Scott Griepentrog <scott@griepentrog.com>

	* CHAOS: avoid crash if string create fails

	  Validate string buffer allocation before using them.

	  ASTERISK-25323

	  Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0

2015-09-11 01:52 +0000 [99aa7cb26e]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* dr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUF

	  Remove repeated code on macro of assigned buffer to SQL vars

	  Change-Id: Icb19ad013124498e172ea1d0b29ccd0ed17deef0

2015-09-17 04:52 +0000 [e4df271a3e]  Walter Doekes <walter+asterisk@wjd.nu>

	* chan_sip: Fix From header truncation for extremely long CALLERID(name).

	  The CALLERID(num) and CALLERID(name) and other info are placed into the
	  `char from[256]` in initreqprep. If the name was too long, the addr-spec
	  and params wouldn't fit.

	  Code is moved around so the addr-spec with params is placed there first,
	  and then fitting in as much of the display-name as possible.

	  ASTERISK-25396 #close

	  Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260

2015-09-17 16:59 +0000 [e1927915bc]  Richard Mudgett <rmudgett@digium.com>

	* CHAOS: res_pjsip_diversion avoid crash if allocation fails

	  Validate ast_malloc buffer returned before using it in
	  set_redirecting_value().

	  ASTERISK-25323

	  Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253

2015-09-17 16:47 +0000 [729a4325da]  Kevin Harwell <kharwell@digium.com>

	* app_queue: AgentComplete event has wrong reason

	  When a queued caller transfers an agent to another extension sometimes the
	  raised AgentComplete event has a reason of "caller" and sometimes "transfer".
	  Since a transfer has taken place this should always be transfer. This occurs
	  because sometimes the stasis hangup event arrives before the transfer event
	  thus writing a different reason out.

	  With this patch, when a hangup event is received during a transfer it will
	  check to see if the channel that is hanging up is part of a transfer. If so
	  it will return and let the subsequently received transfer event handler take
	  care of the cleanup.

	  ASTERISK-25399 #close

	  Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d

2015-09-17 13:09 +0000 [87f04d5acf]  Scott Griepentrog <scott@griepentrog.com>

	* PJSIP: avoid crash when getting rtp peer

	  Although unlikely, if the tech private is returned as
	  a NULL, chan_pjsip_get_rtp_peer() would crash.

	  ASTERISK-25323

	  Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a

2015-09-17 11:31 +0000 [63ede41227]  Kevin Harwell <kharwell@digium.com>

	* app_queue: Crash when transferring

	  During some transfer scenarios involving queues Asterisk would sometimes
	  crash when trying to obtain a channel snapshot (could happen on caller or
	  member channels). This occurred because the underlying channel had already
	  disappeared when trying to obtain the latest snapshot.

	  This patch adds a reference to both the member and caller channels that
	  extends to the lifetime of the queue'd call, thus making sure the channels
	  will always exist when retrieving the latest snapshots.

	  ASTERISK-25185 #close
	  Reported by: Etienne Lessard

	  Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128

2015-09-16 17:36 +0000 [e47396721f]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Eliminate race during initial NOTIFY.

	  There is a slim chance of a race condition occurring where two threads
	  can both attempt to manipulate the same area.

	  Thread A can be handling an incoming initial SUBSCRIBE request. Thread A
	  lets the specific subscription handler know that the subscription has
	  been established.

	  At this point, Thread B may detect a state change on the subscribed
	  resource and queue up a notification task on Thread C, the subscription
	  serializer thread.

	  Now Thread A attempts to generate the initial NOTIFY request to send to
	  the subscriber at the same time that Thread C attempts to generate a
	  state change NOTIFY request to send to the subscriber.

	  The result is that Threads A and C can step on the same memory area,
	  resulting in a crash. The crash has been observed as happening when
	  attempting to allocate more space to hold the body for the NOTIFY.

	  The solution presented here is to queue the subscription establishment
	  and initial NOTIFY generation onto the subscription serializer thread
	  (Thread C in the above scenario). This way, there is no way that a state
	  change notification can occur before the initial NOTIFY is sent, and if
	  there is a quick succession of NOTIFYs, we can guarantee that the two
	  NOTIFY requests will be sent in succession.

	  Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815

2015-08-28 15:42 +0000 [077adf48b8]  Alexander Traud <pabstraud@compuserve.com>

	* translate: Fix transcoding while different in frame size.

	  When Asterisk translates between codecs, each with a different frame size (for
	  example between iLBC 30 and Speex-WB), too large frames were created by
	  ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame
	  length, creating several frames when necessary. Affects all transcoding modules
	  which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex.

	  ASTERISK-25353 #close

	  Change-Id: I2e229569d73191d66a4e43fef35432db24000212

2015-09-10 17:19 +0000 [0a74c80300]  Mark Michelson <mmichelson@digium.com>

	* scheduler: Use queue for allocating sched IDs.

	  It has been observed that on long-running busy systems, a scheduler
	  context can eventually hit INT_MAX for its assigned IDs and end up
	  overflowing into a very low negative number. When this occurs, this can
	  result in odd behaviors, because a negative return is interpreted by
	  callers as being a failure. However, the item actually was successfully
	  scheduled. The result may be that a freed item remains in the scheduler,
	  resulting in a crash at some point in the future.

	  The scheduler can overflow because every time that an item is added to
	  the scheduler, a counter is bumped and that counter's current value is
	  assigned as the new item's ID.

	  This patch introduces a new method for assigning scheduler IDs. Instead
	  of assigning from a counter, a queue of available IDs is maintained.
	  When assigning a new ID, an ID is pulled from the queue. When a
	  scheduler item is released, its ID is pushed back onto the queue. This
	  way, IDs may be reused when they become available, and the growth of ID
	  numbers is directly related to concurrent activity within a scheduler
	  context rather than the uptime of the system.

	  Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2

2015-09-03 21:15 +0000 [45cf79665c]  Matt Jordan <mjordan@digium.com>

	* main/config_options: Check for existance of internal object before derefing

	  Asterisk can load and register an object type while still having an invalid
	  sorcery mapping. This can cause an issue when a creation call is invoked.
	  For example, mis-configuring PJSIP's endpoint identifier by IP address mapping
	  in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a
	  subsequent ARI invocation to create the object will cause a crash, as the
	  internal type may not be registered as sorcery expects.

	  Merely checking for a NULL pointer here solves the issue.

	  Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac

2015-08-21 21:50 +0000 [34aa96bef4]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* chan_sip.c: Validation on module reload

	  Change validation on reload module because now used the cli function for
	  reload. The sip_reload() function never fail and ever return NULL for this
	  reason on reload() now use the call the sip_reload() and return
	  AST_MODULE_LOAD_SUCCESS.

	  This problem is dectected on reload by PUT method on ARI, getting always
	  404 http code when the module is reloaded.

	  ASTERISK-25325 #close
	  Reporte by: Rodrigo Ramírez Norambuena

	  Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb

2015-08-21 17:39 +0000 [69824fdfbf]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.

	  Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093

2015-09-09 12:24 +0000 [2526659432]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Add some notification comments.

	  Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20

2015-08-21 18:01 +0000 [9b290dfe2f]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.

	  We should not try to send a SIP response message because we may be
	  restoring a persistent subscription where we are not responding to a SIP
	  request.

	  Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec

2015-08-21 17:40 +0000 [73eb132012]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Fix off-nominal memory leak.

	  Fix off-nominal visited vector leak in build_resource_tree().

	  Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c

2015-08-21 15:26 +0000 [2b30fc2b2d]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Fix one byte buffer overrun error.

	  ast_sip_pubsub_register_body_generator() did not account for the null
	  terminator set by sprintf() in the allocated output buffer.

	  Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2

2015-08-21 15:25 +0000 [08a182c8e6]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().

	  Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3

2015-08-21 11:04 +0000 [61f30db877]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_pubsub.c: Add missing error return in load_module().

	  Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc

2015-08-21 11:03 +0000 [b8f07527b2]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip/location.c: Use the builtin ao2_callback() match function instead.

	  Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f

2015-09-10 09:49 +0000 [f1a2e82d49]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Copy default_from_user to avoid crash.

	  The default_from_user retrieval function was pulling the
	  default_from_user from the global configuration struct in an unsafe way.
	  If using a database as a backend configuration store, the global
	  configuration struct is short-lived, so grabbing a pointer from it
	  results in referencing freed memory.

	  The fix here is to copy the default_from_user value out of the global
	  configuration struct.

	  Thanks go to John Hardin for discovering this problem and proposing the
	  patch on which this fix is based.

	  ASTERISK-25390 #close
	  Reported by Mark Michelson

	  Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c

2015-09-10 08:39 +0000 [bd71dcd1da]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route

	  We will only rewrite the Contact header if there is no Record-Route header in
	  the received request. If a malfunctioning proxy places a Record-Route header
	  into a REGISTER request, we will decide that we shouldn't update the IP/port
	  in the Contact header, and we will end up storing a contact with an AoR that
	  contains the NAT'd IP address.

	  While it is nice to have the proxy *not* send a Record-Route in a REGISTER
	  request, it's also a good idea to not process the header in a non-dialog
	  message. This patch updates the code to explicitly ignore the Record-Route
	  header in REGISTER requests.

	  ASTERISK-25387 #close

	  Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f

2015-09-09 16:46 +0000 [5bd363010e]  Alexander Anikin <may213@yandex.ru>

	* chan_ooh323: Add ProgressIndicator IE with inband info available

	  Add ProgressIndicator IE with inband info present to Progress and
	  Alerting Q.931 message

	  ASTERISK-25227 #close
	  Reported by: Alexandr Dranchuk

	  Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203
2015-09-08 10:35 +0000 [fcea6910f6]  Scott Griepentrog <scott@griepentrog.com>

	* pjsip: avoid possible crash req_caps allocation failure

	  Make certain that the pjsip session has not failed to
	  allocate the format capabilities structure, which can
	  otherwise cause a crash when referenced.

	  ASTERISK-25323

	  Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750

2015-09-04 16:33 +0000 [8e5ed27a16]  David M. Lee <dlee@respoke.io>

	* res_rtp_asterisk: Add more ICE debugging

	  In working through a recent ICE negotiation bug, I found the debug
	  logging in res_rtp_asterisk to be lacking. This patch adds a number of
	  debug and warning statements that were helpful.

	  Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80
2015-09-08 07:21 +0000 [3628e380b8]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Use hash for contact object identity instead of Contact URI.

	  In the wild it is possible for Contact URIs to be quite long as
	  parameters can exist on them. This can present a problem when storing
	  them in the AstDB as the URI is used as part of the object name and
	  there is a fixed length limit for the AstDB. This will cause
	  the contact to not get stored.

	  This change uses the MD5 hash of the Contact URI as part of the
	  object name instead. This has a fixed length which is guaranteed
	  to not exceed the AstDB length limit.

	  ASTERISK-25295 #close

	  Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02

2015-09-07 13:19 +0000 [d2106c0b21]  Alexander Anikin <may213@yandex.ru>

	* chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy

	      Call ast_rtp_instance_stop on ooh323_destroy to free resources
	      allocated by rtp instance

	      ASTERISK-25299 #close
	      Report by: Alexandr Dranchuk

	  Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f

2015-09-07 11:15 +0000 [ef3358d0c0]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip: Purge contacts when an AoR is deleted

	  When an AoR is deleted by an external mechanism, such as through ARI, we
	  currently do not remove dynamic contacts that were created for that AoR as a
	  result of a received REGISTER request. As a result, re-creating the AoR will
	  cause the dynamic contact to be interpreted as a persistent contact, leading
	  to some rather strange state being created for the contacts/endpoints.

	  This patch adds a sorcery observer for the 'aor' object. When a delete is
	  issued on the underlying sorcery object, the observer is called, and all
	  contacts created and persisted in sorcery for that AoR are also removed. Note
	  that we don't want to perform this action when an AO2 object that is an AoR is
	  destroyed, as the AoR can still exist in the backing storage (and we would
	  thus be removing valid contacts from an AoR that still "exists".)

	  ASTERISK-25381 #close

	  Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328

2015-09-05 14:58 +0000 [86b02228f5]  Matt Jordan <mjordan@digium.com>

	* channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id

	  This patch adds a new option to the CHANNEL function that allows for the
	  extraction of the SIP call-id. It is used in conjunction with the 'pjsip'
	  option, and will return the Call-ID of the INVITE request that established
	  the PJSIP channel.

	  ASTERISK-25352

	  Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a

2015-09-04 16:06 +0000 [27c89053b0]  David M. Lee <dlee@respoke.io>

	* Fix when remote candidates exceed PJ_ICE_MAX_CAND

	  We were passing the wrong count into pj_ice_sess_create_check_list(),
	  causing the create to fail if we ever received more than PJ_ICE_MAX_CAND
	  candidates.

	  Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378

2015-09-04 14:40 +0000 [993ae9a669]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Change default from user value.

	  When Asterisk sends an outbound SIP request, if there is no direct
	  reason to place a specific value for the username in the From header,
	  Asterisk would generate a UUID. For example, this would happen when
	  sending outbound OPTIONS requests when qualifying or when sending
	  outbound INVITE requests when originating (if no explicit caller ID were
	  provided). The issue is that some SIP providers reject these sorts of
	  requests with a "Name too long" error response.

	  This patch aims to fix this by changing the default outbound username in
	  From headers to "asterisk". This value can be overridden by changing the
	  default_from_user option in the global options if desired.

	  ASTERISK-25377 #close
	  Reported by Mark Michelson

	  Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190

2015-09-03 14:07 +0000 [7d981b787c]  Jonathan Rose <jrose@digium.com>

	* ParkAndAnnounce: Add variable inheritance

	  In Asterisk 11, the announcer channel would receive channel variables
	  from the channel being parked by means of normal channel inheritance.
	  This functionality was lost during the big res_parking project in
	  Asterisk 12. This patch restores that functionality.

	  ASTERISK-25369 #close
	  Review: https://gerrit.asterisk.org/#/c/1180/

	  Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e

2015-09-04 09:26 +0000 [7691035312]  Scott Griepentrog <scott@griepentrog.com>

	* endpoint snapshot: avoid second cleanup on alloc failure

	  In ast_endpoint_snapshot_create(), a failure to init the
	  string fields results in two attempts to ao2_cleanup the
	  same pointer.  Removed RAII_VAR to eliminate problem.

	  ASTERISK-25375 #close
	  Reported by: Scott Griepentrog

	  Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979

2015-09-04 05:33 +0000 [be31747db8]  Martin Tomec <tomec.martin@gmail.com>

	* res/pjsip: Mark WSS transport as secure

	  Pjsip is refusing to use unsecure transport with "sips" in url.
	  WSS should be considered as secure transport.

	  ASTERISK-24602 #comment Partially fixed by setting WSS as secure

	  Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353

2015-09-01 10:16 +0000 [fbdb42c9fc]  Guido Falsi <madpilot@freebsd.org>

	* Core/General: Add #ifdef needed on FreeBSD.

	  pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD
	  too.

	  ASTERISK-25310 #close
	  Reported by: Guido Falsi

	  Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4
2015-09-02 17:26 +0000 [c15d8cc0ed]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Fix contact refleak on stateful responses.

	  When sending a stateful response, creation of the transaction can fail,
	  most commonly because we are trying to create a transaction from a
	  retransmitted request. When creation of the transaction fails, we end up
	  leaking a reference to a contact that was bumped when the response was
	  created.

	  This patch adds the missing deref and fixes the reference leak.

	  Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07

2015-09-02 12:41 +0000 [b51cf1e712]  Joshua Colp <jcolp@digium.com>

	* pbx: Fix crash when issuing "core show hints" with long pattern match.

	  When issuing the "core show hints" CLI command a combination of both
	  the hint extension and context is created. This uses a fixed size
	  buffer expecting that the extension will not exceed maximum extension
	  length. When the extension is actually a pattern match this constraint
	  does not hold true, and the extension may exceed the maximum extension
	  length. In this case extra characters are written past the end of the
	  fixed size buffer.

	  This change makes it so the construction of the combined hint extension
	  and context can not exceed the size of the buffer.

	  ASTERISK-25367 #close

	  Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499

2015-09-01 09:05 +0000 [beb568e51c]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: re-re-fix persistent subscription storage.

	  A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as
	  a means of writing an appropriate packet to persistent storage. While
	  this partially solved the issue, it had its own problems.
	  pjsip_msg_print will always add a Content-Length header to the message
	  it prints. Frequent restarts of Asterisk can result in persistent
	  subscriptions being written with five or more Content-Length headers. In
	  addition, sometimes some apparent corruption of individual headers could
	  be seen.

	  This aims to fix the problem by not running a parsed message through an
	  interpreter but rather by taking the raw message and saving it. The
	  logic for what to save is going to be different depending on whether a
	  SUBSCRIBE was received from the wire or if it was pulled from
	  persistence. When receiving a packet from the wire, when using a
	  streaming transport, the rdata->pkt_info.packet may contain multiple SIP
	  messages or fragments. However, the rdata->msg_info.msg_buf will always
	  contain the current SIP message to be processed. When pulling from
	  persistence, though, the rdata->msg_info.msg_buf will be NULL since no
	  transport actually handled the packet. However, since we know that we
	  will always ever pull one SIP message from persistence, we are free to
	  save directly from rdata->pkt_info.packet instead.

	  ASTERISK-25365 #close
	  Reported by Mark Michelson

	  Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b

2015-08-29 10:36 +0000 [fc4d4f5379]  Joshua Colp <jcolp@digium.com>

	* taskprocessor: Fix race condition between unreferencing and finding.

	  When unreferencing a taskprocessor its reference count is checked
	  to determine if it should be unlinked from the taskprocessors
	  container and its listener shut down. In between the time when the
	  reference count is checked and unlinking it is possible for
	  another thread to jump in, find it, and get a reference to it. If
	  the thread then uses the taskprocessor it may find that it is not
	  in the state it expects.

	  This change locks the taskprocessors container during almost the
	  entire unreference operation to ensure that any other thread which
	  may attempt to find the taskprocessor has to wait.

	  ASTERISK-25295

	  Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c

2015-08-28 20:22 +0000 [bb38010c67]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.

	  The keepalive support in res_pjsip_sdp_rtp currently assumes
	  that a stream will only be negotiated once. This is false.
	  If the stream is replaced and later added back it can be
	  negotiated again causing multiple keepalive scheduled items
	  to exist. This change explicitly deletes the existing
	  keepalive scheduled item before adding the new one.

	  The res_pjsip_sdp_rtp module also does not stop RTP
	  keepalives or timeout timer if the stream has been
	  replaced. This change adds a callback to the session media
	  interface to allow a media stream to be stopped without
	  the resources being destroyed. This allows the scheduled
	  items and RTP to be stopped when the stream no longer
	  exists.

	  ASTERISK-25356 #close

	  Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de

2015-08-28 19:57 +0000 [c036e50fbe]  Joshua Colp <jcolp@digium.com>

	* sched: ast_sched_del may return prematurely due to spurious wakeup

	  When deleting a scheduled item if the item in question is currently
	  executing the ast_sched_del function waits until it has completed.
	  This is accomplished using ast_cond_wait. Unfortunately the
	  ast_cond_wait function can suffer from spurious wakeups so the
	  predicate needs to be checked after it returns to make sure it has
	  really woken up as a result of being signaled.

	  This change adds a loop around the ast_cond_wait to make sure that
	  it only exits when the executing task has really completed.

	  ASTERISK-25355 #close

	  Change-Id: I51198270eb0b637c956c61aa409f46283432be61

2015-08-27 12:26 +0000 [229b95d253]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_session: Don't invoke session supplements twice for BYE requests.

	  When a BYE request is received the PJSIP invite session implementation
	  creates and sends a 200 OK response before we are aware of it. This
	  causes the INVITE session state callback to be called into and ultimately
	  the session supplements run on the BYE request. Once this response has
	  been sent the normal transaction state callback is invoked which
	  invokes the session supplements on the BYE request again. This can
	  be problematic in particular with res_pjsip_rfc3326 as it may
	  attempt to update the hangup cause code on the channel while it is
	  in the process of being hung up.

	  This change makes it so the session supplements are only invoked
	  once by the INVITE session state callback.

	  ASTERISK-25318 #close

	  Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a

2015-08-26 15:26 +0000 [6bfa14bdad]  Scott Griepentrog <scott@griepentrog.com>

	* Chaos: handle failed allocation in get_media_encryption_type

	  If the ast_strndup() call fails to allocate a copy of the
	  transport string for parsing, fail gracefully.

	  ASTERISK-25323
	  Reported by: Scott Griepentrog

	  Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28

2015-08-26 14:25 +0000 [490db8ba94]  Scott Griepentrog <scott@griepentrog.com>

	* Chaos: make hangup NULL tolerant

	  In chan_pjsip_new, if allocation of the pvt
	  structure fails, ast_hangup is called.  But
	  it was written to assume pvt was valid, and
	  this change corrects that.

	  ASTERISK-25323
	  Reported by: Scott Griepentrog

	  Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87
2015-08-26 05:40 +0000 [d03d09aad3]  Joshua Colp <jcolp@digium.com>

	* chan_sip: Allow call pickup to set the hangup cause.

	  The call pickup implementation in chan_sip currently sets the channel
	  hangup cause to "normal clearing" if call pickup is successfully
	  performed. This action overwrites the "answered elsewhere" hangup cause
	  set by the call pickup code and can result in the SIP device in
	  question showing a missed call when it should not.

	  This change sets the hangup cause to "normal clearing" as a
	  default initially but allows the call pickup to change it as
	  needed.

	  ASTERISK-25346 #close

	  Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff

2015-08-25 07:17 +0000 [d013ecf748]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Add common ast_sip_get_host_ip API.

	  Modules commonly used the pj_gethostip function for retrieving the
	  IP address of the host. This function does not cache the result and may
	  result in a DNS lookup occurring, or additional work. If the DNS
	  server is unreachable or network issues arise this can cause the
	  pj_gethostip function to block for a period of time.

	  This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string
	  function which does the same thing but caches the host IP address at
	  module load time. This results in no additional work being done each
	  time the local host IP address is needed.

	  ASTERISK-25342 #close

	  Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e

2015-08-24 06:21 +0000 [98d089fb9a]  Joshua Colp <jcolp@digium.com>

	* bridge: Kick channel from bridge if hung up during action.

	  When executing an action in a bridge it is possible for the
	  channel to be hung up without the bridge becoming aware of it.
	  This is most easily reproducible by hanging up when the bridge
	  is streaming DTMF due to a feature timeout. This change makes
	  it so after action execution the channel is checked to determine
	  if it has been hung up and if it has it is kicked from the bridge.

	  ASTERISK-25341 #close

	  Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062

2015-08-24 11:04 +0000 [a408369bac]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced

	  When recreating a subscription it is possible for a freed sub_tree
	  to be referenced when the initial NOTIFY fails to be created.

	  Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788

2015-08-23 18:26 +0000 [3af34441eb]  Matt Jordan <mjordan@digium.com>

	* res_pjsip/pjsip_configuration: Disregard empty auth values

	  When an endpoint is backed by a non-static conf file backend (such as
	  the AstDB or Realtime), the 'auth' object may be returned as being an
	  empty string. Currently, res_pjsip will interpret that as being a valid
	  auth object, and will attempt to authenticate inbound requests. This
	  isn't desired; is an auth value is empty (which the name of an auth
	  object cannot be), we should instead interpret that as being an invalid
	  auth object and skip it.

	  ASTERISK-25339 #close

	  Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7

2015-08-21 23:37 +0000 [89003ea320]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* README*: Remove trailing whitespace

	  Change-Id: I18b7d75187548a9ed55b4f258d21aaaf29d08874

2015-07-28 13:47 +0000 [857923d9c7]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Set preferred rx payload type mapping on incoming offers.

	  ASTERISK-25166
	  Reported by: Kevin Harwell

	  ASTERISK-17410
	  Reported by: Boris Fox

	  Change-Id: I7f04d5c8bee1126fee5fe6afbc39e45104469f4e

2015-07-24 18:46 +0000 [d643b206c6]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_sdp_rtp.c: Set preferred rx payload type mapping on incoming offers.

	  ASTERISK-25166
	  Reported by: Kevin Harwell

	  ASTERISK-17410
	  Reported by: Boris Fox

	  Change-Id: I97ecebc1ab9b5654fb918bf1f4c98c956b852369

2015-07-27 19:19 +0000 [f7df3e1a01]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.c: Get current or create a needed rx payload type mapping.

	  * Make ast_rtp_codecs_payload_code() get the current mapping or create a
	  rx payload type mapping.

	  ASTERISK-25166
	  Reported by: Kevin Harwell

	  ASTERISK-17410
	  Reported by: Boris Fox

	  Change-Id: Ia4b2d45877a8f004f6ce3840e3d8afe533384e56

2015-07-27 19:15 +0000 [38854a9f7b]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.c: Extract rtp_codecs_payload_replace_rx().

	  ASTERISK-25166
	  Reported by: Kevin Harwell

	  ASTERISK-17410
	  Reported by: Boris Fox

	  Change-Id: I34e23bf5b084c8570f9c3e6ccd19b95fe85af239

2015-07-23 19:24 +0000 [1a549ed134]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.c: Initial split of payload types into rx and tx mappings.

	  There are numerous problems with the current implementation of the RTP
	  payload type mapping in Asterisk.  It uses only one mapping structure to
	  associate payload types to codecs.  The single mapping is overkill if all
	  of the payload type values are well known values.  Dynamic payload type
	  mappings do not work as well with the single mapping because RFC3264
	  allows each side of the link to negotiate different dynamic mappings for
	  what they want to receive.  Not only could you have the same codec mapped
	  for sending and receiving on different payload types you could wind up
	  with the same payload type mapped to different codecs for each direction.

	  1) An independent payload type mapping is needed for sending and
	  receiving.

	  2) The receive mapping needs to keep track of previous mappings because of
	  the slack to when negotiation happens and current packets in flight using
	  the old mapping arrive.

	  3) The transmit mapping only needs to keep track of the current negotiated
	  values since we are sending the packets and know when the switchover takes
	  place.

	  * Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
	  use the new function because ast_rtp_codecs_payload_code() was used for
	  mappings in both directions.

	  * Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
	  to pass preferred codec mappings to the peer channel for early media
	  bridging or when we need to prefer the offered mapping that RFC3264 says
	  we SHOULD use.

	  * ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
	  the only new public functions created.  All the others were only used for
	  the tx or rx mapping direction so the function doxygen now reflects which
	  direction the function operates.

	  * chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
	  that makes no sense when processing an incoming SDP.  We would be wiping
	  out any mappings that we set for the possible outgoing SDP we sent
	  earlier.

	  ASTERISK-25166
	  Reported by: Kevin Harwell

	  ASTERISK-17410
	  Reported by: Boris Fox

	  Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac

2015-08-19 12:10 +0000 [21d419e4fc]  Richard Mudgett <rmudgett@digium.com>

	* ari/ari_websockets.c: Fix ast_debug parameter type mismatch.

	  This is a type mismatch fix of the debugging commit
	  c63316eec10e1990a88bf4712238d6deb375bfa9 made to find out why
	  a testsuite test was failing only on one of the continuous
	  integration build agents.

	  Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75

2015-08-19 10:30 +0000 [53e2a6a829]  Scott Griepentrog <scott@griepentrog.com>

	* contrib: script install_prereq should install sqlite3

	  Asterisk needs the sqlite 3 library, which is package
	  sqlite-devel in CentOS. By adding this package to the
	  script, a problem with configure failing is resolved.

	  ASTERISK-25331 #close
	  Reported by: Kevin Harwell

	  Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec

2015-08-18 15:07 +0000 [03eb6cbc10]  Richard Mudgett <rmudgett@digium.com>

	* res_ari_events: Fix shutdown ref leak.

	  ASTERISK-25308 #close
	  Reported by: Joshua Colp

	  Change-Id: I592785bf70ff4b63d00e535b482f40da8e82a082

2015-08-18 14:24 +0000 [e1e7e205bc]  Richard Mudgett <rmudgett@digium.com>

	* res_http_websocket.c: Add missing unref on an off nominal path.

	  Change-Id: I228df6adecd4cb450d03e09e9a38c86bb566e811

2015-08-18 16:06 +0000 [59253a2262]  Richard Mudgett <rmudgett@digium.com>

	* res_http_websocket.c: Fix some off nominal path cleanup.

	  * Remove extraneous unlock on off-nominal path.
	  * Add missing HTTP error reply.

	  Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b

2015-08-18 14:46 +0000 [1f0a9f8a76]  Richard Mudgett <rmudgett@digium.com>

	* res_ari.c: Add missing off nominal unlock and remove a RAII_VAR().

	  Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf

2015-08-14 12:55 +0000 [9fb4a96e15]  Richard Mudgett <rmudgett@digium.com>

	* app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'.

	  Setting the 'paused' and 'ringinuse' options on a queue member using the
	  dialplan function QUEUE_MEMBER did not behave the same way as the
	  equivalent dialplan applications or AMI actions.

	  * Made queue_function_mem_write() call the set_member_paused() and
	  set_member_value() for the 'paused' and 'ringinuse' options respectively.
	  A beneficial side effect is that the queue name is now optional and sets
	  the value in all queues the interface is a member.

	  * Update QUEUE_MEMBER XML documentation.

	  * Fix error checking in QUEUE_MEMBER() write.

	  ASTERISK-25215 #close
	  Reported by: Lorne Gaetz

	  Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb

2015-08-17 16:41 +0000 [87b22969a4]  Richard Mudgett <rmudgett@digium.com>

	* app_queue.c: Extract some functions for simpler code.

	  * Extract set_queue_member_pause() from set_member_paused() for simpler
	  and more consistent code.

	  * Extract set_queue_member_ringinuse() from
	  set_member_ringinuse_help_members() for simpler code.

	  Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306

2015-08-17 13:34 +0000 [5cf98e2459]  Richard Mudgett <rmudgett@digium.com>

	* app_queue.c: Fix error checking in QUEUE_MEMBER() read.

	  Change-Id: I7294e13d27875851c2f4ef6818adba507509d224

2015-08-17 11:00 +0000 [178e1adffb]  Scott Griepentrog <scott@griepentrog.com>

	* CHAOS: prevent sorcery object with null id

	  When allocating a sorcery object, fail if the
	  id value was not allocated.

	  ASTERISK-25323
	  Reported by: Scott Griepentrog

	  Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e

2015-08-14 15:46 +0000 [5a85711568]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_sdp_rtp: Restore removed NULL check.

	  When sending an RTP keepalive, we need to be sure we're not dealing with
	  a NULL RTP instance. There had been a NULL check, but the commit that
	  added the rtp_timeout and rtp_hold_timeout options removed the NULL
	  check.

	  Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64

2015-08-13 12:30 +0000 [7c4cb8618d]  Richard Mudgett <rmudgett@digium.com>

	* audiohook.c: Simplify variable usage in audiohook_read_frame_both().

	  Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c

2015-08-13 12:22 +0000 [bb37473234]  Richard Mudgett <rmudgett@digium.com>

	* audiohook.c: Fix MixMonitor crash when using the r() or t() options.

	  The built frame format in audiohook_read_frame_both() is now set to a
	  signed linear format before the rx and tx frames are duplicated instead of
	  only for the mixed audio frame duplication.

	  ASTERISK-25322 #close
	  Reported by Sean Pimental

	  Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538

2015-08-12 12:59 +0000 [43bdddfc26]  Kevin Harwell <kharwell@digium.com>

	* chan_sip.c: wrong peer searched in sip_report_security_event

	  In chan_sip, after handling an incoming invite a security event is raised
	  describing authorization (success, failure, etc...). However, it was doing
	  a lookup of the peer by extension. This is fine for register messages, but
	  in the case of an invite it may search and find the wrong peer, or a non
	  existent one (for instance, in the case of call pickup). Also, if the peers
	  are configured through realtime this may cause an unnecessary database lookup
	  when caching is enabled.

	  This patch makes it so that sip_report_security_event searches by IP address
	  when looking for a peer instead of by extension after an invite is processed.

	  ASTERISK-25320 #close

	  Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4
2015-08-13 05:26 +0000 [495dfb24b7]  Joshua Colp <jcolp@digium.com>

	* res_http_websocket: When shutting down a session don't close closed socket

	  Due to the use of ast_websocket_close in session termination it is
	  possible for the underlying socket to already be closed when the
	  session is terminated. This occurs when the close frame is attempted
	  to be written out but fails.

	  Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b
2015-08-11 05:24 +0000 [7e65be4ecd]  Joshua Colp <jcolp@digium.com>

	* res_http_websocket: Forcefully terminate on write errors.

	  The res_http_websocket module will currently attempt to close
	  the WebSocket connection if fatal cases occur, such as when
	  attempting to write out data and being unable to. When the
	  fatal cases occur the code attempts to write a WebSocket close
	  frame out to have the remote side close the connection. If
	  writing this fails then the connection is not terminated.

	  This change forcefully terminates the connection if the
	  WebSocket is to be closed but is unable to send the close frame.

	  ASTERISK-25312 #close

	  Change-Id: I10973086671cc192a76424060d9ec8e688602845

2015-08-09 18:42 +0000 [a87e2dd254]  Matt Jordan <mjordan@digium.com>

	* res/res_format_attr_silk: Expose format attributes to other modules

	  This patch adds the .get callback to the format attribute module, such
	  that the Asterisk core or other third party modules can query for the
	  negotiated format attributes.

	  Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c

2015-08-10 13:43 +0000 [87c92d2aee]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF.

	  Pressing DTMF digits on a phone to go out on a DAHDI channel can result in
	  the digit not being recognized or even heard by the peer.

	  Phone -> Asterisk -> DAHDI/channel

	  Turns out the DAHDI behavior with DTMF generation (and any other generated
	  tones) is exposed by the "buffers=" setting in chan_dahdi.conf.  When
	  Asterisk requests to start sending DTMF then DAHDI waits until its write
	  buffer is empty before generating any samples for the DTMF tones.  When
	  Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI
	  immediately stops generating the DTMF samples.  As a result, the more
	  samples there are in the DAHDI write buffer the shorter the time DTMF
	  actually gets sent on the wire.  If there are more samples in the write
	  buffer than the time DTMF is supposed to be sent then no DTMF gets sent on
	  the wire.  With the "buffers=12,half" setting and each buffer representing
	  20 ms of samples then the DAHDI write buffer is going to contain around
	  120 ms of samples.  For DTMF to be recognized by the peer the actual sent
	  DTMF duration needs to be a minimum of 40 ms.  Therefore, the intended
	  duration needs to be a minimum of 160 ms for the peer to receive the
	  minimum DTMF digit duration to recognize it.

	  A simple and effective solution to work around the DAHDI behavior is for
	  Asterisk to flush the DAHDI write buffer when sending DTMF so the full
	  duration of DTMF is actually sent on the wire.  When someone is going to
	  send DTMF they are not likely to be talking before sending the tones so
	  the flushed write samples are expected to just contain silence.

	  * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting
	  to send a DTMF digit.

	  ASTERISK-25315 #close
	  Reported by John Hardin

	  Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a

2015-08-05 14:21 +0000 [b9b957d4e9]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi.c: Lock private struct for ast_write().

	  There is a window of opportunity for DTMF to not go out if an audio frame
	  is in the process of being written to DAHDI while another thread starts
	  sending DTMF.  The thread sending the audio frame could be past the
	  currently dialing check before being preempted by another thread starting
	  a DTMF generation request.  When the thread sending the audio frame
	  resumes it will then cause DAHDI to stop the DTMF tone generation.  The
	  result is no DTMF goes out.

	  * Made dahdi_write() lock the private struct before writing to the DAHDI
	  file descriptor.

	  ASTERISK-25315
	  Reported by John Hardin

	  Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb

2015-08-10 18:23 +0000 [f3f5b45d57]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message.

	  If the saved SUBSCRIBE message is not parseable for whatever reason then
	  Asterisk could crash when libpjsip tries to parse the message and adds an
	  error message to the parse error list.

	  * Made ast_sip_create_rdata() initialize the parse error rdata list.  The
	  list is checked after parsing to see that it remains empty for the
	  function to return successful.

	  ASTERISK-25306
	  Reported by Mark Michelson

	  Change-Id: Ie0677f69f707503b1a37df18723bd59418085256

2015-08-10 07:40 +0000 [991d4da1eb]  Alexander Traud <pabstraud@compuserve.com>

	* chan_sip: Fix negotiation of iLBC 30.

	  iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is
	  supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5,
	  only iLBC 30 is negotiated now.

	  ASTERISK-25309 #close

	  Change-Id: I92d724600a183eec3114da0ac607b994b1a793da

2015-08-09 17:56 +0000 [e188192ad1]  Matt Jordan <mjordan@digium.com>

	* main/format: Add an API call for retrieving format attributes

	  Some codecs that may be a third party library to Asterisk need to have
	  knowledge of the format attributes that were negotiated. Unfortunately,
	  when the great format migration of Asterisk 13 occurred, that ability
	  was lost.

	  This patch adds an API call, ast_format_attribute_get, to the core
	  format API, along with updates to the unit test to check the new API
	  call. A new callback is also now available for format attribute modules,
	  such that they can provide the format attribute values they manage.

	  Note that the API returns a void *. This is done as the format attribute
	  modules themselves may store format attributes in any particular manner
	  they like. Care should be taken by consumers of the API to check the
	  return value before casting and dereferencing. Consumers will obviously
	  need to have a priori knowledge of the type of the format attribute as
	  well.

	  Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3

2015-08-07 22:11 +0000 [d5f0c27122]  David M. Lee <dlee@respoke.io>

	* Replace htobe64 with htonll

	  We don't have a compatability function to fill in a missing htobe64; but
	  we already have one for the identical htonll.

	  Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac

2015-07-24 17:04 +0000 [40caf0ad9b]  David M. Lee <dlee@respoke.io>

	* Replaces clock_gettime() with ast_tsnow()

	  clock_gettime() is, unfortunately, not portable. But I did like that
	  over our usual `ts.tv_nsec = tv.tv_usec * 1000` copy/paste code we
	  usually do when we want a timespec and all we have is ast_tvnow().

	  This patch adds ast_tsnow(), which mimics ast_tvnow(), but returns a
	  timespec. If clock_gettime() is available, it will use that. Otherwise
	  ast_tsnow() falls back to using ast_tvnow().

	  Change-Id: Ibb1ee67ccf4826b9b76d5a5eb62e90b29b6c456e

2015-08-07 14:20 +0000 [12e6f5ac01]  Scott Emidy <jemidy@digium.com>

	* ARI: Retrieve existing log channels

	  An http request can be sent to get the existing Asterisk logs.

	  The command "curl -v -u user:pass -X GET 'http://localhost:8088
	  /ari/asterisk/logging'" can be run in the terminal to access the
	  newly implemented functionality.

	  * Retrieve all existing log channels

	  ASTERISK-25252

	  Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808

2015-08-07 11:14 +0000 [b91ca7ba49]  Scott Emidy <jemidy@digium.com>

	* ARI: Creating log channels

	  An http request can be sent to create a log channel
	  in Asterisk.

	  The command "curl -v -u user:pass -X POST
	  'http://localhost:088/ari/asterisk/logging/mylog?
	  configuration=notice,warning'" can be run in the terminal
	  to access the newly implemented functionality for ARI.

	  * Ability to create log channels using ARI

	  ASTERISK-25252

	  Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782

2015-08-06 15:18 +0000 [f19c4930c2]  Scott Emidy <jemidy@digium.com>

	* ARI: Deleting log channels

	  An http request can be sent to delete a log channel
	  in Asterisk.

	  The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
	  /ari/asterisk/logging/mylog'" can be run in the terminal
	  to access the newly implemented functionally for ARI.

	  * Able to delete log channels using ARI

	  ASTERISK-25252

	  Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6

2015-08-06 12:48 +0000 [382334cc06]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: More accurately persist packet.

	  The pjsip_rx_data structure has a pkt_info.packet field on it that is
	  the packet that was read from the transport. For datagram transports,
	  the packet read from the transport will correspond to the SIP message
	  that arrived. For streamed transports, however, it is possible to read
	  multiple SIP messages in one packet.

	  In a recent case, Asterisk crashed on a system where TCP was being used.
	  This is because at some point, a read from the TCP socket resulted in a
	  200 OK response as well as an incoming SUBSCRIBE request being stored in
	  rdata->pkt_info.packet. When the SUBSCRIBE was processed, the
	  combination 200 OK and SUBSCRIBE was saved in persistent storage. Later,
	  a restart of Asterisk resulted in the crash because the persistent
	  subscription recreation code ended up building the 200 OK response
	  instead of a SUBSCRIBE request, and we attempted to access
	  request-specific data.

	  The fix here is to use the pjsip_msg_print() function in order to
	  persist SUBSCRIBE requests. This way, rather than using the raw socket
	  data, we use the parsed SIP message that PJSIP has given us. If we
	  receive multiple SIP messages from a single read, we will be sure only
	  to save off the relevant SIP message. There also is a safeguard put in
	  place to make sure that if we do end up reconstructing a SIP response,
	  it will not cause a crash.

	  ASTERISK-25306 #close
	  Reported by Mark Michelson

	  Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2

2015-08-04 16:12 +0000 [4b6c657a82]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Ensure sanitized XML is NULL terminated.

	  The ast_sip_sanitize_xml function is used to sanitize
	  a string for placement into XML. This is done by examining
	  an input string and then appending values to an output
	  buffer. The function used by its implementation, strncat,
	  has specific behavior that was not taken into account.
	  If the size of the input string exceeded the available
	  output buffer size it was possible for the sanitization
	  function to write past the output buffer itself causing
	  a crash. The crash would either occur because it was
	  writing into memory it shouldn't be or because the resulting
	  string was not NULL terminated.

	  This change keeps count of how much remaining space is
	  available in the output buffer for text and only allows
	  strncat to use that amount.

	  Since this was exposed by the res_pjsip_pidf_digium_body_supplement
	  module attempting to send a large message the maximum allowed
	  message size has also been increased in it.

	  A unit test has also been added which confirms that the
	  ast_sip_sanitize_xml function is providing NULL terminated
	  output even when the input length exceeds the output
	  buffer size.

	  ASTERISK-25304 #close

	  Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302

2015-08-05 05:23 +0000 [7351d33a1f]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Don't leak temporary key when enabling PFS.

	  A change recently went in which enabled perfect forward secrecy for
	  DTLS in res_rtp_asterisk. This was accomplished two different ways
	  depending on the availability of a feature in OpenSSL. The fallback
	  method created a temporary instance of a key but did not free it.
	  This change fixes that.

	  ASTERISK-25265

	  Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396
2015-08-04 09:47 +0000 [c63316eec1]  Mark Michelson <mmichelson@digium.com>

	* res_http_websocket: Debug write lengths.

	  Commit 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee attempted to fix a
	  test failure observed on 32 bit test agents by ensuring that a cast from
	  a 32 bit unsigned integer to a 64 bit unsigned integer was happening in
	  a predictable place. As it turns out, this did not cause test runs to
	  succeed.

	  This commit adds several redundant debug messages that print the payload
	  lengths of websocket frames. The idea here is that this commit will not
	  cause tests to succeed for the faulty test agent, but we might deduce
	  where the fault lies more easily this way by observing at what point the
	  expected value (537) changes to some ungangly huge number.

	  If you are wondering why something like this is being committed to the
	  branch, keep in mind that in commit
	  39cc28f6ea2140ad6d561fd4c9e9a66f065cecee I noted that the observed test
	  failures only happen when automated tests are run. Attempts to run the
	  tests by hand manually on the test agent result in the tests passing.

	  Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d

2015-08-03 11:06 +0000 [35a98161df]  Mark Michelson <mmichelson@digium.com>

	* res_http_websocket: Avoid passing strlen() to ast_websocket_write().

	  We have seen a rash of test failures on a 32-bit build agent. Commit
	  48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where
	  we were not encoding a 64-bit value correctly over the wire. This
	  commit, however, did not solve the test failures.

	  In the failing tests, ARI is attempting to send a 537 byte text frame
	  over a websocket. When sending a frame this small, 16 bits are all that
	  is required in order to encode the payload length on the websocket
	  frame. However, ast_websocket_write() thinks that the payload length is
	  greater than 65535 and therefore writes out a 64 bit payload length.
	  Inspecting this payload length, the lower 32 bits are exactly what we
	  would expect it to be, 537 in hex. The upper 32 bits, are junk values
	  that are not expected to be there.

	  In the failure, we are passing the result of strlen() to a function that
	  expects a uint64_t parameter to be passed in. strlen() returns a size_t,
	  which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit
	  unsigned value to somewhere where a 64-bit unsigned value is expected
	  would cause no problems. In fact, in manual runs of failing tests, this
	  works just fine. However, ast_websocket_write() uses the Asterisk
	  optional API, which means that rather than a simple function call, there
	  are a series of macros that are used for its declaration and
	  implementation. These macros may be causing some sort of error to occur
	  when converting from a 32 bit quantity to a 64 bit quantity.

	  This commit changes the logic by making existing ast_websocket_write()
	  calls use ast_websocket_write_string() instead. Within
	  ast_websocket_write_string(), the 64-bit converted strlen is saved in a
	  local variable, and that variable is passed to ast_websocket_write()
	  instead.

	  Note that this commit message is full of speculation rather than
	  certainty. This is because the observed test failures, while always
	  present in automated test runs, never occur when tests are manually
	  attempted on the same test agent. The idea behind this commit is to fix
	  a theoretical issue by performing changes that should, at the least,
	  cause no harm. If it turns out that this change does not fix the failing
	  tests, then this commit should be reverted.

	  Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67

2015-07-29 14:17 +0000 [1f02d20da4]  Benjamin Ford <bford@digium.com>

	* ARI: Rotate log channels.

	  An http request can be sent to rotate a specified log channel.
	  If the channel does not exist, an error response will be
	  returned.

	  The command "curl -v -u user:pass -X PUT 'http://localhost:8088
	  /ari/asterisk/logging/logChannelName/rotate'" can be run in the
	  terminal to access this new functionality.

	  * Added the ability to rotate log files through ARI

	  ASTERISK-25252

	  Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01

2015-07-31 11:27 +0000 [fe804b09b3]  Ashley Sanders <asanders@digium.com>

	* ARI: Channels added to Stasis application during WebSocket creation ...

	  Prior to ASTERISK-24988, the WebSocket handshake was resolved before Stasis
	  applications were registered. This was done such that the WebSocket would be
	  ready when an application is registered. However, by creating the WebSocket
	  first, the client had the ability to make requests for the Stasis application
	  it thought had been created with the initial handshake request. The inevitable
	  conclusion of this scenario was the cart being put before the horse.

	  ASTERISK-24988 resolved half of the problem by ensuring that the applications
	  were created and registered with Stasis prior to completing the handshake
	  with the client. While this meant that Stasis was ready when the client
	  received the green-light from Asterisk, it also meant that the WebSocket was
	  not yet ready for Stasis to dispatch messages.

	  This patch introduces a message queuing mechanism for delaying messages from
	  Stasis applications while the WebSocket is being constructed. When the ARI
	  event processor receives the message from the WebSocket that it is being
	  created, the event processor instantiates an event session which contains a
	  message queue. It then tries to create and register the requested applications
	  with Stasis. Messages that are dispatched from Stasis between this point and
	  the point at which the event processor is notified the WebSocket is ready, are
	  stashed in the queue. Once the WebSocket has been built, the queue's messages
	  are dispatched in the order in which they were originally received and the
	  queue is concurrently cleared.

	  ASTERISK-25181 #close
	  Reported By: Matt Jordan

	  Change-Id: Iafef7b85a2e0bf78c114db4c87ffc3d16d671a17

2015-07-29 12:58 +0000 [86034227ca]  Mark Michelson <mmichelson@digium.com>

	* dns_core: Allow zero-length DNS responses.

	  A testsuite test recently failed due to a crash that occurred in the DNS
	  core. The problem was that the test could not resolve an address, did
	  not set a result on the DNS query, and then indicated the query was
	  completed. The DNS core does not handle the case of a query with no
	  result gracefully, and so there is a crash.

	  This changeset makes the DNS system resolver set a result with a
	  zero-length answer in the case that a DNS resolution failure occurs
	  early. The DNS core now also will accept such a response without
	  treating it as invalid input. A unit test was updated to no longer treat
	  setting a zero-length response as off-nominal.

	  Change-Id: Ie56641e22debdaa61459e1c9a042e23b78affbf6

2015-07-29 13:49 +0000 [f49bef08a2]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.c: Fix performance issue with several channel drivers that use RTP.

	  ast_rtp_codecs_get_payload() gets called once or twice for every received
	  RTP frame so it would be nice to not allocate an ao2 object to then have
	  it destroyed shortly thereafter.  The ao2 object gets allocated only if
	  the payload type is not set by the channel driver as a negotiated value.
	  The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323.

	  * Made static_RTP_PT[] an array of ao2 objects that
	  ast_rtp_codecs_get_payload() can return instead of an array of structs
	  that must be copied into a created ao2 object.

	  ASTERISK-25296 #close
	  Reported by: Richard Mudgett

	  Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0

2015-07-29 17:00 +0000 [33a465249b]  Richard Mudgett <rmudgett@digium.com>

	* res_rtp_asterisk.c: Fix off-nominal crash potential.

	  ASTERISK-25296
	  Reported by: Richard Mudgett

	  Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b

2015-07-29 13:48 +0000 [5f925d48b7]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.c: Must protect mime_types_len with mime_types_lock.

	  Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e

2015-07-24 18:38 +0000 [ba7dd38470]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_sdp_rtp.c: Fixup some whitespace.

	  Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973

2015-07-24 18:42 +0000 [3751bf0971]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list.

	  Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2

2015-07-27 19:10 +0000 [e2d5d4db35]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.h: No sense allowing payload types larger than RFC allows.

	  * Tweaked add_static_payload() to not use magic numbers.

	  Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b

2015-07-23 14:04 +0000 [bc1eae55cb]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.c: Minor tweaks.

	  * Fix off nominial ref leak of new_type in
	  ast_rtp_codecs_payloads_set_m_type().

	  * No need to lock static_RTP_PT_lock in
	  ast_rtp_codecs_payloads_set_m_type() and
	  ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type
	  parameter sanity check.

	  * No need to create ast_rtp_payload_type ao2 objects with a lock since the
	  lock is not used.

	  Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4

2015-07-17 16:23 +0000 [d122c1e50b]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip.c: Tweak glue->update_peer() parameter nil value.

	  Change glue->update_peer() parameter from 0 to NULL to better indicate it
	  is a pointer.

	  Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd

2015-07-23 12:41 +0000 [d12dc97fc9]  Richard Mudgett <rmudgett@digium.com>

	* rtp_engine.h: Misc comment fixes.

	  Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43

2015-07-30 17:05 +0000 [077c58cd5c]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Fix crashes seen when call cancelled.

	  Two testsuite tests crashed in the same place as a result of an INVITE
	  being CANCELed.

	  tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified
	  tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp

	  The session pointer is no longer in the inv->mod_data[session_module.id]
	  location because the INVITE transaction has reached the terminated state.

	  ASTERISK-25297 #close
	  Reported by: Richard Mudgett

	  Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427

2015-07-29 14:35 +0000 [5fcd1bc556]  Mark Michelson <mmichelson@digium.com>

	* res_http_websocket: Properly encode 64 bit payload

	  A test agent was continuously failing all ARI tests when run against
	  Asterisk 13. As it turns out, the reason for this is that on those test
	  runs, for some reason we decided to use the super extended 64 bit
	  payload length for websocket text frames instead of the extended 16 bit
	  payload length. For 64-bit payloads, the expected byte order over the
	  network is

	  7, 6, 5, 4, 3, 2, 1, 0

	  However, we were sending the payload as

	  3, 2, 1, 0, 7, 6, 5, 4

	  This meant that we were saying to expect an absolutely MASSIVE payload
	  to arrive. Since we did not follow through on this expected payload
	  size, the client would sit patiently waiting for the rest of the payload
	  to arrive until the test would time out.

	  With this change, we use the htobe64() function instead of htonl() so
	  that a 64-bit byte-swap is performed instead of a 32 bit byte-swap.

	  Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a

2015-07-29 12:23 +0000 [8fb8988fd4]  Mark Michelson <mmichelson@digium.com>

	* Add a test event for inband ringing.

	  This event is necessary for the bridge_wait_e_options test to be able to
	  confirm that ringing is being played on the local channel that runs the
	  BridgeWait() application with the e(r) option.

	  ASTERISK-25292 #close
	  Reported by Kevin Harwell

	  Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e

2015-07-28 05:33 +0000 [1d081ec970]  Mark Duncan <mark@syon.co.jp>

	* res/res_rtp_asterisk: Add ECDH support

	  This will add ECDH support to Asterisk. It will
	  detect auto ECDH support in OpenSSL
	  (1.0.2b and above) during ./configure. If this is
	  available, it will use it,
	  otherwise it will fall back to prime256v1 (this
	  behavior is consistent with
	  other projects such as Apache and nginx).

	  This fixes WebRTC being broken in Firefox 38+ due
	  to Firefox now only supporting
	  ciphers with perfect forward secrecy.

	  ASTERISK-25265 #close

	  Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b

2015-07-16 12:16 +0000 [687597ca8c]  Jonathan Rose <jrose@digium.com>

	* holding_bridge: ensure moh participants get frames

	  Currently, if a blank musiconhold.conf is used, musiconhold will fail
	  to start for a channel going into a holding bridge with an anticipation
	  of getting music on hold. That being the case, no frames will be written
	  to the channel and that can pose a problem for blind transfers in PJSIP
	  which may rely on frames being written to get past the REFER framehook.
	  This patch makes holding bridges start a silence generator if starting
	  music on hold fails and makes it so that if no music on hold functions
	  are installed that the ast_moh_start function will report a failure so
	  that consumers of that function will be able to respond appropriately.

	  ASTERISK-25271 #close

	  Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99
	  (cherry picked from commit 8458b8d441c2f4143ff135163ff3da4f88fe14c8)

2015-07-18 11:16 +0000 [309dd2a409]  Joshua Colp <jcolp@digium.com>

	* pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.

	  This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
	  endpoint options. These allow the channel to be hung up if RTP
	  is not received from the remote endpoint for a specified number of
	  seconds.

	  ASTERISK-25259 #close

	  Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9

2015-07-24 09:46 +0000 [a0c31c7a05]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Add rtp_keepalive to sample config file.

	  Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19

2015-07-23 13:11 +0000 [d97bed46b7]  Mark Michelson <mmichelson@digium.com>

	* Local channels: Alternate solution to ringback problem.

	  Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d solved an issue where a
	  specific scenario involving local channels and a native local RTP bridge
	  could result in ringback still being heard on a calling channel even
	  after the call is bridged.

	  That commit caused many tests in the testsuite to fail with alarming
	  consequences, such as not sending DialBegin and DialEnd events, and
	  giving incorrect hangup causes during calls.

	  This commit reverts the previous commit and implements and alternate
	  solution. This new solution involves only passing AST_CONTROL_RINGING
	  frames across local channels if the local channel is in AST_STATE_RING.
	  Otherwise, the frame does not traverse the local channels. By doing
	  this, we can ensure that a playtones generator does not get started on
	  the calling channel but rather is started on the local channel on which
	  the ringing frame was initially indicated.

	  ASTERISK-25250 #close
	  Reported by Etienne Lessard

	  Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39

2015-07-22 12:24 +0000 [1cc99ba8b6]  Joshua Colp <jcolp@digium.com>

	* audiohook: Use manipulated frame instead of dropping it.

	  Previous changes to sample rate support in audiohooks accidentally
	  removed code responsible for allowing the manipulate audiohooks
	  to work. Without this code the manipulated frame would be dropped
	  and not used. This change restores it.

	  ASTERISK-25253 #close

	  Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13

2015-07-22 09:46 +0000 [0b7148e262]  Mark Michelson <mmichelson@digium.com>

	* Local channels: Do not block control -1 payloads.

	  Control frames with a -1 payload are used as a special signal to stop
	  playtones generators on channels. This indication is sent both by
	  app_dial as well as by ast_answer() when a call is answered in case any
	  tones were being generated on a calling channel.

	  This control frame type was made to stop traversing local channel pairs
	  as an optimization, because it was thought that it was unnecessary to
	  send these indications, and allowing such unnecessary control frames to
	  traverse the local channels would cause the local channels to optimize
	  away less quickly.

	  As it turns out, through some special magic dialplan code, it is
	  possible to have a tones being played on a non-local channel, and it is
	  important for the local channel to convey that the tones should be
	  stopped. The result of having tones continue to be played on the
	  non-local channel is that the tones play even once the channel has been
	  bridged. By not blocking the -1 control frame type, we can ensure that
	  this situation does not happen.

	  ASTERISK-25250 #close
	  Reported by Etienne Lessard

	  Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815

2015-07-22 05:16 +0000 [e5fe8d40c8]  Joshua Colp <jcolp@digium.com>

	* audiohook: Read the correct number of samples based on audiohook format.

	  Due to changes in audiohooks to support different sample rates the
	  underlying storage of samples is in the format of the audiohook
	  itself and not of the format being requested. This means that if a
	  channel is using G722 the samples stored will be at 16kHz. If
	  something subsequently reads from the audiohook at a format which
	  is not the same sample rate as the audiohook the number of samples
	  needs to be adjusted.

	  Given the following example:
	  1. Channel writing into audiohook at 16kHz (as it is using G722).
	  2. Chanspy reading from audiohook at 8kHz.

	  The original code would read 160 samples from the audiohook for
	  each 20ms of audio. This is incorrect. Since the audio in the
	  audiohook is at 16kHz the actual number needing to be read is 320.
	  Failure to read this much would cause the audiohook to reset
	  itself constantly as the buffer became full.

	  This change adjusts the requested number of samples by determining
	  the duration of audio requested and then calculating how many
	  samples that would be in the audiohook format.

	  ASTERISK-25247 #close

	  Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d

2015-07-20 15:59 +0000 [293c9f6894]  Elazar Broad <elazar@thebroadfamily.com>

	* cdr/cdr_adaptive_odbc.c: Fix quoted identifier usage when inserting CDR records

	  Commit a24ce38 added support for the use of quoted indentifiers when inserting
	  CDR records into the database. However, the if statement logic responsible for
	  determining whether to use those identifiers is reversed, resulting in a
	  reference to the quoted identifier character buffer which will be null, hence
	  null terminating the SQL query, resulting in a truncated statement which
	  fails to execute.

	  ASTERISK-25263 #close
	  Reported by: Elazar Broad
	  Tested by: Elazar Broad

	  Change-Id: I40da47309b67cc1572207b1515dcc08ec9b1f644
2015-07-20 12:39 +0000 [d02196448b]  Rusty Newton <rnewton@digium.com>

	* Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c

	   * In sip.conf.sample fix sentence where we said that WS or WSS are supported
	     transports for use in an outbound register definition. They are not
	     supported in that case.
	   * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used
	     to enable CDR on a channel.

	  ASTERISK-24867 #close
	  Reported by: Rusty Newton

	  ASTERISK-24853 #close
	  Reported by: PSDK

	  Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca

2015-07-09 14:17 +0000 [2b42264e66]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Add rtp_keepalive endpoint option.

	  This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
	  chan_sip option, this specifies an interval, in seconds, at which we
	  will send RTP comfort noise frames. This can be useful for keeping RTP
	  sessions alive as well as keeping NAT associations alive during lulls.

	  ASTERISK-25242 #close
	  Reported by Mark Michelson

	  Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d

2015-07-16 09:13 +0000 [8b503f2a10]  Michael Cargile <mikec@vicidial.com>

	* res/res_musiconhold: Add a warning when MOH does not exist

	  Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b

2015-07-19 09:11 +0000 [9475dc9492]  Matt Jordan <mjordan@digium.com>

	* res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf

	  Misconfiguring sorcery.conf with a 'config' wizard with no extra data
	  will currently crash Asterisk on startup, as the wizard requires a comma
	  delineated list to parse. This patch updates res_sorcery_config to check
	  for the presence of the data before it starts manipulating it.

	  Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847

2015-07-16 09:46 +0000 [649460aa44]  Joshua Colp <jcolp@digium.com>

	* chan_pjsip: Don't change formats when frame of unsupported format is received.

	  Receipt of an RTP packet currently causes the formats on an PJSIP channel to
	  change to the format of the RTP packet. In some off-nominal cases it's possible
	  for this to be a format that has not been configured or negotiated. This change
	  makes it so only formats explicitly configured on the endpoint are allowed.

	  ASTERISK-25258 #close

	  Change-Id: If93d641fb6418a285928839300d7854cab8c1020

2015-07-14 16:55 +0000 [4a875e8082]  Richard Mudgett <rmudgett@digium.com>

	* pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable.

	  ASTERISK-25256 #close
	  Reported by: Richard Mudgett

	  Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3

2015-07-17 04:59 +0000 [7908ae4934]  Patric Marschall <patric.marschall@1und1.de>

	* sig_pri.h: force_restart_unavailable_chans in wrong scope

	  In channels/sig_pri.h, struct sig_pri_span, the field
	  force_restart_unavailable_chans is only defined if

	  #if defined(HAVE_PRI_MCID) is true.

	  All other occurences of force_restart_unavailable_chans are outside of the

	  #if defined(HAVE_PRI_MCID)
	  endif

	  scope.

	  ASTERISK-25257 #close
	  Reported by: Patric Marschall

	  Change-Id: I071de89cc2cd0d85927a013036e235851f672549

2015-07-08 16:39 +0000 [254d07b15b]  Matt Jordan <mjordan@digium.com>

	* ARI: Add support for push configuration of dynamic object

	  This patch adds support for push configuration of dynamic, i.e.,
	  sorcery, objects in Asterisk. It adds three new REST API calls to the
	  'asterisk' resource:
	   * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current
	     object given its ID. This returns back a list of ConfigTuples, which
	     define the fields and their present values that make up the object.
	   * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an
	     object. A body may be passed with the request that contains fields to
	     populate in the object. The same format as what is retrieved using
	     the GET operation is used for the body, save that we specify that the
	     list of fields to update are contained in the "fields" attribute.
	   * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic
	     object from its backing storage.

	  Note that the success/failure of these operations is somewhat
	  configuration dependent, i.e., you must be using a sorcery wizard that
	  supports the operation in question. If a sorcery wizard does not support
	  the create or delete mechanisms, then the REST API call will fail with a
	  403 forbidden.

	  ASTERISK-25238 #close

	  Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c

2015-07-15 15:40 +0000 [b34c4528ab]  Richard Mudgett <rmudgett@digium.com>

	* strings.h: Fix issues with escape string functions.

	  Fixes for issues with the ASTERISK-24934 patch.

	  * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is
	  an empty string.  If it were an empty string the functions returned NULL
	  as if there were a memory allocation failure.  This failure caused the AMI
	  VarSet event to not get posted if the new value was an empty string.

	  * Fixed dest buffer overwrite potential in ast_escape() and
	  ast_escape_c().  If the dest buffer size is smaller than the space needed
	  by the escaped s parameter string then the dest buffer would be written
	  beyond the end by the nul string terminator.  The num parameter was really
	  the dest buffer size parameter so I renamed it to size.

	  * Made nul terminate the dest buffer if the source string parameter s was
	  an empty string in ast_escape() and ast_escape_c().

	  * Updated ast_escape() and ast_escape_c() doxygen function description
	  comments to reflect reality.

	  * Added some more unit test cases to /main/strings/escape to cover the
	  empty source string issues.

	  ASTERISK-25255 #close
	  Reported by: Richard Mudgett

	  Change-Id: Id77fc704600ebcce81615c1200296f74de254104

2015-07-14 14:29 +0000 [097c15ac51]  Richard Mudgett <rmudgett@digium.com>

	* parking_applications.c: Fix ast_verb() line terminator.

	  Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775

2015-07-14 14:36 +0000 [8b620c555b]  Richard Mudgett <rmudgett@digium.com>

	* res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park.

	  setup_park_common_datastore() was assuming that a non-NULL string returned
	  for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty
	  strings.  Things got crashy as a result.

	  * Made setup_park_common_datastore() treat the channel variable values the
	  same whether they are NULL or empty for ATTENDEDTRANSFER and
	  BLINDTRANSFER.

	  ASTERISK-25254 #close
	  Reported by: Richard Mudgett

	  Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2

2015-07-10 18:01 +0000 [4af24ec74b]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer().

	  Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb

2015-07-10 10:42 +0000 [71b3bcf5e0]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Add some helpful comments and minor tweaks.

	  Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743

2015-07-10 10:43 +0000 [53c91737a5]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Fix off nominal crash potential in debug message.

	  Change-Id: I09928297927ee85f7655289acee3a586816466bc

2015-07-15 10:31 +0000 [eff6a88a88]  Matt Jordan <mjordan@digium.com>

	* apps/app_dictate: Fix typo in attribution

	  Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian
	  (GameGamer43) for pointing that out.

	  Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106

2015-07-15 10:28 +0000 [e01d93e092]  Benjamin Ford <bford@digium.com>

	* ARI: Fixed unload mode for unload module.

	  Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM,
	  which would unload a module even if it was in use.

	  * Changed unload mode to proper mode

	  ASTERISK-25173

	  Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533

2015-07-10 18:17 +0000 [1b666549f3]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session.c: Fix crash on call disconnect.

	  The crash fix for ASTERISK-25183 backported some code from master to try
	  to make sure that a BYE response is processed by the same serializer used
	  by the BYE request.  The identified race condition causing that backport
	  was the BYE request code had not finished processing after sending the BYE
	  before the BYE response came in for processing under a different thread.
	  Unfortunately, there is still a race condition.  Now the race condition is
	  between destroying the call session's serializer in
	  ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a
	  reference to the serializer for a BYE response.  Even worse, the new race
	  condition is a design limitation of the taskprocessor implementation that
	  didn't matter in versions before v12.  Back then, taskprocessors were only
	  destroyed when a module unloaded.  Now res_pjsip can destroy them when a
	  call ends.

	  However, as noted on the ASTERISK-25183 commit,
	  session_inv_on_state_changed() is disassociating the dialog from the
	  session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED.
	  This is a tad too soon because our BYE request transaction has not
	  completed yet.

	  * Split session_end() that is called by session_inv_on_state_changed() to
	  hold off session destruction until the BYE transaction timeout occurs or a
	  failed initial INVITE transaction timeout occurs in
	  session_inv_on_tsx_state_changed().

	  ASTERISK-25201 #close
	  Reported by: Matt Jordan

	  Change-Id: Iaf8dc8485fd8392a2a3ee4ad3b7f7f04a0dcc961

2015-07-14 13:12 +0000 [9d458b8311]  Benjamin Ford <bford@digium.com>

	* ARI: Added new functionality to reload a single module.

	  An http request can be sent to reload an Asterisk module. If the
	  module can not be reloaded or is not already loaded, an error
	  response will be returned.

	  The command "curl -v -u user:pass -X PUT 'http://localhost:8088
	  /ari/asterisk/modules/{moduleName}'" (or something similar, based
	  on configuration) can be run in the terminal to access this new
	  functionality.

	  For more information, see:
	  https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource

	  * Added new ARI functionality
	  * Asterisk modules can be reloaded through http requests

	  ASTERISK-25173

	  Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1

2015-07-14 08:55 +0000 [f64f1c2772]  Benjamin Ford <bford@digium.com>

	* ARI: Added new functionality to unload a single module.

	  An http request can be sent to unload an Asterisk module. If the
	  module can not be unloaded or is already unloaded, an error response
	  will be returned.

	  The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
	  /ari/asterisk/modules/{moduleName}'" (or something similar, depending
	  on configuration) can be run in the terminal to access this new
	  functionality.

	  For more information, see:
	  https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource

	  * Added new ARI functionality
	  * Asterisk modules can be unloaded through http requests

	  ASTERISK-25173

	  Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57

2015-07-13 16:00 +0000 [aa5707b889]  Benjamin Ford <bford@digium.com>

	* ARI: Added new functionality to load a single module.

	  An http request can be sent to load an Asterisk module. If the
	  module can not be loaded or is loaded already, an error response
	  will be returned.

	  The command curl -v -u user:pass -X POST 'http://localhost:8088/ari
	  /asterisk/modules/{moduleName}'" (or something similar, depending on
	  configuration) can be run in the terminal to access this new
	  functionality.

	  For more information, see:
	  https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource

	  * Added new ARI functionality
	  * Asterisk modules can be loaded through http requests

	  ASTERISK-25173

	  Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33

2015-07-13 10:54 +0000 [6a764db370]  Benjamin Ford <bford@digium.com>

	* ARI: Added new functionality to get information on a single module.

	  An http request can be sent to retrieve information on a single
	  module, including the resource name, description, use count, status,
	  and support level.

	  The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
	  /asterisk/modules/{moduleName}'" (or something similar, depending on
	  configuration) can be run in the terminal to access this new
	  functionality.

	  For more information, see:
	  https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource

	  * Added new ARI functionality
	  * Information on a single module can now be retrieved

	  ASTERISK-25173

	  Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463

2015-07-08 14:56 +0000 [c855523519]  Kevin Harwell <kharwell@digium.com>

	* bridge.c: Fixed race condition during attended transfer

	  During an attended transfer a thread is started that handles imparting the
	  bridge channel. From the start of the thread to when the bridge channel is
	  ready exists a gap that can potentially cause problems (for instance, the
	  channel being swapped is hung up before the replacement channel enters the
	  bridge thus stopping the transfer). This patch adds a condition that waits
	  for the impart thread to get to a point of acceptable readiness before
	  allowing the initiating thread to continue.

	  ASTERISK-24782
	  Reported by: John Bigelow

	  Change-Id: I08fe33a2560da924e676df55b181e46fca604577

2015-05-13 16:22 +0000 [ef82190804]  Matt Jordan <mjordan@digium.com>

	* media cache: Add CLI commands

	  This patch adds five CLI commands for the media cache:
	   * 'media cache show all' - display a summary of all items in the media
	     cache.
	   * 'media cache show <uri>' - display detailed information about a
	     single item in the media cache.
	   * 'media cache delete <uri>' - remove an item from the media cache, and
	     inform the bucket backend for the URI scheme to remove the item as
	     well.
	   * 'media cache refresh <uri>' - refresh a URI. If the item does not
	     exist in the media cache, the bucket backend will pull down the media
	     associated with the URI and create the item in the cache.
	   * 'media cache create <uri>' - create an item in the media cache from
	     some local media storage. Note that the bucket backend for the URI
	     scheme must still permit the item creation.

	  Change-Id: Id1c5707a3b8e2d96b56e4691a46a936cd171f4ae

2015-01-29 08:38 +0000 [3ea0d38396]  Matt Jordan <mjordan@digium.com>

	* media cache: Add a core API and facade for a backend agnostic media cache

	  This patch adds a new API to the Asterisk core that acts as a media
	  cache. The core API itself is mostly a thin wrapper around some bucket
	  API provided implementation that itself acts as the mechanism of
	  retrieval for media. The media cache API in the core provides the
	  following:
	   * A very thin in-memory cache of the active bucket_file items. Unlike a
	     more traditional cache, it provides no expiration mechanisms. Most
	     queries that hit the in-memory cache will also call into the bucket
	     implementations as well. The bucket implementations are responsible
	     for determining whether or not the active record is active and valid.
	     This makes sense for the most likely implementation of a media cache
	     backend, i.e., HTTP. The HTTP layer itself is the actual arbiter of
	     whether or not a record is truly active; as such, the in-memory cache
	     in the core has to defer to it.
	   * The ability to create new items in the media cache from local
	     resources. This allows for re-creation of items in the cache on
	     restart.
	   * Synchronization of items in the media cache to the AstDB. This
	     also includes various pieces of important metadata.

	  The API provides sufficient access that higher level APIs, such as the
	  file or app APIs, do not have to worry about the semantics of the bucket
	  APIs when needing to playback a resource.

	  In addition, this patch provides unit tests for the media cache API. The
	  unit tests use a fake bucket backend to verify correctness.

	  Change-Id: I11227abbf14d8929eeb140ddd101dd5c3820391e

2015-07-11 20:25 +0000 [887945d410]  Matt Jordan <mjordan@digium.com>

	* main/bucket: Add a callback function for ast_bucket_file objects

	  This patch adds a new function to the bucket API for ast_bucket_file
	  objects, ast_bucket_file_metadata_callback. It will call ao2_callback on
	  the ast_bucket_file's ao2_container of metadata, calling the provided
	  ao2_callback_fn callback on each piece of metadata associated with the
	  file.

	  This is particularly useful when a bucket backend has added metadata,
	  and a higher level API wants to be aware of/access said metadata,
	  without knowing for sure what the key is.

	  Change-Id: I96f6757717f47b650df91a437f7df16406227466

2015-07-08 16:28 +0000 [458715d088]  Matt Jordan <mjordan@digium.com>

	* main/sorcery: Don't fail object set creation from JSON if field fails

	  Some individual fields may fail their conversion due to their default
	  values being invalid for their custom handlers. In particular,
	  configuration values that depend on others being enabled (and thus have
	  an empty default value) are notorious for tripping this routine up. An
	  example of this are any of the DTLS options for endpoints. Any of the
	  DTLS options will fail to be applied (as DTLS is not enabled), causing
	  the entire object set to be aborted.

	  This patch makes it so that we log a debug message when skipping a
	  field, and rumble on anyway.

	  ASTERISK-25238

	  Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76

2015-07-08 16:21 +0000 [6ed58014f5]  Matt Jordan <mjordan@digium.com>

	* main/format_cap: Parse capabilities generated by ast_format_cap_get_names

	  We have a strange relationship between the parsing of format
	  capabilities from a string and their representation as a string. We
	  expect the format capabilities to be expressed as a string in the
	  following format:

	  allow = !all,ulaw,alaw
	  disallow = g722

	  While we would generate the string representation of those formats as:

	  allow = (ulaw|alaw)
	  disallow = (ulaw|alaw|g729...)

	  When the configuration framework needs to store values as a string, it
	  generates the format capabilities using the second representation; this
	  representation however cannot be parsed when the entry is rehydrated.
	  This patch fixes that by updating
	  ast_format_cap_update_by_allow_disallow to parse an entry as if it were
	  in the generated format if it has a leading '(' and a trailing ')'.

	  ASTERISK-25238

	  Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca

2015-07-08 16:38 +0000 [e64e586900]  Matt Jordan <mjordan@digium.com>

	* res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails

	  Having a debug message tell us that we attempted to look up an item but
	  failed is nice in circumstances when it isn't clear if the wizard was
	  queried correctly or not.

	  Change-Id: I2600c3bbea87f252196358f62e73f4c7da8632f7

2015-07-08 16:37 +0000 [7c14dfdc61]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_outbound_registration: Fix WARNING message

	  Newlines are nice.

	  Change-Id: Icf0d915db02882e47cd9077ed9009f5d44140d42

2015-07-08 16:35 +0000 [3e286e6b51]  Matt Jordan <mjordan@digium.com>

	* res_pjsip/configuration: Fix a variety of default value problems

	  This patch fixes some bad default value handling in the following
	  settings:

	  * The 'message_context' and 'accountcode' settings are not mandatory. As
	    such, we can allow their stringfield values to be empty.
	  * The 'media_encryption' setting applies a default value of 'none' to
	    the setting, which it then can't parse or understand. Since the value
	    is documented to be 'no', this will now apply that as the default
	    value.

	  Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83

2015-07-08 16:32 +0000 [ffadb5f1de]  Matt Jordan <mjordan@digium.com>

	* main/sorcery: Provide log messages when a wizard does not support an operation

	  If a sorcery wizard does not support one of the 'optional' CRUD
	  operations (namely the CUD), log a WARNING message so we are aware of
	  why the operation failed. This also removes an assert in this case, as
	  the CUD operation may have been triggered by an external system, in
	  which case it is not a programming error but a configuration error.

	  Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53

2015-06-27 17:53 +0000 [5266796432]  Matt Jordan <mjordan@digium.com>

	* tests/test_devicestate: Add additional tests for the device state API

	  This patch adds more tests that exercise the device state API. This includes:

	  * Tests that cover adding a device state provider, as well as deleting a
	    device state provider. This also verifies that you cannot add an
	    already added device state provider, and cannot delete an already
	    deleted device state provider.
	  * A test that covers changing device state and receiving said updates
	    from a device state subscriber. This also covers hitting both the
	    device state cache as well as a custom device state provider.
	  * A test that covers converting device state to channel state and device
	    state values to a string representation and back.
	  * A test that covers obtaining device state from an active channel and a
	    channel driver that provides its own device state.

	  Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d

2015-06-27 17:51 +0000 [f77e688f20]  Matt Jordan <mjordan@digium.com>

	* main/devicestate: Prevent duplicate registration of device state providers

	  Currently, the device state provider API will allow you to register a
	  device state provider with the same case insensitive name more than
	  once. This could cause strange issues, as the duplicate device state
	  providers will not be queried when a device's state has to be polled.
	  This patch updates the API such that a device state provider with the
	  same name as one that has already registered will be rejected.

	  Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2

2015-06-26 10:57 +0000 [1b7760a8aa]  Benjamin Ford <bford@digium.com>

	* ARI: Added new functionality to get all module information.

	  An http request can be sent to retrieve a list of all existing modules,
	  including the resource name, description, use count, status, and
	  support level.

	  The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
	  asterisk/modules" (or something similar, depending on configuration)
	  can be run in the terminal to access this new functionality.

	  For more information, see:
	  https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource

	  * Added new ARI functionality
	  * Information on modules can now be retrieved

	  Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0

2015-07-09 09:18 +0000 [4a25d55416]  Joshua Colp <jcolp@digium.com>

	* bridge_native_rtp.c: Don't start native RTP bridging after attended transfer.

	  The bridge_native_rtp module adds a frame hook to channels which are in
	  a native RTP bridge. This frame hook is used to intercept when a hold
	  or unhold frame traverses the bridge so native RTP can be stopped or
	  started as appropriate. This is expected but exposes a specific bug
	  when attended transfers are involved.

	  Upon completion of an attended transfer an unhold frame is queued up
	  to take one of the channels involved off hold. After this is done
	  the channel is moved between bridges.

	  When the frame hook is involved in this case for the unhold it
	  releases the channel lock and acquires the bridge lock. This
	  allows the bridge core to step in and move the channel
	  (potentially changing the bridging techology) from another thread.
	  Once completed the bridge lock is released by the bridge core.
	  The frame hook is then able to acquire the bridge lock and
	  wrongfully starts native RTP again, despite the channel no longer
	  being in the bridge or needing to start native RTP. In fact at
	  this point the frame hook is no longer attached to the channel.

	  This change makes it so the native RTP bridge data is available to
	  the frame hook when it is invoked. Whether the frame hook has
	  been detached or not is stored on the native RTP bridge data and
	  is checked by the frame hook before starting or stopping native
	  RTP bridging. If the frame hook has been detached it does nothing.

	  ASTERISK-25240 #close

	  Change-Id: I13a73186a05f4e5a764f81e5cd0ccec1ed1891d2

2015-07-08 04:21 +0000 [9276415f65]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used.

	  This change fixes a bug where the DTLS timeout timer would be
	  initialized to 0 if DTLS was not used for an RTP session.

	  ASTERISK-25103

	  Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac

2015-07-07 15:03 +0000 [3cdfd39af7]  Ashley Sanders <asanders@digium.com>

	* DNS: Create a system-level DNS resolver

	  Prior to this patch, the DNS core present in master had no default system-level
	  resolver implementation. Therefore, it was not possible for the DNS core to
	  perform resolutions unless the libunbound library was installed and the
	  res_resolver_unbound module was loaded.

	  This patch introduces a system-level DNS resolver implementation that will
	  register itself with the lowest consideration priority available (to ensure
	  that it is to be used only as a last resort). The resolver relies on low-level
	  DNS search functions to perform a rudimentary DNS search based on a provided
	  query and then supplies the search results to the DNS core.

	  ASTERISK-25146 #close
	  Reported By: Joshua Colp

	  Change-Id: I3b36ea17b889a98df4f8d80d50bb7ee175afa077

2015-07-01 07:55 +0000 [5717340ab3]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.

	  This change moves logic for setting up the DTLS SSL contexts to
	  when the SDP is done being processed instead of when ICE negotiation
	  completes. It also stops handshakes from being initiated when we
	  are acting as a server.

	  Manipulating the SSL context when ICE negotiation has completed
	  is problematic as the SSL context is not protected and if acting
	  as a client the remote side may have started DTLS negotiation
	  already.

	  The retransmission timeout timer code has also been split up
	  and simplified some. Both RTP and RTCP now have their own timers
	  and the points at which the timer is stopped and started is now
	  more specific. When a packet is sent the timer is started. When
	  a response is received but before it is processed the timer is
	  stopped. This provides a guarantee that the timeout is not
	  occurring while the response is processed.

	  ASTERISK-22805 #close
	  ASTERISK-24550 #close
	  ASTERISK-24651 #close
	  ASTERISK-24832 #close
	  ASTERISK-25103 #close
	  ASTERISK-25127 #close

	  Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91

2015-06-26 18:48 +0000 [189841ddb7]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_mwi.c: Fix MWI subscription memory corruption crash.

	  MWI subscriptions can crash or corrupt memory when using the subscription
	  datastore to access the MWI subscription object because the datastore is
	  not holding a reference to the object.

	  * Give the subscription datastore a ref to the MWI subscription object.
	  It is unfortunate that the ref causes a circular ref chain that must be
	  explicitly broken to allow the memory to get released.  The loop is broken
	  when the subscription is shutdown and if the subscription setup fails.

	  ASTERISK-25168 #close
	  Reported by: Carl Fortin

	  Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f

2015-07-02 14:51 +0000 [7cd99be534]  Richard Mudgett <rmudgett@digium.com>

	* PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error.

	  When res_pjsip body generator modules were generating XML or XPIDF
	  response bodies, there was a chance that the generated body would be the
	  exact size of the supplied buffer.  Adding the nul string terminator would
	  then write beyond the end of the buffer and potentially corrupt memory.

	  * Fix MALLOC_DEBUG high fence violations caused by adding a nul string
	  terminator on the end of a buffer for XML or XPIDF response bodies.

	  * Made calls to pj_xml_print() safer if the XML prolog is requested.  Due
	  to a bug in pjproject, the return value could be -1 _or_
	  AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough.

	  * Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the
	  return value of pj_xml_print() when the supplied buffer is not large
	  enough.

	  ASTERISK-25168
	  Reported by: Carl Fortin

	  Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de

2015-06-26 10:36 +0000 [792ed7ce93]  Richard Mudgett <rmudgett@digium.com>

	* PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences.

	  When a caller calls a FAX number and then hangs up right after the call is
	  answered then the T.38 re-INVITE automatic reject timer may still be
	  running after the channel goes away.

	  * Added session NULL channel checks on the code paths that get executed by
	  t38_automatic_reject() to prevent a crash when the T.38 re-INVITE
	  automatic reject timer expires.

	  ASTERISK-25168
	  Reported by: Carl Fortin

	  Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403

2015-06-30 11:17 +0000 [030e8339dd]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_mwi.c: Use safer loop coding in mwi_subscription_mailboxes_str().

	  Change-Id: I6f39d809a6d1b47b35bb32b298f5a12f35d6f907

2015-06-30 11:14 +0000 [453d7b8d69]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_mwi.c: Eliminate a simple RAII_VAR.

	  Change-Id: Ib1843f81e826a6c760c424c88eb70c350d9d61da

2015-06-30 11:11 +0000 [786c6d42ef]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_mwi.c: Fix mid-line log message line breaks.

	  * Add create_mwi_subscriptions_for_endpoint() doxygen comment.

	  Change-Id: I3c3f921f4ec749fb65b62d2f6fa0d4d1888b94e2

2015-06-26 16:10 +0000 [1b91094edd]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_t38.c: Fix always false if test.

	  Calling t38_change_state() sets the t38 state so it makes little sense to
	  then check the state right after the call for something else.

	  * Made the code in t38_interpret_parameters() reject or exit T.38 mode as
	  intended but not implemented.

	  Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2

2015-06-30 15:19 +0000 [74135c8efa]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: Failover when server is not available

	  Previously Asterisk did not properly failover to the next resolved DNS
	  address when a endpoint could not be reached. With this patch, and while
	  using res_pjsip, SIP requests (both in/out of dialog) now attempt to use
	  the next address in the list of resolved addresses until a proper response
	  is received or no more addresses are left.

	  ASTERISK-25076 #close
	  Reported by: Joshua Colp

	  Change-Id: Ief14f4ebd82474881f72f4538f4577f30af2a764

2015-07-06 09:24 +0000 [38a3c27a09]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Execute stale unit test last.

	  In Jenkins there is currently a sporadic test failure of a
	  variable number of sorcery memory cache unit tests. I have not
	  been able to reproduce this on the build agents themselves or
	  on my development machine.

	  My working theory is that the stale unit test is causing a
	  sorcery instance to persist longer than expected, causing subsequent
	  tests to fail when setting up and initializing the next
	  sorcery instance.

	  To see if this is the case this change moves the stale unit test
	  to execute last so no subsequent unit tests can have issues
	  initializing their sorcery instance.

	  Change-Id: Ifd6550a949613be774b75fa5db12c02110f82c4a

2015-06-20 13:54 +0000 [ef8d3f6506]  Matt Jordan <mjordan@digium.com>

	* bucket: Add clone/staleness operations for ast_bucket/ast_bucket_file

	  This patch enhances the bucket API in two ways.

	  First, since ast_bucket and ast_bucket_file instances are immutable, a 'clone'
	  operation has been added that provides a 'clone' of an existing
	  ast_bucket/ast_bucket_file object. Note that this makes use of the
	  ast_sorcery_copy operation, along with the copy callback handler on the
	  "bucket" and "file" object types for the bucket sorcery instance.

	  Second, there is a need for the bucket API to ask a wizard if an object
	  is stale. This is particularly useful with the upcoming media cache
	  enhancements, where we want to ask the backing data storage if the
	  object we are currently operating on has known updates. This patch adds
	  API calls for ast_bucket and ast_bucket_file objects, which callback
	  into their respective sorcery wizards via the sorcery API.

	  Unit tests have also been added to cover the respective
	  ast_bucket/ast_bucket_file clone and staleness operations.

	  Change-Id: Ib0240ba915ece313f1678a085a716021d75d6b4a

2015-07-04 10:03 +0000 [b178f8701b]  Matt Jordan <mjordan@digium.com>

	* sorcery: Add support for object staleness

	  This patch enhances the sorcery API to allow for sorcery wizards to
	  determine if an object is stale. This includes the following:

	  * Sorcery objects now have a timestamp that is set on creation. Since
	    sorcery objects are immutable, this can be used by sorcery wizards to
	    determine if an object is stale.

	  * A new API call has been added, ast_sorcery_is_stale. This API call
	    queries the wizards associated with the object, calling a new callback
	    function 'is_stale'. Note that if a wizard does not support the new
	    callback, objects are always assumed to not be stale.

	  * Unit tests have been added that cover the new API call.

	  Change-Id: Ica93c6a4e8a06c0376ea43e00cf702920b806064

2015-07-04 18:22 +0000 [f35a4b8525]  Joshua Colp <jcolp@digium.com>

	* res/res_http_websocket: Don't send HTTP response fragmented.

	  This change makes it so that when accepting a WebSocket
	  connection the HTTP response is sent as one packet instead of
	  fragmented. Browsers don't like it when you send it fragmented.

	  ASTERISK-25103

	  Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69

2015-06-27 18:47 +0000 [2c17515f3c]  Matt Jordan <mjordan@digium.com>

	* Makefile: Remove coverage files on 'make clean'

	  This patch updates a variety of Makefiles in Asterisk's build system to
	  remove .gcda and .gcno files when 'make clean' is executed. These files
	  are generated when '--enable-coverage' is passed to the Asterisk
	  configure script.

	  Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602

2015-07-02 09:08 +0000 [34323f9f95]  Walter Doekes <walter+asterisk@wjd.nu>

	* chan_sip: Fix early call pickup channel leak.

	  When handle_invite_replaces() was called, and either ast_bridge_impart()
	  failed or there was no bridge (because the channel we're picking up was
	  still ringing), chan_sip would leak a channel.

	  Thanks Matt and Corey for checking the bridge path.

	  ASTERISK-25226 #close

	  Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8

2015-07-01 16:04 +0000 [ef74ccb18d]  Matt Jordan <mjordan@digium.com>

	* sorcery/realtime: Add a bit of debug and warning messages for bad configs

	  When a mapping does not exist between a sorcery.conf defined object and
	  a realtime mapping in extconf, currently, the user will receive a slew
	  of ERROR messages that don't really tell what is happening. Some ERROR
	  messages may even be misleading, as they occur after the sorcery API has
	  already given up on the attempt to load and create the sorcery object.

	  This patch adds a bit of debug and a useful WARNING message for when a
	  wizard's open callback fails for a particular object type. In the bad
	  configurations that resulted in this patch, this provided a 'root cause'
	  WARNING message that pointed in the right direction of the configuration
	  problem.

	  Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b

2015-07-02 06:54 +0000 [f18436642b]  Joshua Colp <jcolp@digium.com>

	* dns: Fix crash when invoking cancel in DNS recurring unit test.

	  The recurring unit test expects the user data on a DNS query
	  created as a result of a recurring DNS query to be the recurring
	  structure itself. This is true, mostly. When invoking the user
	  provided callback this user data is changed to the user provided
	  data. This presents a race condition where the data may or may
	  not point to the recurring data.

	  This change simplifies the callback of the user provided callback
	  by creating a new query and populating it with the expected values.
	  This leaves the recurring DNS query alone and fixes the race
	  condition. This is more in line with how the API should be used
	  overall.

	  ASTERISK-25222 #close

	  Change-Id: I10fb6deec025dff097157e7ec17e6e4921778478

2015-07-02 06:19 +0000 [6fbb58c7f7]  Walter Doekes <walter+asterisk@wjd.nu>

	* chan_mgcp: Don't call close on fd -1.

	  ASTERISK-25220 #close

	  Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3

2015-07-02 06:10 +0000 [13a318bbb1]  Walter Doekes <walter+asterisk@wjd.nu>

	* rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format.

	  When running valgrind on Asterisk, it complained about:

	      ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304)
	      ==32423==    at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...)
	      ==32423==    by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292)
	      ==32423==    by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437)

	  The code in question is a struct assignment, which may be performed by
	  memcpy as a compiler optimization. It is changed to only copy the struct
	  contents if source and destination are different.

	  ASTERISK-25219 #close

	  Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a

2015-07-02 05:16 +0000 [40274e3652]  Walter Doekes <walter+asterisk@wjd.nu>

	* astfd: Fix buffer overflow in DEBUG_FD_LEAKS.

	  If DEBUG_FD_LEAKS was used and more file descriptors than the default of
	  1024 were available, some DEBUG_FD_LEAKS-patched functions would
	  overwrite memory past the fixed-size (1024) fdleaks buffer.

	  This change:
	  - adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe
	  - consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024
	  - stores pointers to constants instead of copying the contents
	  - reorders the fdleaks struct for possibly tighter packing
	  - adds a tiny bit of documentation

	  ASTERISK-25212 #close

	  Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5

2015-07-02 04:57 +0000 [3fab8212e3]  Walter Doekes <walter+asterisk@wjd.nu>

	* res_timing: Don't close FD 0 when out of open files.

	  This fixes so a failure to get a timer file descriptor does not cascade
	  to closing FD 0.

	  On error, both res_timing_kqueue and res_timing_timerfd would call the
	  destructor before setting the file handle. The file handle had been
	  initialized to 0, causing FD 0 to be closed. This in turn, resulted in
	  floods of "CLI>" messages and an unusable terminal.

	  ASTERISK-19277 #close
	  Reported by: Barry Chern

	  For the master branch, this was already fixed. This patch only ensures
	  that we do not attempt to close a negative file descriptor.

	  Change-Id: I147d7e33726c6e5a2751928d56561494f5800350

2015-07-01 17:25 +0000 [41610df8d5]  Richard Mudgett <rmudgett@digium.com>

	* chan_vpb.cc: Fix compiler warning Jenkins found.

	  Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0

2015-07-01 13:34 +0000 [537df26f9c]  Scott Griepentrog <scott@griepentrog.com>

	* Channel alert pipe: improve diagnostic error return

	  When a frame is queued on a channel, any failure in
	  ast_channel_alert_write is logged along with errno.

	  This change improves the diagnostic message through
	  aligning the errno value with actual failure cases.

	  ASTERISK-25224
	  Reported by: Andrey Biglari

	  Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b

2015-06-29 12:45 +0000 [58d18324f0]  Mark Michelson <mmichelson@digium.com>

	* res_sorcery_realtime: Fix leak of sorcery object type.

	  This prevents a leak of a sorcery object type when realtime sorcery
	  objects are retrieved by fields or when multiple objects are retrieved.

	  The extent of this leak is that sorcery object types would be leaked.
	  These are allocated whenever an object type is registered with sorcery,
	  meaning that on module shutdown, these objects would be leaked. This
	  could be problematic if many reloads were performed, but it is not as
	  severe as if every sorcery object retrieved from realtime were being
	  leaked.

	  ASTERISK-25165 #close
	  Reported by Corey Farrell

	  Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8
2015-06-26 22:02 +0000 [80d97290bb]  Matt Jordan <mjordan@digium.com>

	* res/res_corosync: Always decline module load, instead of failing

	  Returns a 'failure' from the module load routine indicates to Asterisk
	  that it should abort loading completely. This is rarely - in fact,
	  really, never - a good option. Aborting load of Asterisk from a dynamic
	  module implies that the core, and the rest of the dynamic modules, don't
	  matter: we should abandon all processing.

	  res_corosync is really not that important.

	  This patch updates the module such that, if it fails to load, it
	  politely declines (emitting ERROR messages along the way), and allows
	  Asterisk to continue to function.

	  Note that this issue was keeping Asterisk unit tests from running on
	  certain build agents.

	  Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f

2015-06-26 20:38 +0000 [892cc5625f]  Matt Jordan <mjordan@digium.com>

	* main/pbx: Resolve case sensitivity regression in PBX hints

	  When 8297136f was merged for ASTERISK-25040, a regression was introduced
	  surrounding the case sensitivity of device names within hints.
	  Previously, device names - such as 'sip/foo' - were compared in a case
	  insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After
	  that patch, only the case sensitive name would match, i.e., 'SIP/foo'.
	  As a result, some dialplan hints stopped working.

	  This patch re-introduces case insensitive matching for device names in
	  hints.

	  ASTERISK-25040

	  ASTERISK-25202 #close

	  Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c
	  (cherry picked from commit 96bbcf495a1da9e607d9b04a44b5c4f49e83cc03)

2015-06-26 16:12 +0000 [e18b22a806]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_nat: Adjust when contact should be rewritten.

	  A previous change made the contact only get rewritten if the dialog's
	  route set was not marked frozen. Unfortunately, while the intent of this
	  is correct, the dialog's route set actually gets marked as frozen
	  earlier than expected, especially for UAS dialogs.

	  Instead, the idea is that the contact needs to not be rewritten if there
	  is a pre-existing route set on the dialog. This is now accomplished by
	  checking the dialog's route set list instead of checking if the route
	  set is frozen.

	  Doing this causes some broken tests to begin passing again.

	  ASTERISK-25196
	  Reported by Mark Michelson

	  Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e

2015-06-19 18:27 +0000 [99b1aa6d26]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Add a serializer shutdown group.

	  The client_state objects contain a serializer used to send the outbound
	  REGISTER messages.  Once all those message transactions are complete then
	  the module can shutdown.

	  ASTERISK-24907 #close
	  Reported by: Kevin Harwell

	  Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547

2015-06-26 10:41 +0000 [f536e9b59c]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_refer: Prevent sending duplicate headers.

	  res_pjsip_refer will attempt to add Referred-By or Replaces headers to
	  outbound INVITEs at times. If the INVITE gets challenged for
	  authentication, then we will resend the INVITE. Prior to this patch, the
	  Referred-By or Replaces header would be re-added to the outbound INVITE,
	  resulting in duplicated headers.

	  ASTERISK-25204 #close
	  Reported by Mark Michelson

	  Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d

2015-06-23 14:34 +0000 [c2d48a2a28]  Richard Mudgett <rmudgett@digium.com>

	* AMI: Add Linkedid to the standard channel snapshot AMI event headers.

	  ASTERISK-25189 #close
	  Reported by: John Hardin

	  Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac
2015-06-23 17:43 +0000 [700606a659]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_nat: Rewrite route set when required.

	  When performing some provider testing, the rewrite_contact option was
	  interfering with proper construction of a route set when sending an ACK
	  after receiving a 200 OK response to an INVITE.

	  The initial INVITE was sent to address sip:foo. The 200 OK had a Contact
	  header with URI sip:bar. In addition, the 200 OK had Record-Route
	  headers for sip:baz and sip:foo, in that order. Since the Record-Route
	  headers had the lr parameter, the result should have been:

	  * Set R-URI of the ACK to sip:bar.
	  * Add Route headers for sip:foo and sip:baz, in that order.

	  However, the rewrite_contact option resulted in our rewriting the
	  Contact header on the 200 OK to sip:foo. The result was:

	  * R-URI remained sip:foo.
	  * We added Route headers for sip:foo and sip:baz, in that order.

	  The result was that sip:bar was not indicated in the ACK at all, so the
	  far end never received our ACK. The call eventually dropped.

	  The intention of rewrite_contact is to rewrite the most immediate
	  destination of our SIP request to be the same address on which we
	  received a request or response. In the case of processing a SIP response
	  with Record-Route headers, this means that instead of rewriting the
	  Contact header, we should instead rewrite the bottom-most Record-Route
	  header. In the case of processing a SIP request with Record-Route
	  headers, this means we rewrite the top-most Record-route header.
	  Like when we rewrite the Contact header, we also ensure to update
	  the dialog's route set if it exists.

	  ASTERISK-25196 #close
	  Reported by Mark Michelson

	  Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f
2015-06-19 16:16 +0000 [af4ae3095e]  Richard Mudgett <rmudgett@digium.com>

	* threadpool, res_pjsip: Add serializer group shutdown API calls.

	  A module trying to unload needs to wait for all serializers it creates and
	  uses to complete processing before unloading.

	  ASTERISK-24907
	  Reported by: Kevin Harwell

	  Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059

2015-06-16 15:06 +0000 [4c133d81cd]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs

	  * handle_client_state_destruction() must always be passed a ref to
	  client_state because it will always unref client_state.
	  handle_registration_response() was not passing a client_state ref.

	  * Made the final un-REGISTER message get sent normally using the pjproject
	  register control structure in handle_client_state_destruction().  The
	  previous code attempted to short circuit the response handling for the
	  module to unload.  That doesn't work for a couple reasons.  One,
	  pjsip_regc_send() may call the registered callback before it returns and
	  unbalance the client_state ref count.  Two, the registered callback
	  handles any authentication for the un-REGISTER message.

	  * Made the distinction between internal registration state and external
	  registration status with sip_outbound_registration_status_str().  This is
	  necessary to avoid altering documented AMI messages with internal
	  changes.

	  * Removed references to client_state->client outside of the serializer
	  thread.  When handle_client_state_destruction() destroys the pjproject
	  register control structure that memory is freed and cannot be referenced
	  anymore.  These accesses were to provide information for debug and
	  off-nominal warning messages.

	  * In sip_outbound_registration_timer_cb() you should not access entry->id
	  after unrefing client_state because the passed in entry is normally
	  pointing to the timer entry in the client_state object.

	  ASTERISK-24907
	  Reported by: Kevin Harwell

	  Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f

2015-06-15 15:28 +0000 [dc63377c60]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API

	  The sorcery pjsip 'registration' config object needs to be destroyed on
	  module unload.  Otherwise, a reload of res_pjsip could try to use
	  callbacks for a previously unloaded instance of the module provided by
	  ast_sorcery_object_register() or one of the variants.  Also, if
	  res_pjsip_outbound_registration were subsequently reloaded, the sorcery
	  config field objects would be registered in sorcery twice.

	  ASTERISK-24907
	  Reported by: Kevin Harwell

	  Change-Id: I304fad13dece2604af48353f6c6d9d5c7b064697

2015-06-15 15:28 +0000 [9ec8a0f3cc]  Richard Mudgett <rmudgett@digium.com>

	* sorcery: Add ast_sorcery_object_unregister() API call.

	  Find and unlink the specified sorcery object type to complement
	  ast_sorcery_object_register().  Without this function you cannot
	  completely unload individual modules that use sorcery for configuration.

	  ASTERISK-24907
	  Reported by: Kevin Harwell

	  Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88

2015-06-15 13:38 +0000 [77ff7325a2]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Reorder load_module() and unload_module().

	  It is best if the loading code creates and initializes the module's
	  infrastructure before letting the system know of its existence.  The
	  unloading code needs to reverse the actions of the loading code and in the
	  reverse order.

	  ASTERISK-24907
	  Reported by: Kevin Harwell

	  Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4

2015-06-25 06:42 +0000 [8d6cf667dc]  Joshua Colp <jcolp@digium.com>

	* channel: Remove ignore of answer on non-outgoing channels.

	  Due to the way that channels can now be moved around inside of
	  Asterisk it is possible for the outgoing flag of a channel to get
	  cleared before it has been answered. This results in the bridge
	  not receiving notification that the outgoing leg has been answered.

	  This most easily exhibits itself with DTMF based blond transfers.
	  Since the answer of the outgoing leg is ignored the other party
	  continues to receive both a locally generated ringing and the
	  media stream of the outgoing leg upon its answer. This results
	  in no media being heard.

	  This change removes the ignore of the answer and allows it
	  to pass through.

	  ASTERISK-25171 #close

	  Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e

2015-06-24 14:30 +0000 [daaa551c92]  Richard Mudgett <rmudgett@digium.com>

	* test.c: Add unit test registration checks for summary and description.

	  Added checks when a unit test is registered to see that the summary and
	  description strings do not end with a new-line '\n' for consistency.

	  The check generates a warning message and will cause the
	  /main/test/registrations unit test to fail.

	  * Updated struct ast_test_info member doxygen comments.

	  Change-Id: I295909b6bc013ed9b6882e85c05287082497534d

2015-06-24 16:39 +0000 [71a4d1a033]  Richard Mudgett <rmudgett@digium.com>

	* Unit tests: Fix more unit test description strings.

	  Analyzing the code shows that the unit test summary and description
	  strings should not end with a new-line character.  Where these strings are
	  used in the code a new-line is provided for output.

	  Change-Id: I2f4f37988ec363c8d1c5077a2fc8ca841c5cd30c

2015-06-24 14:39 +0000 [9c6d72e30d]  Richard Mudgett <rmudgett@digium.com>

	* Unit tests: Fix unit test description strings.

	  Analyzing the code shows that the unit test summary and description
	  strings should not end with a new-line character.  Where these strings are
	  used in the code a new-line is provided for output.

	  Change-Id: I129284f5e7ca93d82532334076da4c462d3d9fba

2015-06-24 16:37 +0000 [a0c2d2089d]  Richard Mudgett <rmudgett@digium.com>

	* DNS unit tests: Fix extraneous description string commas.

	  Change-Id: Icf5f13c8e1c2c92a4473bb573ed2dd856ce1b64e

2015-06-23 11:21 +0000 [3b2b004d69]  Joshua Colp <jcolp@digium.com>

	* app_dial: Hold reference to calling channel formats when dialing outbound.

	  Currently when requesting a channel the native formats of the
	  calling channel are provided to the core for usage when dialing
	  the outbound channel. This occurs without holding the channel lock
	  or keeping a reference to the formats. This is problematic as
	  the channel driver may end up changing the formats during this time.
	  In the case of chan_sip this happens when an SDP negotiation
	  completes.

	  This change makes it so app_dial keeps a reference to the native
	  formats of the calling channel which guarantees that they will
	  remain valid for the period of time needed.

	  ASTERISK-25172 #close

	  Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db

2015-06-17 16:23 +0000 [af66b0f3f7]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Add missing line endings to CLI commands

	  Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7

2015-06-12 14:29 +0000 [3f0708e5fe]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage.

	  Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e

2015-06-12 13:33 +0000 [9ceb848242]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Misc code cleanups.

	  * Break some long lines.

	  * Fix doxygen comment.

	  Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305

2015-06-22 15:11 +0000 [44c3c392e3]  Kevin Harwell <kharwell@digium.com>

	* bridge.c: Hangup attended transfer target if bridged

	  After completing an attended transfer the transfer target channel was not being
	  hung up after leaving the bridge. Added an explicit softhangup to hangup said
	  channel, but only if it was previously bridged.

	  ASTERISK-24782 #close
	  Reported by: John Bigelow

	  Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada

2015-06-17 05:04 +0000 [7846f73432]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_mwi: Set up unsolicited MWI upon registration.

	  The res_pjsip_mwi previously required a reload to set up the proper
	  subscriptions to allow unsolicited MWI to work. This change
	  makes it so the act of registering will also cause this to occur.
	  This is particularly useful if realtime is involved as no reload
	  needs to occur within Asterisk to cause the MWI information
	  to get sent.

	  ASTERISK-25180 #close

	  Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252

2015-06-22 13:57 +0000 [096b27d9d2]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Fix whitespace conflict potential.

	  Change-Id: I82e6e388e3688aebe0783f16c9e0800a747584b5

2015-06-22 09:26 +0000 [1ad9a6b6b6]  Alexander Traud (License 6520)

	* chan_sip: Reload peer without its old capabilities.

	  On reload, previously allowed codecs were not removed. Therefore, it was not
	  possible to remove codecs while Asterisk was running. Furthermore, newly added
	  codecs got appended behind the previous codecs. Therefore, it was not possible
	  to add a codec with a priority of #1. This change removes the old capabilities
	  before the current ones are added.

	  ASTERISK-25182 #close
	  Reported by: Alexander Traud
	  patches:
	   asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520)

	  Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802

2015-06-20 19:38 +0000 [5caefc98a1]  Joshua Colp <jcolp@digium.com>

	* chan_sip: Destroy peers without holding peers container lock.

	  Due to the use of stasis_unsubscribe_and_join in the peer destructor
	  it is possible for a deadlock to occur when an event callback is
	  occurring at the same time.

	  This happens because the peer may be destroyed while holding the
	  peers container lock. If this occurs the event callback will never
	  be able to acquire the container lock and the unsubscribe will
	  never complete.

	  This change makes it so the peers that have been removed from the
	  peers container are not destroyed with the container lock held.

	  ASTERISK-25163 #close

	  Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33

2015-06-18 13:16 +0000 [d7a1e84a1e]  Mark Michelson <mmichelson@digium.com>

	* Resolve race conditions involving Stasis bridges.

	  This resolves two observed race conditions.

	  First, a bit of background on what the Stasis application does:

	  1a Creates a stasis_app_control structure. This structure is linked into
	     a global container and can be looked up using a channel's unique ID.
	  2a Puts the channel in an event loop. The event loop can exit either
	     because the stasis_app_control structure has been marked done, or
	     because of some other factor, such as a hangup. In the event loop, the
	     stasis_app_control determines if any specific ARI commands need to be
	     run on the channel and will run them from this thread.
	  3a Checks if the channel is bridged. If the channel is bridged, then
	     ast_bridge_depart() is called since channels that are added to Stasis
	     bridges are always imparted as departable.
	  4a Unlink the stasis_app_control from the container.

	  When an ARI command is received by Asterisk, the following occurs
	  1b A thread is spawned to handle the HTTP request
	  2b The stasis_app_control(s) that corresponds to the channel(s) in the
	     request is/are retrieved. If the stasis_app_control cannot be
	     retrieved, then it is assumed that the channel in question has exited
	     the Stasis app or perhaps was never in Stasis in the first place.
	  3b A command is queued onto the stasis_app_control, and the channel's
	     event loop thread is signaled to run the command.
	  4b While most ARI commands do nothing further, some, such as adding or
	     removing channels from a bridge, will block until the command they
	     issued has been completed by the channel's event loop.

	  The first race condition that is solved by this patch involves a crash
	  that can occur due to faulty detection of the channel's bridged status
	  in step 3a. What can happen is that in step 2a, the event loop may run
	  the ast_bridge_impart() function to asynchronously place the channel
	  into a bridge, then immediately exit the event loop because the channel
	  has hung up. In step 3a, we would detect that the channel was not
	  bridged and would not call ast_bridge_depart(). The reason that the
	  channel did not appear to be bridged was that the depart_thread that is
	  spawned by ast_bridge_impart() had not yet started. That is the thread
	  where the channel is marked as being bridged. Since we did not call
	  ast_bridge_depart(), the Stasis application would exit, and then the
	  channel would be destroyed Then the depart_thread would start up and
	  try to manipulate the destroyed channel, causing a crash.

	  The fix for this is to switch from using ast_channel_is_bridged() to
	  checking the NULLity of ast_channel_internal_bridge_channel() to
	  determine if ast_bridge_depart() needs to be called. The channel's
	  internal bridge_channel is set when ast_bridge_impart() is called and
	  is NULLed by the call to ast_bridge_depart(). If the channel's internal
	  bridge_channel is non-NULL, then the channel must have been imparted
	  into the bridge and needs to be departed, even if the actual bridging
	  operation has not yet started. By departing the channel when necessary,
	  the thread that is running the Stasis application will block until the
	  bridge gives the okay that the depart_thread has exited.

	  The second race condition that is solved by this patch involves a leak
	  of HTTP handler threads. The problem was that step 2b would successfully
	  retrieve a stasis_app_control structure. Then step 2a would exit the
	  channel from the event loop due to a hangup. Steps 3a and 4a would
	  execute, and then finally steps 3b and 4b would. The problem is that at
	  step 4b, when attempting to add a channel to a bridge, the thread would
	  block forever since the channel would never execute the queued command
	  since it was finished with the event loop. This meant that the HTTP
	  handling thread would be leaked, along with any references that thread
	  may have owned (in my case, I was seeing bridges leaked).

	  The fix for this is to hone in better on when the channel has exited the
	  event loop. The stasis_app_control structure has an is_done field that
	  is now set at each point where the channel may exit the event loop. If
	  step 2b retrieves a valid stasis_app_control structure but the control
	  is marked as done, then the attempted operation exits immediately since
	  there will be nothing to service the attempted command.

	  ASTERISK-25091 #close
	  Reported by Ilya Trikoz

	  Change-Id: If66265b73b4c9f8f58599124d777fedc54576628
2015-06-17 07:00 +0000 [9668a1acb5]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Remove 'prefetch' option.

	  To prevent confusion I am removing the prefetch option until such
	  time as it is implemented. All other functionality, however, has
	  been implemented.

	  ASTERISK-25067

	  Change-Id: I9ce6aa3e5c6c5bc3c5baa8ff90fa036d73939895

2015-06-16 11:13 +0000 [59552c2d08]  Mark Michelson <mmichelson@digium.com>

	* Parking: Add documentation for AMI ParkedCallSwap event.

	  This event was added some time ago in order to clarify when a channel
	  took the place of another channel in a parking lot. However, there was
	  no XML documentation added for the event. This patch adds the XML
	  documentation.

	  ASTERISK-24900 #close
	  Reported by Rusty Newton

	  Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac
2015-06-15 16:40 +0000 [ea9d5f155e]  Corey Farrell <git@cfware.com>

	* func_pjsip_aor: Fix leaked contact from iterator.

	  ASTERISK-25162 #close

	  Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e

2015-06-12 16:58 +0000 [93ac45d3bd]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: Add option to force G.726 to be treated as AAL2 packed.

	  Some phones send g.726 audio packed for AAL2, which differs from what is
	  recommended by RFC 3351. If Asterisk receives audio formatted as such when
	  negotiating g.726 then it sounds a bit distorted. Added an option to
	  res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
	  AAL2 packed.

	  ASTERISK-25158 #close
	  Reported by: Steve Pitts

	  Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615

2015-06-14 19:48 +0000 [15c2208701]  Matt Jordan <mjordan@digium.com>

	* main/cdr: Carry over the disable flag when 'disable all' is specified

	  The CDR_PROP function (as well as the NoCDR application) set the
	  'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This
	  flag is supposed to be applied to all CDRs that are currently in the
	  chain, as well as all CDRs that may be created in the future. Currently,
	  however, the flag is only applied to the existing CDRs in the chain; new
	  CDRs do not receive the 'disable all' flag. In particular, this affects
	  parallel dials, which generate new CDRs for each pair of channels in
	  the dial attempt.

	  This patch carries over the 'disable all' flag when it is specified on a
	  CDR and a new CDR is generated for the chain.

	  ASTERISK-24344 #close

	  Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e
2015-06-12 14:28 +0000 [b8bc15286f]  Matt Jordan <mjordan@digium.com>

	* main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines

	  When a parallel dial occurs, a new CDR will be created for each dial
	  attempt that is made. In most circumstances, the act of creating each
	  CDR in the chain will include a step that updates the Party A snapshot,
	  which causes the context/extension of the Party A to be copied onto the
	  CDR object.

	  However, when the Party A is in a subroutine, we explicitly do *not*
	  copy the context/extension onto the CDR. This prevents the Macro or
	  GoSub routine name from blowing away the context/extension that the
	  channel was originally executing in. For the original CDR, this is not a
	  problem: the original CDR already recorded the last known 'good' state
	  of the channel just prior to it going into the subroutine. However, for
	  newly generated CDRs in a chain, there is no context/extension set on
	  them. Since we are in a subroutine, we will never set the Party A's
	  context/extension on the CDR, and we end up with a CDR with no
	  destination recorded on it.

	  This patch updates the creation of a chained CDR such that it copies
	  over the original CDR's context/extension. This is the last known "good"
	  state of the CDR, and is a reasonable starting point for the newly
	  generated CDR. In the case where we are not in a subroutine, subsequent
	  code will update the location of the CDR from the Party A information;
	  in the case where we are in a subroutine, the context/extension on the
	  original CDR is the correct information.

	  ASTERISK-24443 #close

	  Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a

2015-06-11 08:18 +0000 [19f60d9412]  Damian Ivereigh <damo@launtel.net.au>

	* chan_sip.c: Update dialog fromtag after request with auth

	  If a client sends and INVITE which is 401 rejected, then subsequently
	  sends a new INVITE with the auth info and uses a different fromtag
	  from the first INVITE, Asterisk will accept the new INVITE as part of
	  the original dialog - match_req_to_dialog() specifically ignores the
	  fromtag. However it does not update the stored dialog with the new
	  fromtag.

	  This results in Asterisk being unable to match future packets that are
	  part of this dialog (such as the ACK to the OK or the OK to the BYE),
	  and the call is dropped.

	  This problem was originally found when using an NEC-i SV8100-GE (NEC SIP
	  Card).

	  * After a successful match of a packet to the dialog, if the packet is
	    not a SIP_RESPONSE, authentication is present and the fromtags are
	    different, the stored fromtag is updated with the one from the recent
	    INVITE.

	  ASTERISK-25154 #close
	  Reported by: Damian Ivereigh
	  Tested by: Damian Ivereigh

	  Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e

2015-06-11 18:52 +0000 [bb00b26f35]  Matt Jordan <mjordan@digium.com>

	* chan_pjsip: Set the context and extension on the channel when created

	  Prior to this patch, chan_pjsip was failing to pass the endpoint's
	  context and the desired extension to the ast_channel_alloc_* routine.
	  This caused a new channel snapshot to be issued without a context and
	  extension, which can cause some reporting issues for users of AMI, CEL,
	  and other APIs. The channel driver would later set the context and
	  extension on the channel such that the channel would start in the
	  correct location in the dialplan, but the information reported in the
	  initial event would be incorrect.

	  This patch modifies the channel driver such that it now passes the
	  context and extension directly into the allocation routine. This
	  provides the information in the new channel snapshot published over
	  Stasis.

	  ASTERISK-25156 #close
	  Reported by: cloos

	  Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e

2015-06-10 18:28 +0000 [7230ee2efe]  Joshua Colp <jcolp@digium.com>

	* bridge: When performing a blonde transfer update connected line information.

	  When performing a blonde transfer the code uses the old masquerade
	  mechanism to move a channel around. As a result of this certain information,
	  such as connected line, is moved between the channels involved. Upon
	  completion of the move a frame is queued which is supposed to update the
	  connected line information on the channel. This does not occur as the
	  code considers it a redundant update since the masquerade operation
	  updated the channel (but did not inform it of the new connected line
	  information). The code also does not queue a connected line update
	  to be handled by the thread handling the channel. Without this any
	  other channel that may be loosely involved does not know it is
	  talking to a different caller.

	  This change does the following to resolve this:

	  1. The indicated connected line information is cleared upon
	  completion of the masquerade operation when doing a blonde transfer.
	  This prevents the connected line update from being considered
	  redundant.

	  2. A connected line update frame is now queued upon the completion
	  of the masquerade operation so any other channel loosely involved
	  knows that there is a different caller.

	  ASTERISK-25157 #close
	  Reported by: Joshua Colp

	  Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20

2015-06-11 14:39 +0000 [a657ab12f9]  Richard Mudgett <rmudgett@digium.com>

	* app_directory: Fix crash when using the alias option 'a'.

	  The voicemail.conf mailbox key/value pair is defined as:
	  <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]]
	  Where all fields in the value including the field values are optional.

	  Since the parsing code for the mailbox key/value pair is sloppy, this
	  patch tightens the parsing for the directory information.

	  * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options'
	  respectively in search_directory_sub().  Those names make more sense.

	  * Made sure that search_directory_sub() is dealing with the voicemail.conf
	  mailbox options field if it even exists when looking for the 'hidefromdir'
	  and 'alias' options.

	  * Fix crash if a voicemail.conf mailbox is just
	  <mailbox>=<password>,<name> when the 'a' option is used.  If there were no
	  fields after the name then the 'options' pointer was not checked for NULL.

	  * Fix users.conf alias processing if the 'a' option is used.  The wrong
	  variable was used.

	  ASTERISK-25087 #close
	  Reported by: Chet Stevens

	  Change-Id: I86052ea77307beddddba5279824d39dc0d593374

2015-06-05 15:37 +0000 [30cd559345]  Richard Mudgett <rmudgett@digium.com>

	* DNS: Need to use the same serializer for a pjproject SIP transaction.

	  All send/receive processing for a SIP transaction needs to be done under
	  the same threadpool serializer to prevent reentrancy problems inside
	  pjproject when using an external DNS resolver to process messages for the
	  transaction.

	  * Add threadpool API call to get the current serializer associated with
	  the worker thread.

	  * Pick a serializer from a pool of default serializers if the caller of
	  res_pjsip.c:ast_sip_push_task() does not provide one.

	  This is a simple way to ensure that all outgoing SIP request messages are
	  processed under a serializer.  Otherwise, any place where a pushed task is
	  done that would result in an outgoing out-of-dialog request would need to
	  be modified to supply a serializer.  Serializers from the default
	  serializer pool are picked in a round robin sequence for simplicity.

	  A side effect is that the default serializer pool will limit the growth of
	  the thread pool from random tasks.  This is not necessarily a bad thing.

	  * Made pjsip_resolver.c use the requesting thread's serializer to execute
	  the async callback.

	  * Made pjsip_distributor.c save the thread's serializer name on the
	  outgoing request tdata struct so the response can be processed under the
	  same serializer.

	  ASTERISK-25115 #close
	  Reported by: John Bigelow

	  Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a

2015-06-05 12:16 +0000 [b23f33e7e5]  Richard Mudgett <rmudgett@digium.com>

	* DNS: Fix some corner cases.

	  * Fix query_set destruction before we are done kicking the queries off.

	  * Fixed no queries requested handling.

	  * Add empty queries request unit test.

	  * Added missing allocation check in ast_dns_query_set_add().

	  * Made initial pjsip resolving query vector slightly larger.

	  ASTERISK-25115
	  Reported by: John Bigelow

	  Change-Id: Ie8be8347d0992e93946d72b6e7b1299727b038f2

2015-06-10 17:51 +0000 [ae589da466]  Richard Mudgett <rmudgett@digium.com>

	* DNS: Remove trailing newline from summary and descriptions.

	  Those trailing newlines mess up test formatting.

	  Change-Id: I5e3f3a55b82c9d7acb9661201d4993d1958f1185

2015-06-05 11:43 +0000 [83bc9d366d]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_resolver.c: Fix debug code to only execute at acceptable debug level.

	  Change-Id: I1716c93d6e097ad28128ecb9e806aac7a4180c8a

2015-06-05 11:41 +0000 [6d49dccd85]  Richard Mudgett <rmudgett@digium.com>

	* DNS: Fix doxygen comments.

	  Change-Id: Icafea3fb4ea64ac027561b23cbfe2b17997dc549

2015-06-09 15:31 +0000 [b705c09dbb]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip.h: Fix some doxygen comments.

	  Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976

2015-06-05 13:46 +0000 [aa8479778e]  Richard Mudgett <rmudgett@digium.com>

	* taskprocessor.c: Remove extra unref from off-nominal path.

	  Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60

2015-05-31 12:37 +0000 [07f5f45e5a]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* res_pjsip_transport_websocket: Fix use-after-free bugs.

	  This patch fixes use-after-free bugs caught by AddressSanitizer.

	  1. PJSIP transport manager may decide to destroy transport on its own.
	  For example, when the contact registered via websocket has not renewed
	  its registration in time. The transport was destoyed, but the websocket
	  listener thread was still active until the socket closes, and then tried
	  to call transport_shutdown on transport that has been freed.

	  Also, the transport destructor accessed wstransport->rdata.tp_info.pool
	  right after freeing memory that contained wstransport itself.

	  This patch converts transport to an ao2 object, allowing it to be
	  refcounted, so that it is available until both websocket listener and
	  pjsip transport manager are finished with it.

	  2. The websocket listener deletes the last reference on websocket session
	  when the tcp connection is closed, and it gets destroyed, but
	  the transport manager may still use it, for example when disconnect
	  happens in the middle of a SIP transaction.

	  A new reference to websocket session has been added that is released
	  with the transport to prevent this.

	  ASTERISK-25096 #close
	  Reported by: Josh Kitchens

	  ASTERISK-24963 #close
	  Reported by: Badalian Vyacheslav

	  Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b

2015-06-09 13:41 +0000 [f897f36721]  ibercom <ibercom123@gmail.com>

	* weakref attribute detection broken with gcc 4.6 and higher

	  GCC 4.7 Manual:
	  http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html

	  weakref ("target")

	  A weak reference is an alias that does not by itself require a definition
	  to be given for the target symbol.

	  ASTERISK-22559 #close
	  Reported by: Ibercom

	  Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf

2015-06-08 10:09 +0000 [80621ce3c5]  Corey Farrell <git@cfware.com>

	* Fix unsafe uses of ast_context pointers.

	  Although ast_context_find, ast_context_find_or_create and
	  ast_context_destroy perform locking of the contexts table,
	  any context pointer can become invalid at any time that the
	  contexts table is unlocked. This change adds locking around
	  all complete operations involving these functions.

	  Places where ast_context_find was followed by ast_context_destroy
	  have been replaced with calls ast_context_destroy_by_name.

	  ASTERISK-25094 #close
	  Reported by: Corey Farrell

	  Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa

2015-06-08 09:44 +0000 [53c1126090]  Kevin Harwell <kharwell@digium.com>

	* AMI: Escape string values.

	  So this issue is a bit complicated. Since it is possible to pass values to AMI
	  that contain a '\r\n' (or other similar sequences) these values need to be
	  escaped. One way to solve this is to escape the values and then pass the escaped
	  values to the AMI variable parameter string building function. However, this
	  puts the onus on the pre-build function to escape all string values. This
	  potentially requires a fair amount of changes along with a lot of string
	  allocations/freeing for all values.

	  Surely there is a way to push this complexity down a level into the string
	  building function itself? This of course is possible, but ends up requiring a
	  way to distinguish between strings that need to be escaped and those that don't.
	  The best way to handle this is by introducing a new format specifier in the
	  format string. For instance a %s (no escape) and %S (escape). However, that is
	  a bit weird and unexpected.

	  So faced with those possibilities this patch implements a limited version of the
	  first option. Instead of attempting to escape all string values this patch only
	  escapes those values that make sense. This approach limits the number of changes
	  and doesn't suffer from the odd format specifier problem.

	  ASTERISK-24934 #close
	  Reported by: warren smith

	  Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0

2015-06-02 15:07 +0000 [9fca378b36]  David M. Lee <dlee@respoke.io>

	* Fixes for OS X

	   * Add some type casting so tv_usec can really be a long, instead of
	     some strange platform specific type.

	   * Add some .dylib style files to .gitignore.

	   * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer
	     versions of GCC, when compiling the Homebrew formula for Asterisk,
	     are not properly passing the -Xlinker options to the linker. Given
	     that -Wl, does exactly the [same thing][], and does it properly, this
	     patch changes the -Xlinker options to use -Wl, instead.

	   [reasons unknown]: http://bit.ly/1SUbEYx
	   [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html

	  Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd

2015-06-04 07:14 +0000 [d463bac574]  ibercom <ibercom123@gmail.com>

	* CLI: Cosmetic issue - core show uptime

	  Show uptime information ends with an unnecessary space.

	  Now NEEDCOMMA is better defined.

	  Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1

2015-06-04 13:11 +0000 [128fe4cee8]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Implement expire_on_reload option.

	  This change implements the expire_on_reload option for memory caches.
	  If enabled and a reload is performed all objects within the cache
	  will be expired and the cache emptied.

	  ASTERISK-25067
	  Reported by: Matt Jordan

	  Change-Id: Id46aa1957d660556700e689e195eed57c989b85e

2015-06-02 10:20 +0000 [028edae82e]  Joshua Colp <jcolp@digium.com>

	* test_sorcery_memory_cache_thrash: Add unit tests for thrashing the memory cache.

	  This change adds a CLI command which can perform memory cache thrashing as well
	  as unit tests which perform thrashing under the following configurations:

	  1. Low number of unique objects that go stale after 1 second
	  2. Low number of unique objects that expire after 1 second
	  3. Low number of unique objects which are constantly updated
	  4. Large number of unique objects which exceed a defined cache size
	  5. Large number of unique objects which exceed a defined cache size
	     that also expire and go stale rapidly
	  6. Large number of unique objects which expire and go stale rapidly
	  7. Large number of unique objects

	  For all of the above there are a large number of threads constantly
	  attempting to retrieve random objects and each test runs for a few
	  seconds.

	  ASTERISK-25067
	  Reported by: Matt Jordan

	  Change-Id: I8c8ceff977332c80ed4a31f10d694d48552b2f78

2015-06-04 05:33 +0000 [19de2bbc5f]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Add test event when a refresh occurs.

	  This change adds a testsuite event for when a refresh occurs.
	  This is useful as it provides a guaranteed mechanism of knowing when
	  it has occurred instead of waiting an arbitrary amount of time.

	  ASTERISK-25067
	  Reported by: Matt Jordan

	  Change-Id: Iaa6b8d2d6bab7f99ee08e1c8908b8272a8987e65

2015-06-03 20:12 +0000 [6737ded058]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* install_prereq: Check if is installed aptitude otherwise to install.

	  If in Debian or system based, dont have aptitude installed the script do
	  nothing. This patch checked if aptitude  installed, if not installed.

	  Also, if execute script with all packages installed yet, the script not show
	  nothing and return exit 1 because the command 'grep' get nothing from pipe from
	  'awk'.

	  ASTERISK-25113 #close
	  Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	  Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f

2015-06-03 17:41 +0000 [92ccffd9e6]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip: Prevent access of NULL channels.

	  It is possible to receive incoming requests or responses after the channel
	  on an ast_sip_session has been destroyed and NULLed out. Handlers of these
	  sorts of requests or responses need to be prepared for the possibility
	  that the channel is NULL or else they could cause a crash.

	  While several places have been amended to deal with NULL channels, there
	  were still a couple of places that needed updating.

	  res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to
	  return early if there is no channel on the session.

	  res_pjsip_session.c: When handling a 302 response, we need to stop the
	  redirecting attempt if there is no channel on the session.

	  ASTERISK-25148 #close
	  reported by Mark Michelson

	  Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9

2015-06-03 13:17 +0000 [d355ee7ff3]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip/location: Fix ref leak in contact_apply_handler

	  contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status
	  to force the creation of a contact_status object whenever a new
	  contact is added but it didn't unref the returned object.

	  Added an ao2_cleanup(status) to plug the leak.

	  ASTERISK-25141

	  Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40
	  Reported-by: Corey Farrell

2015-06-02 13:02 +0000 [6d8dc9bb5c]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Remove outgoing authentication code no longer needed.

	  Associated with ASTERISK-25131

	  Change-Id: Iefa3b2066cfd8b108a90d2dd4a64d92c3a195d33

2015-06-02 12:55 +0000 [00a47ffc7e]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session: Fix cherry pick to master compile error.

	  ASTERISK-25131
	  Reported by: Richard Mudgett

	  Change-Id: I87c9c96ae4a8fe2bc8a0ddea6958a2ad9cefd8e3

2015-06-02 12:27 +0000 [9472bbaa95]  Joerg Sonnenberger <joerg@bec.de>

	* Remove const cast from leaf functions.

	  app_control_register_rule and app_control_unregister_rule lock/unlock
	  the queue, which is a mutating operation according to the
	  ao2_lock/_unlock prototype. Depending on the specific (implicit) casts
	  in SCOPED_LOCK and RAII_VAR, the compiler may warn or not. As the only
	  callers of those functions do not have the const, get consistent results
	  by just dropping it.

	  Change-Id: Ib9e6296155a39bc5d627142a3828180c3cfe8fbb

2015-06-02 11:35 +0000 [5f712e82ac]  Joerg Sonnenberger <joerg@bec.de>

	* tcptls.c: Don't use OpenSSL functions when no SSL support is present.

	  Change-Id: I68a85a7fcbdb282140ff333c6274b6763d5f82a3
2015-06-01 12:08 +0000 [2cd40c2bd7]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr/cdr_csv.c: Set file name for csv master to the module when (re)loaded.

	  Compute the location for the csv master file when the module is
	  loaded or reload.  Before it was calculated every time a log
	  entry was written.

	  Change-Id: I3ed9f6a8f965308099db70b71128f43d4d3f5585
2015-05-26 13:56 +0000 [5cdcae5240]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session: Fix in-dialog authentication.

	  When the remote peer requires authentication for in-dialog requests then
	  re-INVITEs to the peer cause the call to be disconnected and other
	  in-dialog requests to the peer like MESSAGE just don't go through.

	  * Made session_inv_on_tsx_state_changed() handle in-dialog authentication
	  for re-INVITEs and other methods.  Initial INVITEs cannot be handled here
	  because the INVITE transaction must be restarted earlier.

	  * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in
	  preparation for removing the file.  The generic outbound authentication
	  code did not work as well as anticipated.

	  * Created outbound_invite_auth() to only handle initial outbound INVITEs.
	  Re-INVITEs cannot be handled here.  The re-INVITE transaction is still in
	  progress and the PJSIP library cannot handle the overlapping INVITE
	  transactions.  Other method types should not be handled here as this code
	  only works on outgoing calls and we need to handle incoming and outgoing
	  calls.

	  ASTERISK-25131 #close
	  Reported by: Richard Mudgett

	  Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0

2015-05-30 20:22 +0000 [9f1939ee27]  Corey Farrell <git@cfware.com>

	* pjsip_configuration: Fix leak in persistent_endpoint_update_state.

	  The loop to find the first available contact of an endpoint grabbed
	  contact from the iterator, then checked for offline state.  This
	  caused the first contact after the state was found to leak a reference.

	  ASTERISK-25141

	  Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08

2015-05-31 11:33 +0000 [0a5f8c0d73]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* Fix buffer overflow in slin sample frames generation.

	  The length of frames retured by sample functions was twice as large as
	  real, what caused global buffer overflow caught by AddressSanitizer.

	  ASTERISK-24717 #close
	  Reported by: Badalian Vyacheslav

	  Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6

2015-05-29 16:19 +0000 [bef000dd7c]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip/location:  Fix memory leak in permanent_uri_handler

	  When permanent_uri_handler was creating the contact status
	  object for each contact, it wasn't unreffing it at the
	  end of the loop.

	  ASTERISK-25141 #close
	  Reported-by: Corey Farrell

	  Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12

2015-05-29 14:52 +0000 [82716410a4]  gtjoseph <george.joseph@fairview5.com>

	* Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change"

	  This reverts commit 6fca75bb628dfff2ab112e80b0228cf3ac0b8a05.

	  Change-Id: Ifee026cc63e22c5ac5717c37867a9f036373ae5a

2015-05-26 07:34 +0000 [dfc45254d1]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Add CLI commands and AMI actions.

	  This change adds the following CLI commands and AMI actions:

	  sorcery memory cache show
	  sorcery memory cache dump
	  sorcery memory cache expire
	  sorcery memory cache stale

	  SorceryMemoryCacheExpire
	  SorceryMemoryCacheExpireObject
	  SorceryMemoryCacheStale
	  SorceryMemoryCacheStaleObject

	  These allow both examination and manipulation of sorcery memory
	  caches from external sources.

	  Cached objects can be explicitly expired from a cache or marked
	  as stale. If expired they are immediately removed. If marked as
	  stale they will be background refreshed when next retrieved.

	  ASTERISK-25067
	  Reported by Matt Jordan

	  Change-Id: I68e03cfd8c34b5e07f4b6ee4fd93a3f4a00a3d9e

2015-05-27 13:22 +0000 [6fca75bb62]  gtjoseph <george.joseph@fairview5.com>

	* endpoint/stasis: Eliminate duplicate events on endpoint status change

	  When an endpoint was created, it's messages were being forwarded to
	  both the tech endpoint topic and the all endpoints topic.  Since
	  the tech topic was also forwarded to all, this was resulting in
	  duplicate messages whenever an endpoint published.  This patch
	  causes the endpoint to only forward to the tech topic and lets
	  the tech topic forward to all.

	  To accomplish this, the existing stasis_cp_single_create function
	  (which both creates and forwards) was cloned and split into 2
	  functions, one that creates the topic and one that sets up the
	  forwarding.  This allows endpoint_internal_create to create
	  the topic from the endpoint_all cache without forwarding it there,
	  then allows it to do the forward to the tech's topic.

	  ASTERISK-25137 #close
	  Reported-by: Vitezslav Novy
	  ASTERISK-25116 #close
	  Reported-by: George Joseph <george.joseph@fairview5.com>
	  Tested-by: George Joseph <george.joseph@fairview5.com>

	  Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c

2015-05-26 13:01 +0000 [2e54e7227c]  Mark Michelson <mmichelson@digium.com>

	* res_sorcery_memory_cache: Add support for refreshing stale objects.

	  This change introduces a check of object_lifetime_stale when retrieving
	  cached objects. If the amount of time the object has been in the cache
	  exceeds the lifetime, then a task is scheduled to update the cached
	  object based on an object retrieved from other sorcery wizards instead.

	  To prevent the cached object from being retrieved during a refresh,
	  thread-local storage is used to mark the thread as being a stale object
	  update. This results in the cache returning no object, leading to
	  sorcery querying other wizards for the object instead.

	  A test has been added for stale objects as well. This test ensures that
	  stale objects are retrieved the same as freshly-cached objects. The test
	  also ensures that after an object is stale, changes in the backend are
	  reflected in the cache, to include if the object has been deleted from
	  the backend.

	  ASTERISK-25067
	  Reported by Matt Jordan

	  Change-Id: I9bd7c049adf6939bfe2899f393c2bfbbf412d217
2015-05-21 17:21 +0000 [b8ac683822]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes

	  Add a new ContactStatus AMI event.
	  Publish the following status/state changes:
	  Created
	  Removed
	  Reachable
	  Unreachable
	  Unknown

	  Contact URI, new status/state, aor and endpoint names, and the
	  last qualify rtt result are included in the event.

	  ASTERISK-25114 #close

	  Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
	  Reported-by: George Joseph <george.joseph@fairview5.com>
	  Tested-by: George Joseph <george.joseph@fairview5.com>

2015-05-07 11:18 +0000 [95b186a174]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* res/res_config_pgsql.c: Use PQescapeStringConn for escaping names.

	  Use function PQescapeStringConn for escaping the name of the table and
	  schema instead of doing it manually.

	  ASTERISK-25132 #close
	  Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	  Change-Id: I302a263f7210d20925f14716b508b081998b7608

2015-05-26 07:44 +0000 [a7af6bca3c]  Joshua Colp <jcolp@digium.com>

	* sorcery: Fix cache creation callback.

	  The cache creation callback function expects to receive a sorcery_details
	  structure and not just a standalone object.

	  Change-Id: I3e4a5a137cb25292eb52d7a14cbb6daa09213450

2015-05-24 13:47 +0000 [23a798fecc]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* Astobj2: Correctly treat hash_fn returning INT_MIN

	  The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0.
	  However, abs(INT_MIN) = INT_MIN and is still negative, as well as
	  abs(INT_MIN) % num_buckets, and as a result this led to a crash.

	  One way to trigger the bug is using host=::80 or 0.0.0.128 in peer
	  configuration section in chan_sip or chan_iax.

	  This patch takes the remainder before applying abs, so that bucket
	  number is always in range.

	  ASTERISK-25100 #close
	  Reported by: Mark Petersen

	  Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899
2015-05-23 04:36 +0000 [70d54ab6c4]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* res_pjsip_transport_websocket: Fix crash on receiving large SIP packets

	  Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves
	  truncated before passing to pjsip_tpmgr_receive_packet, but the length
	  was passed unaltered, thus causing memory corruption and segfault.

	  ASTERISK-25122 #close

	  Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab

2015-05-22 21:50 +0000 [50044fdc15]  Corey Farrell <git@cfware.com>

	* Stasis: Fix unsafe use of stasis_unsubscribe in modules.

	  Many uses of stasis_unsubscribe in modules can be reached through unload.
	  These have been switched to stasis_unsubscribe_and_join.

	  Some subscription callbacks do nothing, for these I've created a noop
	  callback function in stasis.c.  This is used by some modules that monitor
	  MWI topics in order to enable cache, since the callback does not become
	  invalid after dlclose it is safe to use stasis_unsubscribe on these, even
	  during module unload.

	  ASTERISK-25121 #close

	  Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c

2015-05-22 16:52 +0000 [5a1f2a5884]  Corey Farrell <git@cfware.com>

	* Astobj2: Run weakproxy subscription callbacks in reverse order.

	  Modify ao2_weakproxy_subscribe so each new subscription is added
	  to the head of the list.  This ensures that when other objects
	  are allocated and use a subscription to the weakproxy for cleanup,
	  cleanup will occur in the correct order.

	  ASTERISK-25120 #close

	  Change-Id: Ie0476f08ec21330de1b3f5a2dd3d9eb683df3d3d

2015-05-22 12:22 +0000 [f66c41e668]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS

	  In addition to specifying lists of 'presence' and 'message-summary',
	  users can also create lists of type 'dialog'. These should be treated in
	  the same fashion as 'presence'.

	  Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e

2015-05-22 12:18 +0000 [ad7192a8fd]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_exten_state: Fix confusing NOTICE message

	  When a SUBSCRIBE request is made to a dialplan hint that doesn't exist,
	  the current NOTICE message informing users of this swaps the context and
	  extension parameters. This can cause a bit of confusion.

	  Thanks to CptBurger in #asterisk for helping to point this out.

	  Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43

2015-05-17 20:36 +0000 [9cffcca5f9]  Matt Jordan <mjordan@digium.com>

	* res/ari: Register Stasis application on WebSocket attempt

	  Prior to this patch, when a WebSocket connection is made, ARI would not
	  be informed of the connection until after the WebSocket layer had
	  accepted the connection. This created a brief race condition where the
	  ARI client would be notified that it was connected, a channel would be
	  sent into the Stasis dialplan application, but ARI would not yet have
	  registered the Stasis application presented in the HTTP request that
	  established the WebSocket.

	  This patch resolves this issue by doing the following:
	   * When a WebSocket attempt is made, a callback is made into the ARI
	     application layer, which verifies and registers the apps presented in
	     the HTTP request. Because we do not yet have a WebSocket, we cannot
	     have an event session for the corresponding applications. Some
	     defensive checks were thus added to make the application objects
	     tolerant to a NULL event session.
	   * When a WebSocket connection is made, the registered application is
	     updated with the newly created event session that wraps the WebSocket
	     connection.

	  ASTERISK-24988 #close
	  Reported by: Joshua Colp

	  Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636

2015-05-20 11:11 +0000 [29ef6571cb]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip: Refactor endpt_send_transaction (qualify_timeout)

	  This patch refactors the transaction timeout processing to eliminate
	  calling the lower level public pjsip functions and reverts to calling
	  pjsip_endpt_send_request again.  This is the result of me noticing
	  a possible incompatibility with pjproject-2.4 which was causing
	  contact status flapping.

	  The original version of this feature used the lower level calls to
	  get access to the tsx structure in order to cancel the transaction
	  when our own timer expires. Since we no longer have that access,
	  if our own timer expires before the pjsip timer, we call the callbacks
	  and just let the pjsip transaction take it's own course.  When the
	  transaction ends, it discovers the callbacks have already been run
	  and just cleans itself up.

	  A few messages in pjsip_configuration were also added/cleaned up.

	  ASTERISK-25105 #close

	  Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e
	  Reported-by: George Joseph <george.joseph@fairview5.com>
	  Tested-by: George Joseph <george.joseph@fairview5.com>
2015-05-20 17:35 +0000 [81d375baad]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Add support for object_lifetime_maximum.

	  This makes the "object_lifetime_maximum" option operational.

	  On the addition of an object to an empty memory cache a scheduled
	  task is created which, when invoked, expires objects from the cache
	  which have exceeded their lifetime. If more objects have been added
	  the remaining life of the oldest object is used to schedule the
	  next invocation of the scheduled task.

	  If the oldest object is removed from the cache before it can be
	  expired automatically the scheduled task is cancelled, if possible,
	  and the lifetime of the next oldest is used to schedule the task.

	  If during these two operations no additional objects exist in the
	  cache then no task is scheduled.

	  An additional unit test has been added which verifies this
	  functionality.

	  ASTERISK-25067
	  Reported by: Matt Jordan

	  Change-Id: I87409674674a508e7717ee20739ca15cec6ba7b6

2015-05-20 00:45 +0000 [9e2a582d2d]  demon-ru <serov.d.p@gmail.com>

	* res_pjsip_outbound_registration: Check request URI for line.

	  When an inbound call is received the To header is checked
	  for the "line" option. Some remote servers will place this
	  in the request URI instead. This adds an additional check for
	  the option in the request URI.

	  ASTERISK-25072 #close
	  Reported by: Dmitriy Serov

	  Change-Id: Id4e44debbb80baad623b914a88574371575353c8

2015-05-20 15:19 +0000 [071b3d43cb]  Mark Michelson <mmichelson@digium.com>

	* res_sorcery_memory_cache: Add support for maximum_objects.

	  This makes the "maximum_objects" option operational.

	  A heap has been added alongside the hash table in the cache. When
	  objects are added to the cache, they are also added to the heap.
	  Similarly, when objects are removed from the cache, they are removed
	  from the heap.

	  The heap's use comes into play when an item is to be added to a "full"
	  cache. When the cache is full, the oldest item is removed from the
	  cache, using the heap to determine the oldest item.

	  A unit test has been added that verifies that the maximum_objects option
	  works as expected and that the oldest object is removed from the cache
	  when an object beyond the maximum is added.

	  ASTERISK-25067 #close
	  Reported by Matt Jordan

	  Change-Id: I490658830e9c4cbf0b3051e4cdc4913cf9f1b73a

2015-05-16 17:02 +0000 [f2cc766d81]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_memory_cache: Add basic module implementation.

	  This change adds a basic res_sorcery_memory_cache module which implements
	  configuration option parsing, configuration file parsing for threading,
	  sorcery interface implementation, and unit tests.

	  Objects can be added, updated, deleted, and retrieved from the memory
	  cache. Automatic expiration and stale handling will be added in the
	  future.

	  Note that unit tests exist within the module itself in case the
	  threading done as a result of expiration results in asynchronous
	  actions (which it likely will). Providing access and a notification
	  mechanism for an external test module would be complicated and
	  not worth it.

	  ASTERISK-25067 #close
	  Reported by: Matt Jordan

	  Change-Id: Id8a6a357ef5a83d466f81eee56a67d13eeb118b9

2015-05-21 17:51 +0000 [36e5402885]  Corey Farrell <git@cfware.com>

	* res_mwi_external_ami: Use module version of AMI registration.

	  Use ast_manager_register_xml for res_mwi_external_ami manager
	  actions.  This ensures the module is held open while any of
	  the actions are being run.

	  ASTERISK-25117 #close
	  Reported by: Corey Farrell

	  Change-Id: Iececfdc2da498b2c32b9e09042f5f12292007ac7

2015-05-21 13:05 +0000 [3e2a994c71]  Matt Jordan <mjordan@digium.com>

	* ARI: Update version to 1.7.0

	  This patch updates the version of ARI to 1.7.0 to reflect the backwards
	  compatible changes that will be introduced in 13.4.0.

	  Change-Id: I6c36e6144da426412f25828a868e4df916bff60a
	  (cherry picked from commit 9d8a462356a938eea82e8424242d89a682495b57)

2015-05-20 20:53 +0000 [d067847695]  Corey Farrell <git@cfware.com>

	* Logger: Reset defaults before processing config.

	  Reset options to default values before reloading config.  This ensures
	  that if a setting is removed or commented out of the configuration file
	  it is unset on reload.

	  ASTERISK-25112 #close
	  Reported by: Corey Farrell

	  Change-Id: Id24bb1fb0885c2c14cf8bd6f69a0c2ee7cd6c5bd

2015-05-20 19:05 +0000 [31f0d78d7b]  gtjoseph <george.joseph@fairview5.com>

	* app_playback: Suppress warnings on playback if channel hung up

	  If a channel hangs up while an audio file is playing, there's
	  no need to clutter up the logs with a warning so suppress it
	  if ast_check_hangup returns true.

	  Also, change warning to debug/2 in file.c if writing a frame
	  fails.  Same reasoning.

	  Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89
	  Reported-by: George Joseph <george.joseph@fairview5.com>
	  Tested-by: George Joseph <george.joseph@fairview5.com>

2015-04-20 16:00 +0000 [83ff268b9e]  Yousf Ateya <y.ateya@starkbits.com>

	* chan_iax2: Prevent deadlock between hangup and sending lagrq/ping

	  channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/
	  send_ping. This deadlock happens because the scheduled task send_lagrq(or
	  send_ping) starts execution after the call hangup procedure starts but before
	  it deletes the tasks in the scheduler.

	  The solution is to delete scheduled lagrq (and ping) task asynchronously
	  (i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will
	  be called in a new context (doesn't have callno locked).

	  This commit also cleans up the procedure of sending LAGRQ and PING.

	  main/sched.c: Do not assert when deleting non existant entry from scheduler.
	  This assert seems to be the reason for a lot of awkward code to avoid it.

	  ASTERISK-24983 #close
	  Reported by: Y Ateya

	  Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c

2015-05-14 15:21 +0000 [7bf88eb60d]  Kevin Harwell <kharwell@digium.com>

	* audiohook.c: Difference in read/write rates caused continuous buffer resets

	  Currently, everytime a sample rate change occurs (on read or write) the
	  associated factory buffers are reset. If the requested sample rate on a
	  read differed from that of a write then the buffers are continually reset
	  on every read and write. This has the side effect of emptying the buffer,
	  thus there being no data to read and then write to a file in the case of
	  call recording.

	  This patch fixes it so that an audiohook_list's rate always maintains the
	  maximum sample rate among hooks and formats. Audiohook sample rates are
	  only overwritten by this value when slin native compatibility is turned on.
	  Also, the audiohook sample rate can only overwrite the list's sample rate
	  when its rate is greater than that of the list or if compatibility is
	  turned off. This keeps the rate from constantly switching/resetting.

	  ASTERISK-24944 #close
	  Reported by: Ronald Raikes

	  Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f

2015-05-13 09:55 +0000 [5ce54ed74a]  Matt Jordan <mjordan@digium.com>

	* res/res_http_websocket: Add a pre-session established callback

	  This patch updates http_websocket and its corresponding implementation
	  with a pre-session established callback. This callback allows for
	  WebSocket server consumers to be notified when a WebSocket connection is
	  attempted, but before we accept it. Consumers can choose to reject the
	  connection, if their application specific logic allows for it.

	  As a result, this patch pulls out the previously private
	  websocket_protocol struct and makes it public, as
	  ast_websocket_protocol. In order to preserve backwards compatibility
	  with existing modules, the existing APIs were left as-is, and new APIs
	  were added for the creation of the ast_websocket_protocol as well as for
	  adding a sub-protocol to a WebSocket server.

	  In particular, the following new API calls were added:
	  * ast_websocket_add_protocol2 - add a protocol to the core WebSocket
	    server
	  * ast_websocket_server_add_protocol2 - add a protocol to a specific
	    WebSocket server
	  * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object.
	    Consumers can populate this with whatever callbacks they wish to
	    support, then add it to the core server or a specified server.

	  ASTERISK-24988
	  Reported by: Joshua Colp

	  Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2

2015-05-20 12:55 +0000 [ddb7cbef8e]  John Bigelow <jbigelow@digium.com>

	* res/res_resolver_unbound.c: Add missing include of signal.h

	  ASTERISK-25110 #close
	  Reported by: John Bigelow

	  Change-Id: I99a9d93f066f265357b647b8e99a75e45da5a39f

2015-05-06 21:18 +0000 [9c3c7797e5]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cel, cdr: Assigned separator for column name and values.

	  Use a separator string between column names and values for SQL sentences
	  instead of evaluating the separator to use each time.

	  This change adds a space after the comma in constructing SQL sentences.
	  Before the SQL was created like "INSERT INTO cdr(calldate,clid,dst"
	  without spaces between column name and values.

	  The files applied this change are cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c,
	  cel/cel_odbc.c

	  ASTERISK-25109 #close
	  Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	  Change-Id: Ia5a1a161f5e26e1643703b30f8cc9cf0860cc7ea

2015-05-17 07:15 +0000 [d8698b7f3f]  Matt Jordan <mjordan@digium.com>

	* doxygen: Fix doxygen errors

	  This patch fixes a number of errors and warning messages in the doxygen
	  log. Specifically, it addresses:
	  * A number of files incorrectly places a '\brief' tag immediately after
	    a '\file' tag. Doing so emits a warning, as '\file' takes an optional
	    argument specifying which file the doxygen comment is for. As '\brief'
	    is not a file, doxygen was unamused.
	  * A grouping of Stasis Topics and Messages in rtp_engine.h was
	    incorrectly terminated. We now correctly terminate the grouping, which
	    prevents members of rtp_engine.h from showing up in the wrong group.
	  * Group indicators which are not part of the Stasis Topics and Messages
	    group were removed. Group indicators without an \addtogroup or
	    \ingroup have no meaning.

	  Change-Id: Ia1415ffec6767e27233ae1cae5ed5970de5656d4

2015-05-19 13:01 +0000 [d2e998cd68]  Corey Edwards <tensai@zmonkey.org>

	* main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits

	  ASTERISK-24887 #close
	  Reported by: Makoto Dei
	  Tested by: tensai

	  Change-Id: I6a96f572adb17f76b3acafe503a01c48eb5dd9bf
2015-05-14 22:05 +0000 [17129d2c29]  snuffy <snuffy22@gmail.com>

	* chan_pjsip: Fix crash during off-nominal when no endpoint specified.

	  Add missing return -1 when no endpoint name is specified.

	  ASTERISK-25086 #close
	  Reported by: snuffy

	  Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e
2015-05-14 18:01 +0000 [5d93928175]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard/config: Fix template processing

	  The config wizard was always pulling the first occurrence of
	  a variable from an ast_variable list but this gets the template
	  value from the list instead of any overridden value.  This patch
	  creates ast_variable_find_last_in_list() in config.c and updates
	  res_pjsip_config_wizard to use it instead of
	  ast_variable_find_in_list.  Now the overridden values, where they
	  exist, are used instead of template variables.

	  Updated test_config to test the new API.

	  ASTERISK-25089 #close

	  Reported-by: George Joseph <george.joseph@fairview5.com>
	  Tested-by: George Joseph <george.joseph@fairview5.com>
	  Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4

2015-05-15 01:54 +0000 [e48d29054f]  snuffy <snuffy22@gmail.com>

	* cdr: Fix 'core show channel' CDR variable truncation.

	  When the new Bridging API was implemented, the workspace variable
	  changed to a malloc'd string, causing sizeof() to always be 8 (char).

	  Revert back to stored on stack string for workspace.

	  ASTERISK-25090 #close

	  Change-Id: I51e610ae87371df771ce7693a955510efb90f8f7
2015-05-10 09:55 +0000 [8f3f414d8c]  Alexander Traud (License 6520)

	* tcptls: Enable multiple TLS certificate chains (RSA+ECC+DSA) for server socket.

	  When a client connects to a server via SSL/TLS, the server commonly utilizes an
	  RSA key-pair. However, other such algorithms exist (i.e. DSA and ECDSA), and if
	  the server socket is configured with a certificate for either one of those, it
	  would lose its compatibility with RSA-only clients.

	  Now, the server socket can be configured with up to one RSA, ECDSA and DSA key
	  each. For example, if a client is not compatible with SHA-2 hashed certificates
	  like Nokia mobile phones, the server socket still can use RSA/SHA-1 for legacy
	  clients and ECDSA/SHA-2 for everyone else.

	  ASTERISK-24815 #close
	  Reported by: Alexander Traud
	  patches:
	    tls_rsa_ecc_dsa.patch uploaded by Alexander Traud (License 6520)

	  Change-Id: Iada5e00d326db5ef86e0af7069b4dfa1b979da9a

2015-05-14 17:12 +0000 [2415a14ce9]  Maciej Szmigiero <mail@maciej.szmigiero.name>

	* Add X.509 subject alternative name support to TLS certificate
	  verification.

	  This way one X.509 certificate can be used for hosts that
	  can be reached under multiple DNS names or for multiple hosts.

	  Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>

	  ASTERISK-25063 #close

	  Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f

2015-05-13 15:41 +0000 [3e89f01b55]  Jonathan Rose <jrose@digium.com>

	* Message.c: Clear message channel frames on cleanup

	  The message channel is a special channel that doesn't actually process frames.
	  However, certain actions can cause frames to be placed in the channel's read
	  queue including the Hangup application which is called on the channel after
	  each message is processed. Since the channel will continually be reused for
	  many messages, it's necessary to flush these frames at some point.

	  ASTERISK-25083 #close
	  Reported by: Jonathan Rose

	  Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f

2015-05-14 00:06 +0000 [0a46d43b9c]  Corey Farrell <git@cfware.com>

	* Fix potential crash after unload of func_periodic_hook or test_message.

	  These modules save a pointer to the context they create on load, and
	  use that pointer to destroy the context at unload.  It is not safe
	  to save this pointer, it is replaced during load of pbx_config,
	  pbx_lua or pbx_ael.

	  This change causes the modules to pass NULL to ast_context_destroy,
	  a safer way to perform the unregistration since it does not use
	  a pointer that could become invalid.

	  ASTERISK-25085 #close
	  Reported by: Corey Farrell

	  Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835

2015-05-12 08:58 +0000 [478fb4a388]  Corey Farrell <git@cfware.com>

	* MALLOC_DEBUG: Replace WRAP_LIBC_MALLOC with ASTMM_LIBC.

	  There are 3 ways that calls directly to standard allocator functions can
	  be dealt with:
	  1. Block their use, cause them to generate an error.  This is the default.
	  2. Replace them with the Asterisk equivalent function calls.
	  3. Leave them alone.

	  This change allows one of these 3 options to be selected by any source.
	  The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT,
	  or ASTMM_IGNORE to use option 1, 2 or 3 respectively.  Normally ASTMM_BLOCK
	  is the correct option, so it is default when ASTMM_LIBC is not defined.
	  In some cases when building 3rd party code it is desirable to have it use
	  Asterisk functions, without changing the whole source - ASTMM_REDIRECT
	  accomplishes this.  When using 3rd party libraries sometimes a static
	  inline function will make use of malloc or free.  In these cases it may
	  be unsafe to replace the allocator in the header, as it's possible the
	  memory could be freed by the library using standard allocators.  For
	  those cases ASTMM_IGNORE is needed.

	  Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7

2015-05-05 19:49 +0000 [eec010829a]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro.

	  Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723
2015-05-06 05:28 +0000 [46bb8449e8]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cel/cel_pgsql.c: Use the 'SEP' macro when appending a column name

	  When appending a column name to the sql buffer, the predicate, "if first is
	  non-null, use empty string; else, use comma", is identical to the 'SEP' macro
	  definition. Since they are the same, this patch replaces the redundant
	  predicate statement with the 'SEP' macro.

	  Change-Id: Ib8b6138b06a48381723108a05ab8752cb8700509
2015-05-12 17:45 +0000 [0d97d7cb94]  Jonathan Rose <jrose@digium.com>

	* app_voicemail: fix moving when old messages full

	  When completing voicemail playback of a message in the 'INBOX', the
	  message gets moved to the 'Old' messages folder. Without this patch, if
	  the 'Old' folder is already at its set limit, then the 'INBOX' message will
	  simply be deleted. With this patch, the flag to delete the message will be
	  removed if the save_to_folder function indicates that the message could
	  not be moved due to a full folder.

	  ASTERISK-25082 #close
	  Reported by: Jonathan Rose
	  Review: https://gerrit.asterisk.org/#/c/448/

	  Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f
2015-05-12 17:34 +0000 [0bb0d4a603]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision.

	  If an ISDN call is hungup by both sides at the same time a crash could
	  happen.

	  * Added missing NULL checks for the owner channel after calling
	  pri_queue_pvt_cause_data() in two places.  Code after those calls need to
	  check the owner channel pointer for NULL before use because
	  pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the
	  owner and the owner may get hung up.

	  ASTERISK-21893 #close
	  Reported by:  Alexandr Gordeev

	  Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a

2015-05-06 08:31 +0000 [57386dcb67]  Corey Farrell <git@cfware.com>

	* Allow command-line options to override asterisk.conf.

	  Previous versions of Asterisk processed command-line options before
	  processing asterisk.conf.  This meant that if an option was set in
	  asterisk.conf, it could not be overridden with the equivelent command
	  line option.  This change causes Asterisk to process the command-line
	  twice.  First it processes options that are needed to load asterisk.conf,
	  then it processes the remaining options after the config is read.

	  This changes the function of -X slightly.  Previously using -X without
	  disabling execincludes in asterisk.conf caused #exec to be usable in any
	  config.  Now -X only enables #exec for the load of asterisk.conf, if it
	  is wanted in the rest of the system it must be enabled with execincludes
	  in asterisk.conf.  Updated 'asterisk -h' and 'man asterisk' to reflect
	  the limited function of -X.

	  ASTERISK-25042 #close
	  Reported by: Corey Farrell

	  Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7

2015-05-05 15:32 +0000 [52407088f8]  gtjoseph <george.joseph@fairview5.com>

	* sorcery: Add API to insert/remove a wizard to/from an object type's list

	  Currently you can 'apply' a wizard to an object type but the wizard
	  always goes at the end of the object type's wizard list.  This patch
	  adds a new ast_sorcery_insert_wizard_mapping function that allows
	  you to insert a wizard anyplace in the list.  I.E.  You could
	  add a caching wizard to an object type and place it before all
	  wizards.

	  ast_sorcery_get_wizard_mapping_count and
	  ast_sorcery_get_wizard_mapping were added to allow examination
	  of the mapping list.

	  ast_sorcery_remove_mapping was added to remove a mapping by name.

	  As part of this patch, the object type's wizard list was converted
	  from an ao2_container to an AST_VECTOR_RW.

	  A new test was added to test_sorcery for this capability.

	  ASTERISK-25044 #close

	  Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57

2015-05-12 01:31 +0000 [cc853dcf90]  Corey Farrell <git@cfware.com>

	* Fix processing of asterisk.conf debug=yes.

	  The code which reads asterisk.conf supports processing the debug
	  option with ast_true, but ast_true returns -1.  This causes debug
	  to still be off, convert to 1 so debug will be on as requested.

	  ASTERISK-25042
	  Reported by: Corey Farrell

	  Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6

2015-05-10 02:26 +0000 [c624e4bae1]  Sebastian Kemper <sebastian_ml@gmx.net>

	* General: Fix recent menuselect-related cross compile regression

	  MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the
	  target platform. But menuselect is to be run on the build system, so
	  BUILD_CC needs to be used instead - like it was in the past, before the
	  recent changes (https://reviewboard.asterisk.org/r/4370/). This is the
	  patch for ASTERISK-25074.

	  ASTERISK-25074 #close
	  Reported by: Sebastian Kemper
	  Tested by: Sebastian Kemper

	  Change-Id: I8a2b1fc5deb6ad2b80f49baca35b1b13d468ebf8
2015-05-01 12:22 +0000 [e6daafb8a6]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr_pgsql, cel_pgsql: Store maximum buffer size to prevent reallocation

	  The code previously used a fixed size of 512 for the SQL
	  queries. Depending on the size this may require it to grow.

	  This change makes it so if the buffer size does grow the size
	  is stored and next time the buffer will be large enough.

	  Change-Id: I55385899f1c06dee47e4274c2d21538037b2d895
2015-05-09 16:58 +0000 [87d8b36755]  gtjoseph <george.joseph@fairview5.com>

	* vector:  Add REMOVE, ADD_SORTED and RESET macros

	  Based on feedback from Corey Farrell and Y Ateya, a few new
	  macros have been added...

	  AST_VECTOR_REMOVE which takes a parameter to indicate if
	  order should be preserved.

	  AST_VECTOR_ADD_SORTED which adds an element to
	  a sorted vector.

	  AST_VECTOR_RESET which cleans all elements from the vector
	  leaving the storage intact.

	  Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14

2015-05-11 07:07 +0000 [e6ebddd9ae]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* pbx/pbx_spool: Fix issue when call files were executed too early

	  pbx_spool used to delete/move the call file upon successful outgoing
	  call completion, but did not delete it from in-memory list of files
	  (dirlist, used only when compiled with inotify/kqueue support).
	  That resulted in an extra attempt to process that filename after
	  retrytime seconds.
	  Then, if a new file with the same name appears that is scheduled
	  in future further than the completed one plus its retrytime,
	  then it gets executed earlier than expected.

	  This patch fixes remove_from_queue function to also remove the entry
	  from the dirlist.

	  ASTERISK-17069 #close
	  Reported by: Jeremy Kister

	  ASTERISK-24442 #close
	  Reported by: tootai

	  Change-Id: If9ec9b88073661ce485d6b008fd0b2612e49a28b

2015-05-01 23:43 +0000 [c61b146238]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr_pgsql: Use PQescapeStringConn for escaping names.

	  Use function PQescapeStringConn for escaping the name
	  of the table and schema instead of doing it manually.

	  Change-Id: I6709165e2d00463e9c813d24f17830ad4910b599
2015-05-10 07:37 +0000 [2ab5d22c0d]  Yousf Ateya <y.ateya@starkbits.com>

	* res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS.

	  First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC
	  https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values.

	  Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31

2015-05-10 08:36 +0000 [f82bd76e3c]  Joshua Colp <jcolp@digium.com>

	* dns_srv: Fix SRV sorting when records with priority zero exist with non-zero.

	  The DNS SRV sorting code currently has an issue when records with a priority
	  of zero exist with records of a non-zero priority. This occurs because the
	  sorting code considers zero to mean unset when in reality is a valid
	  value. If the current priority is zero it will get replaced with any remaining
	  record that has a priority of non-zero, until no records of those exist after
	  which the records of priority zero are handled.

	  This change makes it so that the priority of the first remaining record is
	  the current starting priority. There is also a small optimization to prevent
	  iterating records when the starting priority is already zero.

	  Change-Id: I103511f35b50428f770bd4db3ffef70fb6f82d35

2015-05-08 18:01 +0000 [1503d0c14c]  Alexandre Fournier <alexandre.fournier@kiplink.fr>

	* res_config_mysql: Fix broken column type checking

	  MySQL configuration engine contains a bug in require_mysql(). This
	  function is used for column type checking in tables. This bug only
	  affects DATETIME, DATE and FLOAT types.

	  It came from mixing the first condition (switch-case-like
	  if/then/else), to check the expected column type, with the second
	  condition, to check the actual column type against the expected column
	  type. Both conditions must be checked separately in order to avoid the
	  execution of the wrong block.

	  ASTERISK-18252 #comment This patch might fix the issue
	  Reported by: Gareth Blades

	  ASTERISK-25041 #close
	  Reported by: Alexandre Fournier
	  Tested by: Alexandre Fournier

	  Change-Id: I0b8bf7e68ab938be8e6525a249260cb648cb0bfa

2015-05-08 14:47 +0000 [5e361e1476]  Rusty Newton <rnewton@digium.com>

	* configs/basic-pbx: Modified main IVR to play new Allison prompt.

	  The main IVR was playing demo-congrats. I've switched it over to the
	  basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt
	  has Allison prompting the user with the actual IVR menu.

	  ASTERISK-24892 #close

	  Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d

2015-05-08 12:30 +0000 [2d4dc0c963]  Corey Farrell <git@cfware.com>

	* Fix error's produced by astmm.h when standard allocators are used.

	  astmm.h includes defines that are meant to cause error's when standard
	  allocators (malloc, calloc, free, etc) are used.  It actually only
	  causes a warning, which is not always caught on certain sources.  In
	  modules this unknown symbol is not detected until runtime, where the
	  module fails to load.  This modifies the define's so that using one
	  of the blocked functions will cause a compile error regardless of
	  CFLAGS.

	  Moved spandsp header includes to before asterisk.h so the static inline
	  functions can continue using malloc and free.  Although these functions
	  are never called and optimized away, the updated replacement macro's
	  would still cause a failure.

	  Change-Id: I532640aca0913ba9da3b18c04a0f010ca1715af5

2015-05-08 10:39 +0000 [63c71c9f4a]  Sean Bright <sean@malleable.com>

	* res_rtp_asterisk: Issue ERROR if res_srtp is not found.

	  While trying to get WebRTC working with chan_pjsip, I was running
	  into the following error:

	      Attempted to set an invalid DTLS-SRTP configuration on RTP
	      instance...

	  Josh helpfully pointed out that res_srtp.so might not be loaded, and
	  sure enough, it wasn't. This patch adds a ERROR indiciating as much
	  to hopefully help others having a similar problem.

	  Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f

2015-05-07 17:49 +0000 [60bf9ed91a]  Rusty Newton <rnewton@digium.com>

	* sounds: Add Swedish sounds to Makefile and XML

	  Added the necessary lines to the Makefile and sounds.xml so we'll have the
	  Swedish sounds in all available formats in menuselect.

	  See also: Swedish sounds were added into the core sounds release 1.4.27.

	  ASTERISK-24744 #close

	  Reported by: Tove Hjelm
	  Tested by: Rusty Newton

	  Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908

2015-05-08 10:30 +0000 [f93b3a22d6]  Corey Farrell <git@cfware.com>

	* Fix crash in codec_lpc10 when MALLOC_DEBUG is enabled.

	  This switches codecs/lpc10/lpcini.c back to including "asterisk.h"
	  instead of <stdlib.h>.  lpcini.c allocates memory that is freed by
	  codec_lpc10.c, so it is important to use MALLOC_DEBUG allocator.
	  Added #define WRAP_LIBC_MALLOC to the start of the source to prevent
	  runtime symbol link error's.

	  Change-Id: I74f63fd09fdeb673ee7753122c3bb4722ab6e1ac

2015-05-07 14:54 +0000 [cf637f2510]  gtjoseph <george.joseph@fairview5.com>

	* doc: Make progdocs play nice with git

	  Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in

	  Changed /Makefile to copy asterisk-ng-doxygen.in to
	  asterisk-ng-doxygen then modify it with version instead of
	  modifying asterisk-ng-doxygen directly.  Updated clean
	  targets as well.

	  Updated /.gitignore and doc/.gitignore.

	  Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622

2015-05-04 14:43 +0000 [b885f719bf]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* contrib/editors: Fix vim syntax highlighting of comments in config files

	   * Added a lookbehind to one-line comment matcher to skip escaped
	     semicolons.
	   * Added support for block comments.

	  Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7

2015-05-06 13:24 +0000 [e33682cae2]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination

	  The res_pjsip_exten_state module currently has a race condition between
	  processing the extension state callback from the PBX core and processing
	  the subscription shutdown callback from res_pjsip_pubsub. There is currently
	  no synchronization between the two. This can present a problem as while
	  the SIP subscription will remain valid the tree it points to may not.
	  This is in particular a problem as a task to send a NOTIFY may get queued
	  which will try to use the tree that may no longer be valid.

	  This change does the following to fix this problem:

	  1. All access to the subscription tree is done within the task that
	  sends the NOTIFY to ensure that no other thread is modifying or
	  destroying the tree. This task executes on the serializer for the
	  subscriptions.

	  2. A reference to the subscription serializer is kept to ensure it
	  remains valid for the lifetime of the extension state subscription.

	  3. The NOTIFY task has been changed so it will no longer attempt
	  to send a NOTIFY if the subscription has already been terminated.

	  ASTERISK-25057 #close
	  Reported by: Matt Jordan

	  Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643

2015-05-05 20:22 +0000 [c886be5df2]  gtjoseph <george.joseph@fairview5.com>

	* vector:  Additional enhancements and fixes

	  After using the new vector stuff for real I found...

	  A bug in AST_VECTOR_INSERT_AT that could cause a seg fault.

	  The callbacks needed to be closer to ao2_callback in behavior
	  WRT to CMP_MATCH and CMP_STOP behavior and the ability to return
	  a vector of matched entries.

	  A pre-existing issue with APPEND and REPLACE was also fixed.

	  I also added a new macro to test.h that acts like ast_test_validate
	  but also accepts a return code variable and a cleanup label.  As well
	  as printing the error, it sets the rc variable to AST_TEST_FAIL and
	  does a goto to the specified label on error.  I had a local version
	  of this in test_vector so I just moved it.

	  ASTERISK-25045

	  Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc

2015-05-06 17:37 +0000 [1f5db1c7e3]  Kevin Harwell <kharwell@digium.com>

	* res_stasis_snoop: Spying on a single direction continually increases CPU

	  Creating a snoop channel in ARI and spying only on a single direction (in or
	  out) results in CPU utilization continually increasing until the CPU is fully
	  consumed. This occurs because frames are being put in the opposing direction's
	  slin factory queue, but not being removed.

	  Fixed the problem by always reading and disposing of frames from the opposite
	  queue of the direction selected.

	  ASTERISK-24938 #closes

	  Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60

2015-05-06 16:00 +0000 [7103b374ef]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi: Improve force_restart_unavailable_chans option description.

	  ASTERISK-25034
	  Reported by: Richard Mudgett

	  Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30

2015-05-06 04:32 +0000 [d2e2271874]  Joshua Colp <jcolp@digium.com>

	* manager: Fix build due to missing variable usage.

	  Change-Id: I26d4d2cb9cee924632ff59ef0b30a7e6a1e2b00d

2015-05-04 20:11 +0000 [6b40bbf5bb]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* main/manager.c: Bugfix sort action_manager by alphabetically

	  Fix the alphabetic order added on ast_manager_register_struct. The order
	  for struct manager_action added is not working, this change fixes the
	  problem.

	  Change-Id: I149da0cd06c3c4445d7516cc303358e9f26f8b4b

2015-05-05 18:17 +0000 [6c4d1c3223]  Richard Mudgett <rmudgett@digium.com>

	* features: Fix crash when transferee hangs up during DTMF attended transfer.

	  A crash happens with this sequence of steps:
	  1) Party A is connected to party B.
	  2) Party B starts a DTMF attended transfer.
	  3) Party A hangs up while party B is dialing party C.

	  When party A hangs up the bridge that party A and party B are in is
	  dissolved and party B is kicked out of the bridge.  When party B finishes
	  dialing party C he attempts to move to the new bridge with party C.  Since
	  party B is no longer in a bridge the attempted move dereferences a NULL
	  bridge_channel pointer and crashes.

	  * Made the hold(), unhold(), ringing(), and the bridge_move() functions
	  tolerant of the channel not being in a bridge.  The assertion that party B
	  is always in a bridge is not true if the bridged peer of party B hangs up
	  and dissolves the bridge.  Being tolerant of not being in a bridge allows
	  the peer hangup stimulus to be processed by the FSM.

	  * Made the bridge_move() function return void since where the return value
	  for a failed move was checked generated a FSM coding ERROR message for a
	  normal off-nominal condition.

	  * Eliminated most uses of RAII_VAR in bridge_basic.c.

	  ASTERISK-25003 #close
	  Reported by: Artem Volodin

	  Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada

2015-05-05 14:48 +0000 [90bfc02e84]  Ivan Poddubny <ivan.poddubny@gmail.com>

	* app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter

	  This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3
	  parameters: position, original position and waiting time.

	  ASTERISK-25038 #close
	  Reported by: Etienne Lessard

	  Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618

2015-05-05 13:34 +0000 [bebf0b9b27]  Joshua Colp <jcolp@digium.com>

	* chan_unistim: Fix build failure due to ACL changes.

	  Change-Id: I57081045c72b9fcf12d5c84493278f9272c31b32

2015-05-05 11:35 +0000 [247fef6653]  Alexander Traud (License 6520)

	* tcptls: Avoiding ERR_remove_state in OpenSSL.

	  ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by 
	  ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were 
	  called by SSL_load_error_strings already and got removed. These changes allow 
	  OpenSSL forks like BoringSSL to be used with Asterisk.

	  ASTERISK-25043 #close
	  Reported by: Alexander Traud
	  patches:
	    asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520)

	  Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629
2015-05-05 09:47 +0000 [c541923ac3]  Corey Farrell <git@cfware.com>

	* res_ari_bridges: Add missing dependencies.

	  Missed this module in the previous commit.  res_ari_bridges uses symbols
	  from res_stasis_playback and res_stasis_recording.

	  ASTERISK-25027 #close
	  Reported by: Corey Farrell

	  Change-Id: I90bf756abd25adfc4920d2869ebe7feb636b8c5f

2015-05-05 09:27 +0000 [8a3e93a349]  Corey Farrell <git@cfware.com>

	* pbx_config: Register manager actions with module version of macro.

	  Switch manager actions in pbx_config to use the registration macro that
	  passes the module pointer, allowing pbx_config reference to be bumped
	  while the manager actions run.

	  ASTERISK-25061 #close
	  Reported by: Corey Farrell

	  Change-Id: I422c50dd74814616ac10c5e9c6598a0b1bc2c44e

2015-05-01 22:14 +0000 [cb79b8ab80]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cel_pgsql: Add support for setting schema

	  Add feature to set optional schema parameter on configuration file via
	  'schema' setting.

	  Fix query to get columns from table while considering schema. If in
	  the database there exists two tables with same name in distinct schemas
	  it will return an error when inserting record.

	  ASTERISK-24967 #close

	  Change-Id: I691fd2cbc277fcba10e615f5884f8de5d8152f2c

2015-05-04 12:16 +0000 [11f650c6ac]  Joshua Colp <jcolp@digium.com>

	* stasis: Fix dial masquerade datastore lifetime

	  A recent change went into Asterisk which added reference counts to the
	  channels stored in a dial masquerade datastore. Unfortunately this
	  included a reference to the caller in a dialing operation. While all
	  of the dialed targets have the datastore removed from them upon dialing
	  completion this did not occur for the caller, causing it to have a
	  reference to itself that could go never go away (as it depended on
	  the destruction of the datastore which only happened when the channel
	  was destroyed). This resulted in the caller channel remaining on the
	  system despite it having hung up.

	  This change does the following to fix this issue:

	  1. The dial masquerade datastore is now removed from the caller upon
	  dialing completion, just like the dialed targets.
	  2. Upon destruction of the caller all the dialed targets are also
	  removed from the dial masquerade datastore (just in case).
	  3. The reference to the caller has been removed as it should not be
	  possible for the datastore to now be valid/useful after the lifetime
	  of the caller has ended.

	  ASTERISK-25025 #close

	  Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f

2015-04-21 17:27 +0000 [a24ce38e5e]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr_adaptive_odbc: Add ability to set character for quoted identifiers.

	  Added the ability to set the character to quote identifiers. This
	  allows adding the character at the start and end of table and column
	  names. This setting is configurable for cdr_adaptive_odbc via the
	  quoted_identifiers in configuration file cdr_adaptive_odbc.conf.

	  ASTERISK-25006

	  Change-Id: I0b9a56b79ca13a727a803d88ed3b8643e37632b8

2015-05-04 22:57 +0000 [39cf642d40]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr: standardizes tab for options of AST_MODULE_INFO

	  Change-Id: I3c6de30b4859717873100092a7c06e206713a301

2015-05-04 16:41 +0000 [df6c1d755f]  Corey Farrell <git@cfware.com>

	* CLI: Enable automatic references to modules.

	  * Pass module to ast_cli_register and ast_cli_register_multiple.
	  * Add a module reference before executing any CLI callback, remove
	    the reference when complete.

	  ASTERISK-25049 #close
	  Reported by: Corey Farrell

	  Change-Id: I7aafc7c9f2b912918f28fe51d51e9e8a755750e3

2015-05-04 14:26 +0000 [a8bfa9e104]  Corey Farrell <git@cfware.com>

	* Modules: Make ast_module_info->self available to auxiliary sources.

	  ast_module_info->self is often needed to register items with the core.  Many
	  modules have ad-hoc code to make this pointer available to auxiliary sources.
	  This change updates the module build process to make the needed information
	  available to all sources in a module.

	  ASTERISK-25056 #close
	  Reported by: Corey Farrell

	  Change-Id: I18c8cd58fbcb1b708425f6757becaeca9fa91815

2015-05-01 19:25 +0000 [6d5941297b]  gtjoseph <george.joseph@fairview5.com>

	* vector:  Traversal, retrieval, insert and locking enhancements

	  Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really
	  does replace not insert.  The few users of AST_VECTOR_INSERT were
	  refactored.  Because these are macros, there should be no ABI
	  compatibility issues.

	  Added AST_VECTOR_INSERT_AT that actually inserts an element into the
	  vector at a specific index pushing existing elements to the right.

	  Added AST_VECTOR_GET_CMP that can retrieve from the vector based
	  on a user-provided compare function.

	  Added AST_VECTOR_CALLBACK function that will execute a function
	  for each element in the vector.  Similar to ao2_callback and
	  ao2_callback_data functions although the vector callback can take
	  a variable number of arguments.  This should allow easy migration
	  to a vector where a container might be too heavy.

	  Added read/write locked vector and lock manipulation macros.

	  Added unit tests.

	  ASTERISK-25045 #close

	  Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0

2015-05-03 13:55 +0000 [4f4aaa0c30]  Corey Farrell <git@cfware.com>

	* main/test.c: Add test to verify there were no registration errors.

	  This adds a test that will fail if any test failed to register. Also fail
	  if any test registration produced a warning about missing a leading or
	  trailing slash.

	  ASTERISK-25053 #close
	  Reported by: Corey Farrell

	  Change-Id: I93e50b8fcbcfa7f1f5b41b2c44a51685c09529c3

2015-04-21 11:52 +0000 [ebe371357e]  Martin Tomec <tomec.martin@gmail.com>

	* res_odbc: Use negative connection cache for all connections

	  Apply the negative connection cache setting to all connections,
	  even those that are not pooled. This ensures that the connection
	  will not be  re-established before the negative connection cache
	  time is met.

	  ASTERISK-22708 #close

	  Change-Id: I431cc2e8584ab0b6908b3523d0a0e18c9a527271
2015-05-03 21:03 +0000 [981084f08c]  Corey Farrell <git@cfware.com>

	* Format Interfaces: Prevent unload except by shutdown.

	  Format interfaces cannot be unregistered, so the modules that provide them
	  need to be held open except by shutdown.

	  ASTERISK-25054 #close
	  Reported by: Corey Farrell

	  Change-Id: Iadbd9675bf0d30b8fded5a739b163db3ea2db8f3

2015-05-03 20:28 +0000 [75c0aa6979]  Matt Jordan <mjordan@digium.com>

	* contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update

	  The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa7755
	  failed to add ENUM support for Postgres databases. This requires a
	  specific import from the sqlalchemy.dialects.postgresql package. This
	  patch corrects this error, which allows for Postgres update scripts to
	  be generated.

	  ASTERISK-24706

	  Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015

2015-05-03 13:36 +0000 [1368dae773]  Corey Farrell <git@cfware.com>

	* main/presencestate.c: Add trailing slash to test category.

	  ASTERISK-25053
	  Reported by: Corey Farrell

	  Change-Id: I8c0375dd0818747b2d2e1ceaea87bfbeb2daf8d4

2015-04-20 13:03 +0000 [305ce3defd]  Diederik de Groot <ddegroot@talon.nl>

	* Update configure.ac/Makefile for clang

	  Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which
	  checks compiler requirements for RAII:
	  gcc: -fnested-functions support
	  clang: -fblocks (and if required -lBlocksRuntime)
	  The original check was implemented in configure.ac and now has it's
	  own file. This function also sets C_COMPILER_FAMILY to either gcc or
	  clang for use by makefile

	  Created autoconf/ast_check_strsep_array_bounds.m4 (contains
	  AST_CHECK_STRSEP_ARRAY_BOUNDS):
	  which checks if clang is able to handle the optimized strsep & strcmp
	  functions (linux). If not, the standard libc implementation should be
	  used instead. Clang + the optimized macro's work with:
	  strsep(char *, char []), but not with strsepo(char *, char *).
	  Instead of replacing all the occurences throughout the source code,
	  not using the optimized macro version seemed easier

	  See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h':
	  llvm-comment: Normally, this array-bounds warning are suppressed for
	  macros, so that unused paths like the one that accesses __s1[3] are
	  not warned about.  But if you preprocess manually, and feed the
	  result to another instance of clang, it will warn about all the
	  possible forks of this particular if statement. Instead of switching
	  of this optimization, another solution would be to run the preproces-
	  sing step with -frewrite-includes, which should preserve enough
	  information so that clang should still be able to suppress the diag-
	  nostic at the compile step later on.

	  See also "https://llvm.org/bugs/show_bug.cgi?id=20144"
	  See also "https://llvm.org/bugs/show_bug.cgi?id=11536"

	  Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning
	  suppressions:
	  -Wno-unused-value
	  -Wno-parentheses-equality
	  In an earlier review (reviewboard: 4550 and 4554), they were deemed a
	  nuisace and less than benefitial.

	  configure.ac:
	  Added AST_CHECK_RAII() see earlier
	  Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier
	  Removed moved content

	  ASTERISK-24917
	  Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb

2015-04-28 04:49 +0000 [8886b724ae]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8

	  This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR
	  columns added in Asterisk 1.8. The columns are:
	   * peeraccount
	   * linkedid
	   * sequence
	  When enabled, the columns in the database entry will be populated with the data
	  from the CDR.

	  ASTERISK-24976 #close

	  Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b
2015-05-03 04:39 +0000 [94532b2c22]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* main/asterisk.c: Update Asterisk copyright year

	  Change-Id: I5e75d7f7e2c096d74edd9e8735268a894f4b93ab

2015-05-03 04:09 +0000 [2ed5e6a9ba]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* utils: Remove trailing whitespace

	  Change-Id: I4644f43a6a1ca9b5130cd2a6746772b888eb4f7a

2015-05-02 18:58 +0000 [c3ec5da156]  Corey Farrell <git@cfware.com>

	* Remove unneeded uses of optional_api providers.

	  A few cases exist where headers of optional_api provders are included but
	  not needed.  This causes unneeded calls to ast_optional_api_use.

	  * Don't include optional_api.h from sip_api.h.
	  * Move 'struct ast_channel_monitor' to channel.h.
	  * Don't include monitor.h from chan_sip.c, channel.c or features.c.

	  The move of struct ast_channel_monitor is needed since channel.c depends on
	  it.  This has no effect on users of monitor.h since channel.h is included
	  from monitor.h.

	  ASTERISK-25051 #close
	  Reported by: Corey Farrell

	  Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478

2015-05-02 02:15 +0000 [44bbdbe3a4]  Corey Farrell <git@cfware.com>

	* res_pjsip_dlg_options: Fix MODULEINFO section.

	  Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options.
	  This extra space prevented any of the dependencies from being seen by
	  menuselect, so building with default options would fail if PJSIP was
	  not installed.

	  This also makes the tool that extracts information for menuselect
	  tolerant of multiple spaces in the future.

	  ASTERISK-25033 #close
	  Reported by: Peter Whisker

	  Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698

2015-05-01 19:50 +0000 [e4f0a55f7f]  D Tucny <d@tucny.com>

	* term: send proper reset sequence when black background is forced

	  When using the force black background command-line option or configuration
	  option an invalid reset sequence is sent following a coloured output item 
	  in the CLI, the result is that the colour is not 'turned off' and continues
	  until the next non-default coloured text output.

	  A reset sequence is already defined in term.c, but the ast_term_reset
	  function doesn't use it, instead building it's own invalid sequence and 
	  returning that.

	  This patch changes that behaviour, removing the building of a reset sequence
	  and instead using the pre-built constant 'enddata' which is a suitable reset
	  sequence for this purpose.

	  ASTERISK-24896 #close
	  Reported by: Dan Tucny

	  Change-Id: I56323899123ae3264900389cae1f5b252aa3bf43
2015-05-01 13:22 +0000 [8f3cee1258]  Corey Farrell <git@cfware.com>

	* Astobj2: Fix initialization order of refdebug and AO2_DEBUG.

	  This ensures that refdebug is initialized before AO2_DEBUG if
	  both are enabled, since AO2_DEBUG allocates a container.

	  This change also makes AO2_DEBUG initialization critical, a
	  failure will abort Asterisk startup.  This is needed since
	  the failure would be caused by reg_containers allocation
	  failure, and that would result in a segmentation fault by
	  ao2_container_register later in startup.

	  ASTERISK-25048 #close
	  Reported by: Corey Farrell

	  Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244

2015-04-29 14:49 +0000 [7ac28be04b]  Matt Jordan <mjordan@digium.com>

	* main/pbx: Improve performance of dialplan reloads with a large number of hints

	  The PBX core maintains two hash tables for hints: a container of the
	  actual hints (hints), along with a container of devices that are watching that
	  hint (hintdevices). When a dialplan reload occurs, each hint in the hints
	  container is destroyed; this requires a lookup in the container of devices to
	  find the device => hint mapping object. In the current code, this performs an
	  ao2_callback, iterating over each of the device to hint objects in the
	  hintdevices container. For a large number of hints, this is extremely
	  expensive: dialplan reloads with 20000 hints could take several minutes
	  in just this phase.

	  This patch improves the performance of this step in the dialplan reloads
	  by caching which devices are watching a hint on the hint object itself.
	  Since we don't want to create a circular reference, we just cache the
	  name of the device. This allows us to perform a smarter ao2_callback on
	  the hintdevices container during hint removal, hashing on the name of the
	  device and returning an iterator to the matching names. The overall
	  performance improvement is rather large, taking this step down to a number of
	  seconds as opposed to minutes.

	  In addition, this patch also registers the hint containers in the PBX
	  core with the astobj2 library. This allows for reasonable debugging to
	  hash collisions in those containers.

	  ASTERISK-25040 #close
	  Reported by: Matt Jordan

	  Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360

2015-04-30 15:54 +0000 [6b208d8c3b]  Corey Farrell <git@cfware.com>

	* Sample Configs: Fix syntax error in pjsip.conf

	  The sample pjsip.conf has a few comment lines that are missing the
	  semicolons at the start of the comment, causing the config to fail
	  load.

	  Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352

2015-04-30 15:20 +0000 [dc23204aca]  Mark Michelson <mmichelson@digium.com>

	* Prevent potential crash on blond transfer.

	  Scenario:
	  Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects
	  the incoming call (or some other immediate circumstance causes Carol not
	  to answer the call)

	  What occurs in this case is that when the bridge between Alice and Bob
	  breaks, Alice is told to masquerade into Bob's channel that had placed
	  the call to Carol. The actual masquerade goes down without a hitch.
	  However, a channel fixup callback that attempts to publish dial events
	  over Stasis has a crash. The reason for this crash is that the datastore
	  on Bob's channel that placed the outbound call to Carol only had a bare
	  pointer to Carol's channel. Since Carol rejected the incoming call,
	  Carol's channel has been hung up and freed, meaning accessing her
	  channel results in a crash.

	  The fix here is simple. The dial fixup code has been altered to hold
	  references to the involved channels and to drop those references when
	  freeing data.

	  ASTERISK-25025 #close
	  Reported by Chet Stevens

	  Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197

2015-04-30 14:40 +0000 [47fa2ad10b]  Corey Farrell <git@cfware.com>

	* Build System: Fix issue with addons moduleinfo.

	  The build system now scans additional sources when generating
	  moduleinfo for menuselect.  Unfortunately the extra sources
	  for format_mp3 only exist if downloaded.

	  Use the Makefile macro 'wildcard' to allow moduleinfo generator
	  to ignore sources that do not exist.

	  Change-Id: I596604713b7345ce994f32197f8f6bfd9bcf4170

2015-04-30 13:42 +0000 [bb6ddb3dc8]  Joshua Colp <jcolp@digium.com>

	* res_ari_device_states: Fix dependency on res_stasis_device_state.

	  The res_ari_device_states module depends on res_stasis_device_state,
	  not res_stasis_device_states.

	  Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de

2015-04-28 17:00 +0000 [11ffcf662f]  Mark Michelson <mmichelson@digium.com>

	* Restrict functionality when ACLs are misconfigured.

	  This patch has two main purposes:

	  1) Improve warning messages when ACLs are configured improperly.
	  2) Prevent misconfigured ACLs from allowing potentially unwanted
	  traffic.

	  To acomplish point (2) in most cases, whatever configuration object that
	  the ACL belonged to was not allowed to load.

	  The one exception is res_pjsip_acl. In that case, ACLs are their own
	  configuration object. Furthermore, the module loading code has no
	  indication that a ACL configuration had a failure. So the tactic taken
	  here is to create an ACL that just blocks everything.

	  ASTERISK-24969
	  Reported by Corey Farrell

	  Change-Id: I2ebcb6959cefad03cea4d81401be946203fcacae

2015-04-29 14:29 +0000 [03c51cf525]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option.

	  Some telco switches occasionally ignore ISDN RESTART requests.  The fix
	  for ASTERISK-19608 added an escape clause for B channels in the restarting
	  state if the telco ignores a RESTART request.  If the telco fails to
	  acknowledge the RESTART then Asterisk will assume the telco acknowledged
	  the RESTART on the second call attempt requesting the B channel by the
	  telco.  The escape clause is good for dealing with RESTART requests in
	  general but it does cause the next call for the restarting B channel to be
	  rejected if the telco insists the call must go on that B channel.

	  chan_dahdi doesn't really need to issue a RESTART request in response to
	  receiving a cause 44 (Requested channel not available) code.  Sending the
	  RESTART in such a situation is not required (nor prohibited) by the
	  standards.  I think chan_dahdi does this for historical reasons to deal
	  with buggy peers to get channels unstuck in a similar fashion as the
	  chan_dahdi.conf resetinterval option.

	  * Add the chan_dahdi.conf force_restart_unavailable_chans compatability
	  option that when disabled will prevent chan_dahdi from trying to RESTART
	  the channel in response to a cause 44 code.

	  ASTERISK-25034 #close
	  Reported by: Richard Mudgett

	  Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65
2015-04-29 21:54 +0000 [556653d937]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr/cdr_csv.c: Refactor, function to write content of csv file.

	  Create a function for write content of CDR on csv files. Before used same
	  code for write two distinct files (account and master cdr) instead use a
	  function for thats.

	  Reduced to one lock when files are written.

	  Change-Id: Idce707f4c108083252e0aeb948f421d924953e65

2015-04-30 06:04 +0000 [80aa9aee5d]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_outbound_registration: Fix double unref on error return.

	  When the PJSIP pjsip_regc_send function is invoked and an error
	  status returned the caller currently decrements the reference count
	  of the client state that it just incremented, assuming the
	  registration callback would not have been invoked. In practice
	  this is not correct. If the failure happens after the transaction
	  has been set up the callback will still be invoked. This will
	  cause the reference count to be incorrectly decremented twice, once
	  by the registration callback and second by the caller of
	  pjsip_regc_send.

	  This change makes it so that whether the callback is invoked or
	  not is known by the caller of pjsip_regc_send. Depending on
	  this it can know whether it is responsible for decrementing the
	  reference count of the client state or not.

	  ASTERISK-25037 #close
	  Reported by: Joshua Colp

	  Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de

2015-04-30 02:07 +0000 [7ff3b2d479]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* include/asterisk/channel.h: Fix typo

	  Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3

2015-04-29 16:15 +0000 [39d3e1ef6e]  Matt Jordan <mjordan@digium.com>

	* main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8

	  The patch in 0b6410c4f8 did correctly fix a memory leak of the DTLS
	  structures in the RTP engine. However, when a 'core reload' is issued, a
	  double free of the memory pointed to by the char *'s in the DTLS
	  configuration struct can occur, as ast_rtp_dtls_cfg_free does not set
	  the pointers to NULL when they are freed.

	  This patch sets those pointers to NULL, preventing a second call to
	  ast_rtp_dtls_cfg_free from corrupting memory.

	  ASTERISK-25022

	  Change-Id: I820471e6070a37e3c26f760118c86770e12f6115

2015-04-29 13:05 +0000 [5d0c182885]  Kevin Harwell <kharwell@digium.com>

	* res_fax: allow 2400 transmission rate according to v.27ter standard

	  A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so
	  a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits
	  per second. This reverts all or some of those patches since according to the
	  v.27ter standard a rate of 2400 bits per second is also supported.

	  One of the original patches also added 9600 bits per second support for v.27.
	  This patch also removes that since v.27ter only supports 2400/4800 bits per
	  second.

	  Also, since Asterisk specifically supports v.27ter the enum was renamed to
	  better reflect this.

	  ASTERISK-24955 #close
	  Reported by: Matt Jordan

	  Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733

2015-04-28 23:35 +0000 [c9c03998cc]  Corey Farrell <git@cfware.com>

	* Astobj2: Add ao2_weakproxy_ref_object function.

	  This function allows code to run ao2_ref against the real
	  object associated with a weakproxy.  It is useful when
	  all of the following conditions are true:
	  * You have a pointer to weakproxy.
	  * You do not have or need a pointer to the real object.
	  * You need to ensure the real object exists and is not
	    destroyed during a process.

	  In this case it's wasteful to store a pointer to the real
	  object just for the sake of releasing it later.

	  Change-Id: I38a319b83314de75be74207a8771aab269bcca46

2015-04-27 16:13 +0000 [4f1db2070d]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_outbound_registration: Don't fail on delayed processing.

	  Odd behaviors have been observed during outbound registrations. The most
	  common problem witnessed has been one where a request with
	  authentication credentials cannot be created after receiving a 401
	  response. Other behaviors include apparently processing an incorrect SIP
	  response.

	  Inspecting the code led to an apparent issue with regards to how we
	  handle transactions in outbound registration code. When a response to a
	  REGISTER arrives, we save a pointer to the transaction and then push a
	  task onto the registration serializer. Between the time that we save the
	  pointer and push the task, it's possible for the transaction to be
	  destroyed due to a timeout. It's also possible for the address to be
	  reused by the transaction layer for a new transaction.

	  To allow for authentication of a REGISTER request to be authenticated
	  after the transaction has timed out, we now hold a reference to the
	  original REGISTER request instead of the transaction. The function for
	  creating a request with authentication has been altered to take the
	  original request instead of the transaction where the original request
	  was sent.

	  ASTERISK-25020
	  Reported by Mark Michelson

	  Change-Id: I756c19ab05ada5d0503175db9676acf87c686d0a
2015-04-29 10:46 +0000 [ed5715eb39]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_config: Fix build issue due to syntax error.

	  Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac

2015-04-29 06:46 +0000 [f226bd6f60]  Corey Farrell <git@cfware.com>

	* ARI: Fix missing dependencies.

	  ARI modules that are generated by 'make ari-stubs' are all dependent on
	  res_ari_model.  Additionally some of the same modules depend on one or more
	  res_stasis_* modules.

	  ASTERISK-25027 #close
	  Reported by: Corey Farrell

	  Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153

2015-04-29 06:26 +0000 [881844297a]  Corey Farrell <git@cfware.com>

	* res_pjsip: Remove incorrect MODULEINFO from presence_xml.c.

	  Remove incorrect MODULEINFO block and unneeded header includes
	  from presence_xml.c.

	  ASTERISK-25027
	  Reported by: Corey Farrell

	  Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb

2015-04-29 06:17 +0000 [c232ff3af0]  Corey Farrell <git@cfware.com>

	* Git Migration: Create doc/rest-api when needed.

	  Create the directory './doc/rest-api' at the start of 'make ari-stubs'
	  to prevent an error when documentation is generated.  The directory is
	  also added to git ignores.

	  ASTERISK-25027
	  Reported by: Corey Farrell

	  Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b

2015-04-29 03:03 +0000 [5d997ecc83]  Corey Farrell <git@cfware.com>

	* Build System: Prevent unneeded changes to asterisk/buildopts.h.

	  * Add AST_DEVMODE to BUILDOPTS
	  * Use BUILDOPTS to generate AST_BUILDOPT_SUM.
	  * Remove loop that defined AST_MODULE_*

	  These changes ensure that only ABI effecting options are considered for
	  AST_BUILDOPT_SUM.  This also reduces unneeded full system rebuilds caused
	  by enabling or disabling one module that another is dependent on.

	  ASTERISK-25028 #close
	  Reported by: Corey Farrell

	  Change-Id: I2c516d93df9f6aaa09ae079a8168c887a6ff93a2

2015-04-29 00:02 +0000 [55a780d211]  Corey Farrell <git@cfware.com>

	* Git Conversion: Switch Non-C files to ASTERISK_REGISTER_FILE.

	  This switches files used to generate other sources to use the new
	  ASTERISK_REGISTER_FILE macro.

	  ASTERISK-25026 #close
	  Reported by: Corey Farrell

	  Change-Id: Ieb2537b83421cad07c8955e5f90c405ccf079740

2015-04-28 13:28 +0000 [5ebfed8ef3]  Yousf Ateya <y.ateya@starkbits.com>

	* chan_iax2: Ensure that IAX flags are 64 bits.

	  Flags are 64 bits.  Without LLU suffix the value of 1<<31 is negative.
	  Although it doesn't have an effect on the current implementation, it will
	  be problem if more flags are added.

	  Change-Id: Ic290c81cfbbbf062872392d99d3322932cc49487
2015-04-28 00:29 +0000 [46cf643c75]  Ashley Sanders <asanders@digium.com>

	* chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR
	              Sections Exist in pjsip.conf

	  This patch modifies the current loading strategy of the pjsip configuration. If
	  duplicate sections (e.g. sections containing the same [id/type]) are defined in
	  [pjsip.conf], the loader will consider the configuration for the given type as
	  invalid when the duplicate section is encountered. The entire configuration
	  (including what was previously loaded) for the duplicate [id/type] sections
	  will be rejected and destroyed, an error message is logged and the load
	  processing for the given stops.

	  ASTERISK-24996
	  Reported By: Ashley Sanders

	  Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef
2015-04-28 11:50 +0000 [0bbe2c35cf]  Richard Mudgett <rmudgett@digium.com>

	* chan_vpb: Fix compile error due to use of ASTERISK_FILE_VERSION.

	  Change-Id: I51179e2a83937423676da522b766f1126de4059e
2015-04-27 14:44 +0000 [f47fed2e12]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_outbound_registration: Add debugging messages.

	  When problems occur regarding outbound registrations, it currently
	  is difficult to debug. Most off-nominal paths had warning messages,
	  but sometimes we want to know what's going on before hitting the
	  off-nominal path. This patch adds lots of debugging output that
	  should give a clearer picture of what is happening with regards
	  to outbound registrations.

	  ASTERISK-25020
	  Reported by Mark Michelson

	  Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45

2015-04-28 05:38 +0000 [5e96584829]  Steve Davies <steve@one47.co.uk>

	* res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS

	  ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created.
	  The resources are linked into a table, but the original alloc refs
	  are never released. ast_strdup leak in rtp_engine.c. If
	  ast_rtp_dtls_cfg_copy() is called twice on the same destination struct,
	  a pointer to an alloc'd string is overwritten before the string is free'd.

	  ASTERISK-25022
	  Reported by: one47

	  Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b

2015-04-28 04:28 +0000 [d6a2d92353]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr/cdr_csv.c: Add missing space after comma.

	  Change-Id: I3866a20019b1a3a2f10fe36640053929330b0fcb

2015-04-27 22:01 +0000 [542bfee881]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* CHANGES: Add missing spaces.

	  Change-Id: I534ea0f22759e3633585dfa9b145b4a284efe67f

2015-04-17 02:16 +0000 [5c1d07baf0]  Corey Farrell <git@cfware.com>

	* Astobj2: Allow reference debugging to be enabled/disabled by config.

	  * The REF_DEBUG compiler flag no longer has any effect on code that uses
	    Astobj2.  It is used to determine if reference debugging is enabled by
	    default.  Reference debugging can be enabled or disabled in asterisk.conf.
	  * Caller information is provided in logger errors for ao2 bad magic numbers.
	  * Optimizes AO2 by merging internal functions with the public counterpart.
	    This was possible now that we no longer require a dual ABI.

	  ASTERISK-24974 #close
	  Reported by: Corey Farrell

	  Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1

2015-04-27 12:11 +0000 [356568dc7f]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip: Fix SEGV on pending-qualify contacts

	  Permanent contacts that hadn't been qualified yet were missing
	  their contact_status entries causing SEGVs when running CLI
	  commands.

	  This patch makes sure that contact_statuses are created for
	  both dynamic and permanent contacts when they are created.
	  It also adds checks in the CLI code to make sure there's a
	  contact_status, just in case.

	  ASTERISK-25018 #close
	  Reported-by: Ivan Poddubny
	  Tested-by: Ivan Poddubny
	  Tested-by: George Joseph

	  Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029

2015-04-15 18:55 +0000 [358080e86e]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version

	  Add new column to INSERT new columns added in cdr 1.8 version. The columns are:
	   * peeraccount
	   * linkedid
	   * sequence
	  This feature is configurable in cdr_odbc.conf using a new configuration
	  option, 'newcdrcolumns'.

	  ASTERISK-24976 #close

	  Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127
2015-04-26 17:21 +0000 [d7f4788341]  Matt Jordan <mjordan@digium.com>

	* channels/chan_skinny: Fix compilation error introduced in f8e21a1adf

	  A typo in commit f8e21a1adf resulted in a compilation error in
	  chan_skinny. This patch fixes the typo.

	  ASTERISK-24917

	  Change-Id: Id7f4ad1fe948eb2408622e80c27936ce4516c33c

2015-04-23 17:29 +0000 [9f65ea482e]  Kevin Harwell <kharwell@digium.com>

	* app_confbridge: Default the template option to a compatible default profile.

	  Confbridge dynamic profiles did not have a default profile unless you
	  explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a
	  template was not set prior to the bridge being created then some
	  options were left with no default values set. This patch makes it so
	  the default templates are set to the default bridge and user profiles.

	  ASTERISK-24749 #close
	  Reported by: philippebolduc

	  Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a

2015-04-23 07:31 +0000 [cafdb7a049]  Olle E. Johansson <oej@edvina.net>

	* CREDITS: Update credits for Olle Johansson

	  Change-Id: I8f3d0a6c3f1075a1f7d8308593394611a96749de
2015-04-24 09:17 +0000 [bd61c9300c]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_outbound_authenticator: Increase CSeq on authed requests.

	  The way PJSIP generates an authenticated request is to use a previous
	  request as a template. This means that the authenticated request will
	  have the same Call-ID, From header (including tag), and CSeq as the
	  original request. PJSIP generates a new branch on the Via header to
	  indicate that this is a new transaction, though.

	  There are some SIP implementations, though, that do not notice the
	  change in the branch and therefore will match the authed request to the
	  original request's transaction. Since the CSeq is the same, the server
	  will repeat the response it sent to the original request.

	  This patch aids interoperability by increasing the CSeq of the authed
	  request by one.

	  ASTERISK-24845 #close
	  Reported by: Carl Fortin
	  Tested by: Carl Fortin

	  Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01

2015-04-22 04:17 +0000 [f8e21a1adf]  Diederik de Groot <ddegroot@talon.nl>

	* Clang: Fix some more tautological-compare warnings.

	  clang can warn about a so called tautological-compare, when it finds
	  comparisons which are logically always true, and are therefor deemed
	  unnecessary.

	  Exanple:
	  unsigned int x = 4;
	  if (x > 0)    // x is always going to be bigger than 0

	  Enum Case:
	  Each enumeration is its own type. Enums are an integer type but they
	  do not have to be *signed*. C leaves it up to the compiler as an
	  implementation option what to consider the integer type of a particu-
	  lar enumeration is. Gcc treats an enum without negative values as
	  an int while clang treats this enum as an unsigned int.

	  rmudgett & mmichelson: cast the enum to (unsigned int) in assert.
	  The cast does have an effect. For gcc, which seems to treat all enums
	  as int, the cast to unsigned int will eliminate the possibility of
	  negative values being allowed. For clang, which seems to treat enums
	  without any negative members as unsigned int, the cast will have no
	  effect. If for some reason in the future a negative value is ever
	  added to the enum the assert will still catch the negative value.

	  ASTERISK-24917
	  Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62

2015-04-20 13:06 +0000 [1e74793061]  Diederik de Groot <ddegroot@talon.nl>

	* Example script for scan-build (the llvm static analyzer)

	   - Added Pre-amble (Options / Flags / Usage Example / GNU License)
	   - Extended Configurability
	   - Made Executable

	  ASTERISK-24917
	  Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8

2015-04-23 12:54 +0000 [89a3fc0572]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX.

	  When Asterisk originates a channel to an application, the channel is
	  hung up once the application finishes executing. When the application
	  in question is SendFax, the Asterisk PJSIP code will attempt to reinvite
	  the T.38 session to audio after the FAX completes. The hangup of the
	  channel happens in the midst of this reinvite transaction. In most
	  circumstances, this works out okay because the BYE is delayed until the
	  reinvite transaction can complete.

	  However, if the reinvite that Asterisk sends receives a 401/407
	  response, then Asterisk's attempt to re-send the reinvite with
	  authentication will fail. This is because the session supplement in
	  res_pjsip_t38 makes the assumption that the channel on the session will
	  always be non-NULL. Since the channel has been hung up, though, the
	  channel is now NULL. Attempting to operate on the channel causes a
	  crash.

	  This patch fixes the issue by ensuring that the channel on the session
	  is not NULL before attempting to mess with the T.38 framehook.

	  This patch also contains some corrections for comments that were
	  incorrect and really confused me when I first started looking at the
	  code.

	  ASTERISK-25004 #close
	  Reported by Mark Michelson

	  Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0

2015-04-23 09:16 +0000 [75666ad7c6]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip:  Validate that contact uris start with sip: or sips:

	  Currently we use pjsip_parse_hdr to validate contact uris but it
	  appears that it allows uris without a scheme if there's a port
	  supplied.  I.E myexample.com will fail but myexample.com:5060 will
	  pass even though it has no scheme.  This causes SEGVs later on
	  whenever the uri is used.

	  To prevent this, permanent_contact_validate has been updated to check
	  that the scheme is either 'sip' or 'sips'.

	  2 uses of possibly-null endpoint have also been fixed in
	  create_out_of_dialog_request.

	  ASTERISK-24999

	  Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2
	  Reported-by: Brad Latus

2015-04-23 08:00 +0000 [ca7193167e]  Diederik de Groot <ddegroot@talon.nl>

	* Clang: change previous tautological-compare fixes.

	  clang can warn about a so called tautological-compare, when it finds
	  comparisons which are logically always true, and are therefor deemed
	  unnecessary.

	  Exanple:
	  unsigned int x = 4;
	  if (x > 0)    // x is always going to be bigger than 0

	  Enum Case:
	  Each enumeration is its own type. Enums are an integer type but they
	  do not have to be *signed*. C leaves it up to the compiler as an
	  implementation option what to consider the integer type of a particu-
	  lar enumeration is. Gcc treats an enum without negative values as
	  an int while clang treats this enum as an unsigned int.

	  rmudgett & mmichelson: cast the enum to (unsigned int) in assert.
	  The cast does have an effect. For gcc, which seems to treat all enums
	  as int, the cast to unsigned int will eliminate the possibility of
	  negative values being allowed. For clang, which seems to treat enums
	  without any negative members as unsigned int, the cast will have no
	  effect. If for some reason in the future a negative value is ever
	  added to the enum the assert will still catch the negative value.

	  ASTERISK-24917

	  Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a

2015-04-22 16:22 +0000 [cc77440deb]  gtjoseph <george.joseph@fairview5.com>

	* res_corosync: Add check for config file before calling corosync apis

	  On some systems, res_corosync isn't compatible with the installed version of
	  corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE,
	  and Asterisk terminates.  The work around has been to remember to add
	  res_corosync as a noload in modules.conf.  A better solution though is to have
	  res_corosync check for its config file before attempting to call corosync apis
	  and return LOAD_DECLINE if there's no config file.  This lets Asterisk loading
	  continue.

	  If you have a res_corosync.conf file and res_corosync fails, you get the same
	  behavior as today and the fatal error tells you something is wrong with the
	  install.

	  ASTERISK-24998

	  Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889
2015-04-22 15:17 +0000 [c231c85ea4]  Corey Farrell <git@cfware.com>

	* Astobj2: Ensure all calls to __adjust_lock pass a valid object.

	  __adjust_lock doesn't check for invalid objects, and doesn't have an
	  appropriate return value for invalid objects.  Most callers of
	  __adjust_lock pass objects that have already been confirmed valid,
	  this change adds checks before the remaining calls.

	  ASTERISK-24997 #close
	  Reported by: Corey Farrell

	  Change-Id: I669100f87937cc3f867cec56a27ae9c01292908f

2015-04-22 16:32 +0000 [0722e11f26]  gtjoseph <george.joseph@fairview5.com>

	* .gitignore:  Add .gcno and .gcda

	  Products of --enable-coverage

	  Change-Id: Ie20882d64b60692e2c941ea8872ab82a86ce77a3

2015-04-22 11:28 +0000 [7216e3c608]  Joshua Colp <jcolp@digium.com>

	* dns: Make query sets hold on to queries for their lifetime.

	  The query set documentation states that upon completion queries can be
	  retrieved for the lifetime of the query set. This is a reasonable
	  expectation but does not currently occur. This was originally done
	  to resolve a circular reference between queries and query sets, but
	  in practice the query can be kept.

	  This change makes it so a query does not have a reference to the
	  query set until it begins resolving. It also makes it so that the
	  reference is given up upon the query being completed. This allows
	  the queries to remain for the lifetime of the query set. As the
	  query set on the query is only useful to the query set functionality
	  and only for the lifetime that the query is resolving this is safe
	  to do.

	  ASTERISK-24994 #close
	  Reported by: Joshua Colp

	  Change-Id: I54e09c0cb45475896654e7835394524e816d1aa0

2015-04-20 13:01 +0000 [09c7c678a3]  Diederik de Groot <ddegroot@talon.nl>

	* Fix/Update clang-RAII macro implementation

	  - When you need to refer to 'variable XXX' outside a block, it needs
	  to be declared as '__block XXX', otherwise it will not be available with-
	  in the block, making updating that variable hard to do, and ast_free
	  lead to issues.

	  - Removed the #error message
	  because it creates complications when compiling external projects
	  against asterisk For example when using a different compiler than the
	  one used to compile asterisk. The warning/error should be generated
	  during the configure process not the compilation process

	  ASTERISK-24917
	  Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2

2015-04-14 14:04 +0000 [190fa4f333]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers.

	  Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon
	  a mailbox state change (such as a new message being left, or one being deleted).
	  In practice this is not sufficient to keep clients aware of the current MWI status.

	  This change makes the module send unsolicited MWI NOTIFY on startup so that
	  clients are guaranteed to have the most up to date MWI information. It also makes
	  clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware
	  of the current MWI status they receive it.

	  ASTERISK-24982 #close
	  Reported by: Joshua Colp

	  Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58

2015-04-21 17:45 +0000 [2a36bb5d9a]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* CHANGES remove tab space

	  Change-Id: I6b43e43474bf6fb77b8227eadb036036f8e90521

2015-04-21 15:17 +0000 [5757d2d30d]  Corey Farrell <git@cfware.com>

	* Check for ao2_alloc failure in __ast_channel_internal_alloc.

	  Fix a crash that could occur in __ast_channel_internal_alloc if
	  ao2_alloc fails.

	  ASTERISK-24991 #close

	  Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90

2015-04-20 14:30 +0000 [6331be0638]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs.

	  When SUBSCRIBE dialogs were established, we never associated
	  the endpoint that created the subscription with the dialog
	  we end up creating. In most cases, this ended up not causing
	  any problems.

	  The actual bug that was observed was that when a device that
	  was behind NAT established a subscription with Asterisk, Asterisk
	  would end up sending in-dialog NOTIFY requests to the device's
	  private IP addres instead of the public address of the NAT router.

	  When Asterisk receives the initial SUBSCRIBE from the device,
	  res_pjsip_nat rewrites the contact to the public address on which the
	  SUBSCRIBE was received. This allows for the dialog to have its target
	  address set to the proper public address. Asterisk then would send a 200
	  OK response to the SUBSCRIBE, then a NOTIFY with the initial
	  subscription state. The device would then send a 200 OK response to
	  Asterisk's NOTIFY.

	  Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat
	  did not rewrite the address in the Contact header. Then, when the PJSIP
	  dialog layer processed the 200 OK, PJSIP would perform a comparison
	  between the IP address in the Contact header and its saved target
	  address for the dialog. Since they differed, PJSIP would update the
	  target dialog address to be the address in the Contact header. From this
	  point, if Asterisk needed to send a NOTIFY to the device, the result was
	  that the NOTIFY would be sent to the private address that the device
	  placed in the Contact header.

	  The reason why res_pjsip_nat did not rewrite the address when it
	  received the 200 OK response was that it could not associate the
	  incoming response with a configured endpoint. This is because on a
	  response, the only way to associate the response to an endpoint is by
	  finding the dialog that the response is associated with and then finding
	  the endpoint that is associated with that dialog. We do not perform
	  endpoint lookups on responses. res_pjsip_pubsub skipped the step of
	  associating the endpoint with the dialog we created, so res_pjsip_nat
	  could not find the associated endpoint and therefore couldn't rewrite
	  the contact.

	  This commit message is like 50x longer than the actual fix.

	  ASTERISK 24981 #close
	  Reported by Mark Michelson

	  Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd
2015-04-16 22:34 +0000 [2f418c052e]  Gareth Palmer <gareth@acsdata.co.nz>

	* New AMI Command Output Format

	  This change modifies how the the output from a CLI command is sent
	  to a client over AMI.

	  Output from the CLI command is now sent as a series of zero-or-more
	  Output: headers.

	  Additionally, commands that fail to execute (eg: no such command,
	  invalid syntax etc.) now cause an Error response instead of Success.

	  If the command executed successfully, but the manager unable to
	  provide the output the reason will be included in the Message:
	  header. Otherwise it will contain 'Command output follows'.

	  Depends on a new version of starpy (> 1.0.2) that supports the new
	  output format.

	  See pull-request https://github.com/asterisk/starpy/pull/34

	  ASTERISK-24730

	  Change-Id: I6718d95490f0a6b3f171c1a5cdad9207f9a44888
2015-04-20 18:00 +0000 [614f506690]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels.

	  The chan_dahdi channel driver is a very old driver.  The ability for it to
	  support ISDN was added well after the initial analog support.  Setting the
	  softhangup flags is a carry over from the original analog code.  The
	  driver was not updated to call ast_queue_hangup() which will post the AMI
	  HangupRequest event.

	  * Changed sig_pri.c to call ast_queue_hangup() instead of setting the
	  softhangup flag when the remote party initiates a hangup.

	  ASTERISK-24895 #close
	  Reported by: Andrew Zherdin

	  Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325

2015-04-20 13:40 +0000 [bff3064578]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr/cdr_adaptive_odbc.c: Refactor concatenate columns name.

	  The concatenate for columns name to INSERT INTO is always the same. It is
	  possible to do it on one line.

	  ASTERISK-24980

	  Change-Id: Ib8bb53c42535378581d4ef729cc5ebbb22b067ac
2015-04-20 09:53 +0000 [06ba1e59cb]  gtjoseph <george.joseph@fairview5.com>

	* pjsip_options:  Fix format specifier for int64_t rtt.

	  Contact status rtt is an int64_t and needs the PRId64 macro to
	  properly create the format specifier on 32-bit systems.

	  Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7

2015-04-18 13:36 +0000 [298faf7c50]  gtjoseph <george.joseph@fairview5.com>

	* pjsip_options:  Fix non-qualified contacts showing as unavailable

	  The "Add qualify_timeout processing and eventing" patch introduced
	  an issue where contacts that had qualify_frequency set to 0 were
	  showing Unavailable instead Unknown.  This patch checks for
	  qualify_frequency=0 and create an "Unknown"  contact_status
	  with an RTT = 0.

	  Previously, the lack of contact_status implied Unknown but since
	  we're now changing endpoint state based on contact_status, I've
	  had to add new UNKNOWN status so that changes could trigger the
	  appropriate contact_status observers.

	  ASTERISK-24977: #close

	  Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7

2015-04-19 15:49 +0000 [8e903b17ea]  Matt Jordan <mjordan@digium.com>

	* main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple

	  When a PBX registrar is unloaded, it will fail to remove its extension from
	  the context root_table if a dialplan application used by that extension is
	  still loaded. This can be the case for AGI, which can be unloaded after several
	  of the standard PBX providers. Often, this is harmless; however, if the
	  extension's priorities are removed during the failed unloading *and* the
	  dialplan application later unregisters, it leaves a ticking timebomb for the
	  next PBX provider that attempts to iterate over the extensions. When that
	  occurs, the peer_table pointer on the extension will already be set to NULL.
	  The current code does not check to see if the pointer is NULL before passing
	  it to a hashtab function this is not NULL tolerant.

	  Since it is possible for the peer_table to be NULL when we normally would not
	  expect that to be the case, the solution in this patch is to simply skip over
	  processing an extension's priorities if peer_table is NULL.

	  Prior to this patch, the tests/pbx/callerid_match test would crash during
	  module unload. With this patch, the test no longer crashes after running.

	  ASTERISK-24774 #close
	  Reported by: Corey Farrell

	  Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40

2015-04-17 18:05 +0000 [1269dd06bc]  Richard Mudgett <rmudgett@digium.com>

	* res_fax: Fix latent bug exposed by ASTERISK-24841 changes.

	  Three fax related tests started failing as a result of changes made for
	  ASTERISK-24841:
	  tests/fax/pjsip/gateway_t38_g711
	  tests/fax/sip/gateway_mix1
	  tests/fax/sip/gateway_mix3

	  Historically, ast_channel_make_compatible() did nothing if the channels
	  were already "compatible" even if they had a sub-optimal translation path
	  already setup.  With the changes from ASTERISK-24841 this is no longer
	  true in order to allow the best translation paths to always be picked.  In
	  res_fax.c:fax_gateway_framehook() code manually setup the channels to go
	  through slin and then called ast_channel_make_compatible().  With the
	  previous version of ast_channel_make_compatible() this was always a
	  no-operation.

	  * Remove call to ast_channel_make_compatible() in fax_gateway_framehook()
	  that now undoes what was just setup when the framehook is attached.

	  * Fixed locking around saving the channel formats in
	  fax_gateway_framehook() to ensure that the formats that are saved are
	  consistent.

	  * Fix copy pasta errors in fax_gateway_framehook() that confuses read and
	  write when dealing with saved channel formats.

	  ASTERISK-24841
	  Reported by: Matt Jordan

	  Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d

2015-04-17 16:19 +0000 [c1d44ff043]  Corey Farrell <git@cfware.com>

	* Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled.

	  When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be
	  called as a function.  This causes a compile error with raw threadstorage as
	  it uses NULL for cleanup.  This fix uses a macro that provides NULL when
	  DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);"
	  with "{};" when DEBUG_THREADLOCALS is enabled.

	  ASTERISK-24975 #close
	  Reported by: Ashley Sanders

	  Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402

2015-04-15 10:38 +0000 [aae45acbda]  Mark Michelson <mmichelson@digium.com>

	* Detect potential forwarding loops based on count.

	  A potential problem that can arise is the following:

	  * Bob's phone is programmed to automatically forward to Carol.
	  * Carol's phone is programmed to automatically forward to Bob.
	  * Alice calls Bob.

	  If left unchecked, this results in an endless loops of call forwards
	  that would eventually result in some sort of fiery crash.

	  Asterisk's method of solving this issue was to track which interfaces
	  had been dialed. If a destination were dialed a second time, then
	  the attempt to call that destination would fail since a loop was
	  detected.

	  The problem with this method is that call forwarding has evolved. Some
	  SIP phones allow for a user to manually forward an incoming call to an
	  ad-hoc destination. This can mean that:

	  * There are legitimate use cases where a device may be dialed multiple
	  times, or
	  * There can be human error when forwarding calls.

	  This change removes the old method of detecting forwarding loops in
	  favor of keeping a count of the number of destinations a channel has
	  dialed on a particular branch of a call. If the number exceeds the
	  set number of max forwards, then the call fails. This approach has
	  the following advantages over the old:

	  * It is much simpler.
	  * It can detect loops involving local channels.
	  * It is user configurable.

	  The only disadvantage it has is that in the case where there is a
	  legitimate forwarding loop present, it takes longer to detect it.
	  However, the forwarding loop is still properly detected and the
	  call is cleaned up as it should be.

	  Address review feedback on gerrit.

	  * Correct "mfgium" to "Digium"
	  * Decrement max forwards by one in the case where allocation of the
	    max forwards datastore is required.
	  * Remove irrelevant code change from pjsip_global_headers.c

	  ASTERISK-24958 #close

	  Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
2015-04-16 10:51 +0000 [56a2baa21d]  Kevin Harwell <kharwell@digium.com>

	* bridge.c: NULL app causes crash during attended transfer

	  Due to a race condition there was a chance that during an attended transfer the
	  channel's application would return NULL. This, of course, would cause a crash
	  when attempting to access the memory. This patch retrieves the channel's app
	  at an earlier time in processing in hopes that the app name is available.
	  However, if it is not then "unknown" is used instead. Since some string value
	  is now always present the crash can no longer occur.

	  ASTERISK-24869 #close
	  Reported by: viniciusfontes
	  Review: https://gerrit.asterisk.org/#/c/133/

	  Change-Id: I5134b84c4524906d8148817719d76ffb306488ac

2015-04-11 17:04 +0000 [c6ed681638]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip: Add global option to limit the maximum time for initial qualifies

	  Currently when Asterisk starts initial qualifies of contacts are spread out
	  randomly between 0 and qualify_timeout to prevent network and system overload.
	  If a contact's qualify_frequency is 5 minutes however, that contact may be
	  unavailable to accept calls for the entire 5 minutes after startup.  So while
	  staggering the initial qualifies is a good idea, basing the time on
	  qualify_timeout could leave contacts unavailable for too long.

	  This patch adds a new global parameter "max_initial_qualify_time" that sets the
	  maximum time for the initial qualifies.  This way you could make sure that all
	  your contacts are initialy, randomly qualified within say 30 seconds but still
	  have the contact's ongoing qualifies at a 5 minute interval.

	  If max_initial_qualify_time is > 0, the formula is initial_interval =
	  min(max_initial_interval, qualify_timeout * random().  If not set,
	  qualify_timeout is used.

	  The default is "0" (disabled).

	  ASTERISK-24863 #close

	  Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
	  Tested-by: George Joseph <george.joseph@fairview5.com>

2015-04-16 13:20 +0000 [664d3263e4]  Scott Griepentrog <scott@griepentrog.com>

	* res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced

	  This change makes the send_notify of the sub_tree
	  not happen when the sub_tree has been deleted due
	  to the notify call failing, which avoids a crash.

	  ASTERISK-24970 #close

	  Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf
2015-04-11 16:56 +0000 [51886c68dc]  gtjoseph <george.joseph@fairview5.com>

	* pjsip_options: Add qualify_timeout processing and eventing

	  This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
	  discussion at
	  http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html

	  The basic issues are that changes in contact status don't cause events to be
	  emitted for the associated endpoint.  Only dynamic contact add/delete actions
	  update the endpoint.  Also, the qualify timeout is fixed by pjsip at 32 seconds
	  which is a long time.

	  This patch makes use of the new transaction timeout feature in r4585 and
	  provides the following capabilities...

	  1.  A new aor/contact variable 'qualify_timeout' has been added that allows the
	  user to specify the maximum time in milliseconds to wait for a response to an
	  OPTIONS message.  The default is 3000ms.  When the timer expires, the contact is
	  marked unavailable.

	  2.  Contact status changes are now propagated up to the endpoint as follows...
	  When any contact is 'Available', the endpoint is marked as 'Reachable'.  When
	  all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'.  The
	  existing endpoint events are generated appropriately.

	  ASTERISK-24863 #close

	  Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
	  Tested-by: Dmitriy Serov
	  Tested-by: George Joseph <george.joseph@fairview5.com>

2015-04-11 16:39 +0000 [ab6382cafd]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip: Refactor endpt_send_request to include transaction timeout

	  This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the
	  discussion at
	  http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html

	  Since we currently have no control over pjproject transaction timeout, this
	  patch pulls the pjsip_endpt_send_request function out of pjproject and into
	  res_pjsip/endpt_send_transaction in order to implement that capability.

	  Now when the transaction is initiated, we also schedule our own pj_timer with
	  our own desired timeout.

	  If the transaction completes before either timeout, pjproject cancels its timer,
	  and calls our tsx callback where we cancel our timer and run the app callback.

	  If the pjproject timer times out first, pjproject calls our tsx callback where
	  we cancel our timer and run the app callback.

	  If our timer times out first, we terminate the transaction which causes
	  pjproject to cancel its timer and call our tsx callback where we run the app
	  callback.

	  Regardless of the scenario, pjproject is calling the tsx callback inside the
	  group_lock and there are checks in the callback to make sure it doesn't run
	  twice.

	  As part of this patch ast_sip_send_out_of_dialog_request was created to replace
	  its similarly named private function.  It takes a new timeout argument in
	  milliseconds (<= 0 to disable the timeout).

	  ASTERISK-24863 #close
	  Reported-by: George Joseph <george.joseph@fairview5.com>
	  Tested-by: George Joseph <george.joseph@fairview5.com>

	  Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747
2015-04-15 16:08 +0000 [043c38f6de]  gtjoseph <george.joseph@fairview5.com>

	* More .gitignore updates

	  Added .pyc and .sha1 to the top-level .gitignore.

	  Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a
	  Tested-by: George Joseph <george.joseph@fairview5.com>

2015-04-14 02:36 +0000 [abf10a1d4c]  Corey Farrell <git@cfware.com>

	* Build System: Enable use of ~/.asterisk.makeopts and /etc/asterisk.makeopts.

	  The Makefile claims that you can set default menuselect options by creating
	  ~/.asterisk.makeopts or /etc/asterisk.makeopts, but they are never read.
	  The rule for menuselect.makeopts is only allowed to run if the active target
	  is 'menuselect', but the menuselect target doesn't depend on
	  menuselect.makeopts.  A dot (wildcard character) was added so the rule will
	  be active for the targets that cause it to run: nmenuselect, cmenuselect,
	  and gmenuselect.

	  ASTERISK-13271 #close
	  Reported by: John Nemeth

	  Change-Id: Ibde804ff196283def49ccb9432fbf224a22586e2
2015-04-13 08:47 +0000 [a3cec44a0a]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Add external PJSIP resolver implementation using core DNS API.

	  This change adds the following:

	  1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked.
	  2. Unit tests for the query set implementation.
	  3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups.

	  For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV
	  are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A,
	  with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit
	  transport has been provided. Configured transports on the system are taken into account to
	  eliminate resolved addresses which have no hope of completing.

	  ASTERISK-24947 #close
	  Reported by: Joshua Colp

	  Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e

2015-04-14 13:16 +0000 [33a319ae73]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cel_pgsql: Fix name string for log on unable allocate memory.

	  The LOG_ERROR has reference to CDR instead of CEL  for LENGTHEN_BUF1 and
	  LENGTHEN_BUF2.

	  ASTERISK-24965 #close
	  Reported by: Rodrigo Ramirez Norambuena

	  Change-Id: Icc818697d7d66d34bfe3048cdd15ca2b06c89744
2015-04-14 15:59 +0000 [f89481e39c]  Corey Farrell <git@cfware.com>

	* test_astobj2_weaken: Fix source file registration.

	  Update test_astobj2_weaken to use the new AST_REGISTER_FILE macro.

	  Change-Id: Ieedadf16610f2e042f393e0501a36447cd07f83d

2015-04-13 05:28 +0000 [62508d6891]  Corey Farrell <git@cfware.com>

	* Build System: Create Makefile macro MOD_ADD_SOURCE.

	  This new macro allows a single line to add all additional
	  sources to a module.  This helps prevent modules from
	  missing steps, and makes future changes easier since
	  they can be made in a single place.

	  ASTERISK-24960 #close
	  Reported by: Corey Farrell

	  Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b

2015-04-12 09:08 +0000 [23a180cade]  Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>

	* cdr_pgsql: Fix CLI "cdr show pgsql status" command.

	  The command always showed the usage information.

	  * Fix the error in command validation for CLI_SHOWUSAGE.

	  ASTERISK-24959 #close
	  Reported by: Rodrigo Ramirez Norambuena

	  Change-Id: I584f0936bb01001336a468a55c1d05d79fe795d5
2015-04-13 19:06 +0000 [bf46ef35ca]  gtjoseph <george.joseph@fairview5.com>

	* .gitignore updates for master/13

	  Added products of ./bootstrap

	  Added nmenuselect and gmenuselect to menuselect/

	  Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35

2015-04-13 06:52 +0000 [62e95065d6]  Corey Farrell <git@cfware.com>

	* AMI: Fix improper handling of lines that are exactly 1025 bytes long.

	  When AMI receives a line that is 1025 bytes long, it sends two error
	  messages.  Copy the last byte in the buffer to the first postiion,
	  set the length to 1.

	  ASTERISK-20524 #close
	  Reported by: David M. Lee

	  Change-Id: Ifda403e2713b59582c715229814fd64a0733c5ea

2015-04-12 03:22 +0000 [cb6bf3094e]  Corey Farrell <git@cfware.com>

	* astobj2: Add support for weakproxy objects.

	  This implements "weak" references.  The weakproxy object is a real ao2 with
	  normal reference counting of its own.  When a weakproxy is pointed to a normal
	  object they hold references to each other.  The normal object is automatically
	  freed when a single reference remains (the weakproxy).  The weakproxy also
	  supports subscriptions that will notify callbacks when it does not point
	  to any real object.

	  ASTERISK-24936 #close
	  Reported by: Corey Farrell

	  Change-Id: Ib9f73c02262488d314d9d9d62f58165b9ec43c67

2015-04-13 14:41 +0000 [a573b77f78]  David M. Lee <dlee@respoke.io>

	* Fixing extconf compile

	  During the mass code deletion for clang support, a stray backslash was
	  left behind that was causing utils to fail to compile.

	  Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1

2015-04-13 09:54 +0000 [3f9aa29945]  Matt Jordan <mjordan@digium.com>

	* build_tools/make_version: Update version parsing for Git migration

	  External systems - such as the Asterisk Test Suite - require knowledge of the
	  upstream branch. Unfortunately, after moving to Git, the Asterisk version
	  currently consists of only a 'GIT" prefix followed by an object blob,
	  e.g., GIT-as08d7. This makes it difficult for such systems to know what
	  features are available in a particular check out of Asterisk.

	  This patch fixes this by hardcoding the branch in a variable in the
	  make_version script. Since the mainline branches are not changed often -
	  typically only once a year - this is a reasonable approach to solving
	  the problem, and is more reliable than parsing the output of 'git branch
	  -vv'. Branches that track off of an upstream primary branch will then get the
	  benefit of knowing which mainline branch they are currently based off
	  of.

	  ASTERISK-24954 #close

	  Change-Id: I8090d5d548b6d19e917157ed530b914b7eaf9799

2015-04-13 05:57 +0000 [fbc8ddfe63]  Corey Farrell <git@cfware.com>

	* Optional API: Fix handling of sources that are both provider and user.

	  OPTIONAL_API has conditionals to define AST_OPTIONAL_API and
	  AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined.
	  Unfortunately this is inside the include protection block, so only the
	  first status of AST_API_MODULE is respected.  For example res_monitor
	  is an optional API provider, but uses func_periodic_hook.  This makes
	  func_periodic_hook non-optional to res_monitor.

	  This changes optional_api.h so that AST_OPTIONAL_API and
	  AST_OPTIONAL_API_ATTR is redefined every time the header is included.

	  ASTERISK-17608 #close
	  Reported by: Warren Selby

	  Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679

2015-04-11 21:38 +0000 [4a58261694]  Matt Jordan <mjordan@digium.com>

	* git migration: Refactor the ASTERISK_FILE_VERSION macro

	  Git does not support the ability to replace a token with a version
	  string during check-in. While it does have support for replacing a
	  token on clone, this is somewhat sub-optimal: the token is replaced
	  with the object hash, which is not particularly easy for human
	  consumption. What's more, in practice, the source file version was often
	  not terribly useful. Generally, when triaging bugs, the overall version
	  of Asterisk is far more useful than an individual SVN version of a file. As a
	  result, this patch removes Asterisk's support for showing source file
	  versions.

	  Specifically, it does the following:

	  * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
	    remove passing the version in with the macro. Other facilities
	    than 'core show file version' make use of the file names, such as
	    setting a debug level only on a specific file. As such, the act of
	    registering source files with the Asterisk core still has use. The
	    macro rename now reflects the new macro purpose.

	  * main/asterisk:
	    - Refactor the file_version structure to reflect that it no longer
	      tracks a version field.
	    - Remove the "core show file version" CLI command. Without the file
	      version, it is no longer useful.
	    - Remove the ast_file_version_find function. The file version is no
	      longer tracked.
	    - Rename ast_register_file_version/ast_unregister_file_version to
	      ast_register_file/ast_unregister_file, respectively.

	  * main/manager: Remove value from the Version key of the ModuleCheck
	    Action. The actual key itself has not been removed, as doing so would
	    absolutely constitute a backwards incompatible change. However, since
	    the file version is no longer tracked, there is no need to attempt to
	    include it in the Version key.

	  * UPGRADE: Add notes for:
	    - Modification to the ModuleCheck AMI Action
	    - Removal of the "core show file version" CLI command

	  Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e

2015-04-12 06:12 +0000 [5d34bce635]  Corey Farrell <git@cfware.com>

	* main/editline: Add .gitignore.

	  This patch adds a .gitignore for main/editline to ignore all build results.

	  Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d

2015-04-11 23:22 +0000 [d6605b3c10]  Matt Jordan <mjordan@digium.com>

	* .gitignore: Ignore tarballs (*.gz)

	  This patch updates the root .gitignore file to ignore files with a .gz
	  extension. This will cause git to ignore downloaded sound tarballs in
	  the the sounds/ directory.

	  Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59

2015-04-11 13:20 +0000 [b35e184d41]  gtjoseph <george.joseph@fairview5.com>

	* Add .gitignore and .gitreview files

	  Add the .gitignore and .gitreview files to the asterisk repo.

	  NB:  You can add local ignores to the .git/info/exclude file
	  without having to do a commit.

	  Common ignore patterns are in the top-level .gitignore file.
	  Subdirectory-specific ignore patterns are in their own .gitignore
	  files.

	  Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69
	  Tested-by: George Joseph

2015-04-11 10:27 +0000 [356b770632]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix various warnings for tests

	  This patch fixes a variety of clang compiler warnings for unit tests. This
	  includes autological comparison issues, ignored return values, and
	  interestingly enough, one embedded function. Fun!

	  Review: https://reviewboard.asterisk.org/r/4555

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4555.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434705 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434706 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-11 10:11 +0000 [5f181bcccd]  Juergen Spies (License 6698)

	* res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram

	  Prior to this patch, the far_max_datagram value on the UDPTL structure would
	  remain -1 if the remote endpoint fails to provide the SDP media attribute
	  T38FaxMaxDatagram. This can result in the INVITE request being rejected. With
	  this patch, we will now properly initialize the value with either the default
	  value or with the value provided by pjsip.conf's t38_udptl_maxdatagram
	  parameter.

	  Review: https://reviewboard.asterisk.org/r/4589

	  ASTERISK-24928 #close
	  Reported by: Juergen Spies
	  Tested by: Juergen Spies
	  patches:
	    pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698)
	  ........

	  Merged revisions 434688 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 18:37 +0000 [c499cabf53]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices.

	  With this patch, chan_pjsip/res_pjsip now sets the native formats to the
	  codecs negotiated by a call.

	  * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native
	  formats to include all the negotiated audio codecs instead of only the
	  initial preferred audio codec and later the currently received audio
	  codec.

	  * The audio frame handling in channel.c:ast_read() is more streamlined and
	  will automatically adjust to changes in received frame formats.  The new
	  policy is to remove translation and pass the new frame format to the
	  receiver except if the translation was to a signed linear format.  A more
	  long winded version is commented in ast_read() along with some caveats.

	  * The audio frame handling in channel.c:ast_write() is more streamlined
	  and will automatically adjust any needed translation to changes in the
	  frame formats sent.  Frame formats sent can change for many reasons such
	  as a recording is being played back or the bridged peer changed the format
	  it sends.  Since it is a normal expectation that sent formats can change,
	  the codec mismatch warning message is demoted to a debug message.

	  * Removed the short circuit check in
	  channel.c:ast_channel_make_compatible_helper().  Two party bridges need to
	  make channels compatible with each other.  However, transfers and moving
	  channels among bridges can result in otherwise compatible channels having
	  sub-optimal translation paths if the make compatible check is short
	  circuited.  A result of forcing the reevaluation of channel compatibility
	  is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc
	  options take effect consistently now.  It is unfortunate that these two
	  options are enabled by default and negate some of the benefits to the
	  changes in channel.c:ast_read() by forcing translation through signed
	  linear on a two party bridge.

	  * Improved the softmix bridge technology to better control the translation
	  of frames to the bridge.  All of the incoming translation is now normally
	  handled by ast_read() instead of splitting any translation steps between
	  ast_read() and the slin factory.  If any frame comes in with an unexpected
	  format then the translation path in ast_read() is updated for the next
	  frame and the slin factory handles the current frame translation.

	  This is the final patch in a series of patches aimed at improving
	  translation path choices.  The other patches are on the following reviews:
	  https://reviewboard.asterisk.org/r/4600/
	  https://reviewboard.asterisk.org/r/4605/

	  ASTERISK-24841 #close
	  Reported by: Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4609/
	  ........

	  Merged revisions 434671 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434672 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 16:06 +0000 [66f3fd0028]  Kevin Harwell <kharwell@digium.com>

	* chan_sip: make progressinband default to no

	  After the "progressinband" value setting of "never" was updated to never send a
	  183 this separated its use from the "no" value. Since "never" was the default,
	  but most users probably expect "no" this patch updates the default for the
	  "progressinband" setting to "no."

	  ASTERISK-24835 #close
	  Reported by: Andrew Nagy
	  Review: https://reviewboard.asterisk.org/r/4606/
	  ........

	  Merged revisions 434654 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434655 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 12:56 +0000 [8bae18ab93]  yaron nahum (License 6676)

	* res_pjsip: Add an 'auto' option for DTMF Mode

	  This patch adds support for automatically detecting the type of DTMF that a
	  PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
	  the channel created for an endpoint will attempt to determine if RFC 4733
	  DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
	  for the channel will be set to inband.

	  Review: https://reviewboard.asterisk.org/r/4438

	  ASTERISK-24706 #close
	  Reported by: yaron nahum
	  patches:
	    yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
	  ........

	  Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 12:00 +0000 [f69e46de25]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard: Cleanup load unload

	  While investigating other unload issues I realized that the load/unload process 
	  for the config wizard was pretty ugly so I've refactored it as follows...

	  When the res_pjsip sorcery instance is created the config_wizard bumps it's own 
	  module reference to prevent it from unloading while the sorcery instance is 
	  still active.  When res_pjsip unloads and it's sorcery instance is destroyed, 
	  the config wizard unrefs itself which then allows itself to unload cleanly.  
	  Since the config wizard now can't load after res_pjsip or unload before it 
	  (which should have been the correct behavior all along), I was able to remove 
	  the chunks of code in both load_module and unload_module that handled that case.

	  Ran the testsuite tests to insure there were no functional changes and REF_DEBUG 
	  to insure that Asterisk was shutting down cleanly with no FRACKs or leaks.

	  Tested-by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/4610/
	  ........

	  Merged revisions 434619 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434620 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 11:38 +0000 [6f1a7fe05f]  Richard Mudgett <rmudgett@digium.com>

	* bridge_softmix.c,channel.c: Minor code simplification and cleanup.

	  * Made code easier to follow in bridge_softmix.c:analyse_softmix_stats()
	  and made some debug messages more helpful.

	  * Made some debug and warning messages more helpful in
	  channel.c:set_format().

	  Review: https://reviewboard.asterisk.org/r/4607/
	  ........

	  Merged revisions 434617 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434618 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 11:32 +0000 [0b805cb875]  Richard Mudgett <rmudgett@digium.com>

	* translate.c: Only select audio codecs to determine the best translation choice.

	  Given a source capability of h264 and ulaw, a destination capability of
	  h264 and g722 then ast_translator_best_choice() would pick h264 as the
	  best choice even though h264 is a video codec and Asterisk only supports
	  translation of audio codecs.  When the audio starts flowing, there are
	  warnings about a codec mismatch when the channel tries to write a frame to
	  the peer.

	  * Made ast_translator_best_choice() only select audio codecs.

	  * Restore a check in channel.c:set_format() lost after v1.8 to prevent
	  trying to set a non-audio codec.

	  This is an intermediate patch for a series of patches aimed at improving
	  translation path choices for ASTERISK-24841.

	  This patch is a complete enough fix for ASTERISK-21777 as the v11 version
	  of ast_translator_best_choice() does the same thing.  However, chan_sip.c
	  still somehow tries to call ast_codec_choose() which then calls
	  ast_best_codec() with a capability set that doesn't contain any audio
	  formats for the incoming call.  The remaining warning message seems to be
	  a benign transient.

	  ASTERISK-21777 #close
	  Reported by: Nick Ruggles

	  ASTERISK-24380 #close
	  Reported by: Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4605/
	  ........

	  Merged revisions 434614 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434615 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434616 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 09:56 +0000 [894153b8b1]  Matt Jordan <mjordan@digium.com>

	* res/ari: Fix model validation for ChannelHold event

	  When the ChannelHold event was added, the 'musicclass' parameter was
	  erroneously removed. This caused the ChannelHold events to be rejected as
	  they failed model validation. This patch updates the Swagger schema such that
	  it now properly reflects the event that is being created.

	  Hooray for tests that catch things like this.
	  ........

	  Merged revisions 434597 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434598 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 08:32 +0000 [02a0a4d65f]  Joshua Colp <jcolp@digium.com>

	* dns: Fix build when TEST_FRAMEWORK is not defined.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434583 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 07:40 +0000 [80c443bea4]  Y Ateya (License 6693)

	* channels/chan_iax2: Improve POKE expiration time calculation for lossy networks

	  POKE is used to check for peer availability; however, in networks with packet
	  loss, the current calculations may result in POKE expiration times that are too
	  short. This patch alters the expiration/retry time logic to take into account
	  the last known qualify round trip time, as opposed to always using a static
	  value for each peer.

	  Review: https://reviewboard.asterisk.org/r/4536

	  ASTERISK-22352 #close
	  Reported by: Frederic Van Espen

	  ASTERISK-24894 #close
	  Reported by: Y Ateya
	  patches:
	    poke_noanswer_duration.diff submitted by Y Ateya (License 6693)
	  ........

	  Merged revisions 434564 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434565 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-10 07:23 +0000 [b3d01f1fbf]  Y Ateya (License 6693)

	* channels/chan_iax2: Add a configuration parameter for call token expiration

	  This patch adds a new configuration parameter, 'calltokenexpiration', that
	  controls how long before an authentication call token is expired. The default
	  maintains the RFC specified 10 seconds. Setting it to a higher value may be
	  useful in lossy networks.

	  Review: https://reviewboard.asterisk.org/r/4588

	  ASTERISK-24939 #close
	  Reported by: Y Ateya
	  patches:
	    ctoken_configuration.diff submitted by Y Ateya (License 6693)


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 18:12 +0000 [ed6b6e3c03]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_phoneprov_provider: Fix reference leak on unload

	  res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to 
	  a missing OBJ_NODATA in an ao2_callback in load_users().  Rather than adding the 
	  OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator.  
	  This plugged the leak but exposed an unload order issue between 
	  res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip.

	  res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip.  
	  Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it 
	  unloads, it's objects are still in the sorcery instance.  When res_pjsip 
	  unloads, it destroys all its objects including res_pjsip_phoneprov_provider's.  
	  The phoneprov destructor then attempts to unregister the extension from 
	  res_phoneprov but because res_phoneprov is already cleaned up, its users 
	  container is gone and we get a FRACK.

	  Simple solution, check for the NULL users container before attempting to remove 
	  the entry. Duh.

	  Ran tests/res_phoneprov/res_phoneprov_provider.  No leaks in 
	  res_pjsip_phoneprov_provider and no FRACKs.

	  Reported-by: Corey Farrell
	  Tested-by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/4608/
	  ASTERISK-24935 #close
	  ........

	  Merged revisions 434545 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434547 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 18:08 +0000 [9a63ada03a]  gtjoseph <george.joseph@fairview5.com>

	* loader/main: Don't set ast_fully_booted until deferred reloads are processed

	  Until we have a true module management facility it's sometimes necessary for one 
	  module to force a reload on another before its own load is complete.  If 
	  Asterisk isn't fully booted yet, these reloads are deferred.  The problem is 
	  that asterisk reports fully booted before processing the deferred reloads which 
	  means Asterisk really isn't quite ready when it says it is.

	  This patch moves the report of fully booted after the processing of the deferred 
	  reloads is complete.

	  Since the pjsip stack has the most number of related modules, I ran the 
	  channels/pjsip testsuite to make sure there aren't any issues.  All tests 
	  passed.

	  Tested-by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/4604/
	  ........

	  Merged revisions 434544 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 17:07 +0000 [520b9f2174]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: add CLI command to show global and system configuration

	  Added a new CLI command for res_pjsip that shows both global and system
	  configuration settings: pjsip show settings

	  ASTERISK-24918 #close
	  Reported by: Scott Griepentrog
	  Review: https://reviewboard.asterisk.org/r/4597/
	  ........

	  Merged revisions 434527 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434528 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 11:09 +0000 [b2b1f24af6]  Richard Mudgett <rmudgett@digium.com>

	* chan_iax2.c: Fix ref leak in iax2_request().

	  * Increased warning message format capability string buffer size in
	  iax2_request().

	  Review: https://reviewboard.asterisk.org/r/4601/
	  ........

	  Merged revisions 434510 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434511 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 11:05 +0000 [459171be12]  Richard Mudgett <rmudgett@digium.com>

	* bridge_native_rtp.c: Defer allocation and check if it fails in native_rtp_bridge_compatible().

	  Review: https://reviewboard.asterisk.org/r/4601/
	  ........

	  Merged revisions 434508 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434509 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 10:43 +0000 [3ef0a17b1f]  yaron nahum (License 6676)

	* res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests

	  This patch adds a new session supplement that handles in-dialog OPTIONS
	  requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup
	  for the OPTIONS request would already have been done by the time the
	  session supplement receives the inbound request.

	  ASTERISK-24862 #close
	  Reported by: yaron nahum
	  patches:
	    res_pjsip_dlg_options.c submitted by yaron nahum (License 6676)
	  ........

	  Merged revisions 434506 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 09:58 +0000 [c08ebc6eeb]  Mark Michelson <mmichelson@digium.com>

	* Reduce duplication of common DNS code.

	  The NAPTR and SRV branches were worked on independently and
	  resulted in some code being duplicated in each. Since both
	  have been merged into trunk now, this patch reduces the
	  duplication by factoring out common code into its own
	  source files.



	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-09 07:57 +0000 [ea0098724e]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix autological comparisons

	  This fixes autological comparison warnings in the following:
	   * chan_skinny: letohl may return a signed or unsigned value, depending on the
	     macro chosen
	   * func_curl: Provide a specific cast to CURLoption to prevent mismatch
	   * cel: Fix enum comparisons where the enum can never be negative
	   * enum: Fix comparison of return result of dn_expand, which returns a signed
	     int value
	   * event: Fix enum comparisons where the enum can never be negative
	   * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be
	     negative
	   * presencestate: Use the actual enum value for INVALID state
	   * security_events: Fix enum comparisons where the enum can never be negative
	   * udptl: Don't bother to check if the return value from encode_length is less
	     than 0, as it returns an unsigned int
	   * translate: Since the parameters are unsigned int, don't bother checking
	     to see if they are negative. The cast to unsigned int would already blow
	     past the matrix bounds.
	   * res_pjsip_exten_state: Use a temporary value to cache the return of
	     ast_hint_presence_state
	   * res_stasis_playback: Fix enum comparisons where the enum can never be
	     negative
	   * res_stasis_recording: Add an enum value for the case where the recording
	     operation is in error; fix enum comparisons
	   * resource_bridges: Use enum value as opposed to -1
	   * resource_channels: Use enum value as opposed to -1

	  Review: https://reviewboard.asterisk.org/r/4533
	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4533.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434470 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 21:05 +0000 [2201e27340]  Stefan Engström (License 6691)

	* apps/app_queue: Prevent possible crash when evaluating queue penalty rules

	  Although it only occurred once, a crash occurred when a queue attempted to
	  evaluate a queue penalty rule that appeared to have already been destroyed.
	  In many locations in app_queue, a test is done to see if qe->pr is NULL;
	  however, when we dispose of a queue's penalty rules, we don't set the pointer
	  to NULL after free'ing it. This patch does that to prevent any dangling
	  pointers from lingering on the queue object.

	  Review: https://reviewboard.asterisk.org/r/4522

	  ASTERISK-23319 #close
	  Reported by: Vadim
	  patches:
	    rb4552.patch submitted by Stefan Engström (License 6691)
	  ........

	  Merged revisions 434448 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434449 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434450 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 13:32 +0000 [a759714101]  Jonathan Rose <jrose@digium.com>

	* res_pjsip_t38: Fix FAX failures when using PJSIP with authentication

	  Without this patch, if a PJSIP endpoint with udptl enabled and authentication
	  set attempted to use sendFax, the FAX session would fail during setup. This
	  was because the invite issued in response to being auth challenged would cause
	  the PJSIP channel performing the FAX to receive a second T38 framehook and
	  this would cause frames to be consumed in an inappropriate manner.

	  ASTERISK-24933 #close
	  Reported by: Jonathan Rose
	  Review: https://reviewboard.asterisk.org/r/4577/
	  ........

	  Merged revisions 434425 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434431 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 13:20 +0000 [09df34d880]  Richard Mudgett <rmudgett@digium.com>

	* Bridging: Eliminate the unnecessary make channel compatible with bridge operation.

	  When a channel enters the bridging system it is first made compatible with
	  the bridge and then the bridge technology makes the channel compatible
	  with the technology.  For all but the DAHDI native and softmix bridge
	  technologies the make channel compatible with the bridge step is an
	  effective noop because the other technologies allow all audio formats.
	  For the DAHDI native bridge technology it doesn't matter because it is not
	  an initial bridge technology and chan_dahdi allows only one native format
	  per channel.  For the softmix bridge technology, it is a noop at best and
	  harmful at worst because the wrong translation path could be setup if the
	  channel's native formats allow more than one audio format.

	  This is an intermediate patch for a series of patches aimed at improving
	  translation path choices.

	  * Removed code dealing with the unnecessary step of making the channel
	  compatible with the bridge.

	  ASTERISK-24841
	  Reported by: Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4600/
	  ........

	  Merged revisions 434424 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434430 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 11:49 +0000 [8ec9a82b9a]  Maciej Szmigiero <mail@maciej.szmigiero.name> (license 6085)

	* Security/tcptls: MitM Attack potential from certificate with NULL byte in CN.

	  When registering to a SIP server with TLS, Asterisk will accept CA signed
	  certificates with a common name that was signed for a domain other than the
	  one requested if it contains a null character in the common name portion of
	  the cert. This patch fixes that by checking that the common name length
	  matches the the length of the content we actually read from the common name
	  segment. Some certificate authorities automatically sign CA requests when
	  the requesting CN isn't already taken, so an attacker could potentially
	  register a CN with something like www.google.com\x00www.secretlyevil.net
	  and have their certificate signed and Asterisk would accept that certificate
	  as though it had been for www.google.com - this is a security fix and is
	  noted in AST-2015-003.

	  ASTERISK-24847 #close
	  Reported by: Maciej Szmigiero
	  Patches:
	   asterisk-null-in-cn.patch submitted by mhej (license 6085)
	  ........

	  Merged revisions 434337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 434338 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434384 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434385 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 11:31 +0000 [2bd9e008a7]  Richard Mudgett <rmudgett@digium.com>

	* format_cache.c: Add missing slin12 format to ast_format_cache_is_slinear().
	  ........

	  Merged revisions 434357 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434383 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 07:02 +0000 [3f54af689f]  Matt Jordan <mjordan@digium.com>

	* chan_iax2: Fix compilation issue due to funky merge

	  Don't mix declarations and code!


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434294 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 07:00 +0000 [a9b6a62461]  Jaco Kroon (License 5671)

	* chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno]

	  This patch fixes an access to the peer callnumber that is unprotected by a
	  corresponding mutex. The peer->callno value can be changed by multiple threads,
	  and all data inside the iaxs array must be procted by a corresponding lock
	  of iaxsl.

	  The patch moves the unprotected access to a location where the mutex is
	  safely obtained.

	  Review: https://reviewboard.asterisk.org/r/4599/

	  ASTERISK-21211 #close
	  Reported by: Jaco Kroon
	  patches:
	    asterisk-11.2.1-iax2_poke-segfault.diff submitted by Jaco Kroon (License 5671)
	  ........

	  Merged revisions 434291 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434292 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434293 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 06:54 +0000 [477536ef25]  Valentin Vidić (License 6697)

	* chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled

	  When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will
	  attempt to handle both IPv4 and IPv6 addresses, although the information will
	  be stored in a struct with an AF_INET6 address type. However, the current
	  NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly.
	  This patch adds an additional check for the mapped address case, allowing
	  the NAT code to handle clients even when the address is IPv6.

	  Review: https://reviewboard.asterisk.org/r/4563/

	  ASTERISK-18032 #close
	  Reported by: Christoph Timm
	  patches:
	    nat_with_ipv6.diff submitted by Valentin Vidić (License 6697)
	  ........

	  Merged revisions 434288 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434289 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434290 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 06:45 +0000 [b8fa8aa775]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix pointer-bool-converesion warnings

	  This patch fixes several warnings pointed out by the clang compiler.
	  * chan_pjsip: Removed check for data->text, as it will always be non-NULL.
	  * app_minivm: Fixed evaluation of etemplate->locale, which will always
	    evaluate to 'true'. This patch changes the evaluation to use
	    ast_strlen_zero.
	  * app_queue:
	    - Fixed evaluation of qe->parent->monfmt, which always evaluates to
	      true. Instead, we just check to see if the dereferenced pointer
	      evaluates to true.
	    - Fixed evaluation of mem->state_interface, wrapping it with a call to
	      ast_strlen_zero.
	  * res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.

	  Review: https://reviewboard.asterisk.org/r/4541

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4541.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434286 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-08 06:35 +0000 [016fba12e2]  Rodrigo Ramirez Norambuena (License 6577)

	* cel_pgsl: Add support for GMT timestamps

	  This patch adds a new option to cel_pgsl, "usegmtime", which causes timestamps
	  to be logged in GMT.

	  Review: https://reviewboard.asterisk.org/r/4571/

	  ASTERISK-23186 #close
	  Reported by: Rodrigo Ramirez Norambuena
	  patches:
	    cel_pgsql.c_add_usegmtime2.patch submitted by Rodrigo Ramirez Norambuena (License 6577)



	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-07 14:40 +0000 [d923ec80b9]  Scott Griepentrog <sgriepentrog@digium.com>

	* pjsip: resolve compatibility problem with ast_sip_session

	  A change in r430179 inserted a variable near the top of a
	  structure caused a problem when running DPMA in a version
	  of Asterisk compiled across the change.  This patch moves
	  the new variable to the end of the structure, eliminating
	  the problem.

	  Review: https://reviewboard.asterisk.org/r/4574/
	  ........

	  Merged revisions 433944 from http://svn.asterisk.org/svn/asterisk/branches/13
	  ........

	  Merged revisions 434261 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-07 11:42 +0000 [153c4044e4]  Kevin Harwell <kharwell@digium.com>

	* bridge.c: Hangup attended transfer target after it has been swapped out

	  After completing an attended transfer the transfer target channel (the one that
	  gets swapped out) was not being hung up after leaving the bridge. This resulted
	  in a channel possibly being left around. Added an explicit softhangup for the
	  channel in question after the transfer is successfully completed in order to
	  make sure the channel is hung up.

	  ASTERISK-24782 #close
	  Reported by: John Bigelow
	  Review: https://reviewboard.asterisk.org/r/4575/
	  ........

	  Merged revisions 434240 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434241 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-07 10:34 +0000 [1eba6abae5]  Mark Michelson <mmichelson@digium.com>

	* Do not queue message requests that we do not respond to.

	  If we receive a MESSAGE request that we cannot send a response
	  to, we should not send the incoming MESSAGE to the dialplan.

	  This commit should help the bouncing message_retrans test to
	  pass consistently.
	  ........

	  Merged revisions 434218 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-07 10:22 +0000 [c2f50ba6f4]  Matt Jordan <mjordan@digium.com>

	* ARI: Add the ability to intercept hold and raise an event

	  For some applications - such as SLA - a phone pressing hold should not behave
	  in the fashion that the Asterisk core would like it to. Instead, the hold
	  action has some application specific behaviour associated with it - such as
	  disconnecting the channel that initiated the hold; only playing MoH to channels
	  in the bridge if the channels are of a particular type, etc.

	  One way of accomplishing this is to use a framehook to intercept the
	  hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
	  accomplishes that using a new dialplan function, HOLD_INTERCEPT.

	  In addition, some general cleanup of raising hold/unhold Stasis messages was
	  done, including removing some RAII_VAR usage.

	  Review: https://reviewboard.asterisk.org/r/4549/

	  ASTERISK-24922 #close
	  ........

	  Merged revisions 434216 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434217 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 21:10 +0000 [af4d802773]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix sometimes-initialized warning in func_math

	  This patch fixes a bug in a unit test in func_math where a variable could be
	  passed to ast_free that wasn't allocated. This patch corrects the issue and
	  ensures that we only attempt to free a variable if we previously allocated
	  it.

	  Review: https://reviewboard.asterisk.org/r/4552

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4552.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434190 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434191 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 21:03 +0000 [c1cfe3fae2]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix non-literal-null-conversion warnings

	  Clang will flag errors when a char pointer is set to '\0', as opposed to a
	  value that the char pointer points to. This patch fixes this warning
	  in a variety of locations.

	  Review: https://reviewboard.asterisk.org/r/4551

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4551.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434187 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434188 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434189 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 16:54 +0000 [79fb8c32a6]  Mark Michelson <mmichelson@digium.com>

	* Uncomment test case.



	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434170 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 16:13 +0000 [fc314cb43f]  Mark Michelson <mmichelson@digium.com>

	* Add missing DNS NAPTR test file.



	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434154 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 14:23 +0000 [87d7c90e4e]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: config option 'timers' can't be set to 'no'

	  When setting the configuration option 'timers' equal to 'no' the bit flag was
	  not properly negated. This patch clears all associated flags and only sets the
	  specified one. pjsip will handle any necessary flag combinations. Also went
	  ahead and did similar for the '100rel' option.

	  ASTERISK-24910 #close
	  Reported by: Ray Crumrine
	  Review: https://reviewboard.asterisk.org/r/4582/
	  ........

	  Merged revisions 434131 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 14:04 +0000 [e48f2e7897]  gtjoseph <george.joseph@fairview5.com>

	* build: Fixes for gcc 5 compilation

	  These are fixes for compilation under gcc 5.0...

	  chan_sip.c:    In parse_request needed to make 'lim' unsigned.
	  inline_api.h:  Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99 
	                 inline semantics (same as clang).
	  ccss.c:        In ast_cc_set_parm, needed to fix weird comparison.
	  dsp.c:         Needed to work around a possible compiler bug.  It was throwing 
	                 an array-bounds error but neither
	                 sgriepentrog, rmudgett nor I could figure out why.
	  manager.c:     In action_atxfer, needed to correct an array allocation.

	  This patch will go to 11, 13, trunk.

	  Review: https://reviewboard.asterisk.org/r/4581/
	  Reported-by: Jeffrey Ollie
	  Tested-by: George Joseph
	  ASTERISK-24932 #close
	  ........

	  Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434114 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434115 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 13:18 +0000 [0543879228]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Remove large chunks of unused code from extconf

	  This patch fixes a warning caught by clang, in which it detected that large
	  chunks of extconf were unused. Frankly, I wish we could pretend that all of
	  extconf was unused, but alas, that is not yet the case.

	  A few extraneous functions in the parking tests were removed as well, for
	  the same reason.

	  Review: https://reviewboard.asterisk.org/r/4553

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4553.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434093 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434097 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434099 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 13:03 +0000 [e309a91e2d]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix sometimes-uninitialized warning in pbx_config

	  This patch fixes a warning caught by clang, in which a char pointer could be
	  assigned to before it was initialized. The patch re-organizes the code to
	  ensure that the pointer is always initialized, even on off nominal paths.

	  Review: https://reviewboard.asterisk.org/r/4529

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4529.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434090 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434091 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 12:52 +0000 [ed3cf8761b]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix format specified in framehook

	  This patch fixes an invalid format specifier used in the formatting of an
	  ERROR message in the framehook code. The format specifier specifies a
	  type of 'unsigned short', but the argument passed to it is of type 'int'.
	  The patch changes the format specifier to 'i'.

	  Review: https://reviewboard.asterisk.org/r/4540

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4535.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 434087 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 434088 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 12:05 +0000 [0a26602b8c]  Mark Michelson <mmichelson@digium.com>

	* Merge NAPTR support into trunk.

	  This adds NAPTR record allocation and sorting, as well as
	  unit tests that verify that NAPTR records are parsed and
	  sorted correctly.

	  Review: https://reviewboard.asterisk.org/r/4542



	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434068 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 11:02 +0000 [edf9da4365]  Mark Michelson <mmichelson@digium.com>

	* Ensure that a non-zero sample rate is returned for all formats.

	  Versions of Asterisk prior to 12 defaulted to 8000 as a sample rate
	  if one was not provided by a format. In Asterisk 13, this was removed.
	  The result was that some calculations which involve dividing by the
	  sample rate resulted in dividing by 0. The fix being put in place
	  here is to have the same default fallback that was present in previous
	  versions of Asterisk.

	  Asterisk-24914 #close
	  Reported by Marcello Ceschia
	  ........

	  Merged revisions 434046 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434047 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 10:17 +0000 [ffd7319df3]  Corey Farrell <git@cfware.com>

	* res_pjsip_phoneprov_provider: Revert 433996 / 433997.

	  res_pjsip_phoneprov_provider is using ao2_callback with OBJ_MULTIPLE, then
	  ignoring the return.  OBJ_NODATA flag was to prevent a reference leak, but
	  this caused the module to FRACK on unload.  Revert change until this can
	  be investigated further.

	  ASTERISK-24935
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4578/
	  ........

	  Merged revisions 434025 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434026 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-06 09:51 +0000 [53af579d4c]  Mark Michelson (license #5049)

	* ParkedCall: Don't allow dialplan fallthrough after retrieving parked call.

	  This is a change to align behavior with that of Asterisk 11 and previous versions.
	  In those versions, if a parked call were retrieved, and the call ended, the parked
	  call retriever would be hung up after the ParkedCall application ran. Prior to this
	  patch, in Asterisk 13, the same situation would result in the parked call retriever
	  falling through to additional priorities in the extension where the ParkedCall
	  application was called. With this patch, the behavior between Asterisk 11 and 13
	  aligns.

	  ASTERISK-24899 #close
	  Reported by Malcolm Davenport
	  Patches:
	  	ASTERISK-24899.patch uploaded by Mark Michelson(license #5049)
	  ........

	  Merged revisions 434022 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-05 07:55 +0000 [e6f0410028]  Corey Farrell <git@cfware.com>

	* res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.

	  res_pjsip_phoneprov_provider was using ao2_callback with OBJ_MULTIPLE, then
	  ignoring the return.  Added OBJ_NODATA flag to prevent a reference leak.

	  ASTERISK-24935 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4578/
	  ........

	  Merged revisions 433996 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-03 16:54 +0000 [3439487a81]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_messaging: Serialize outbound SIP MESSAGEs

	  Outbound SIP MESSAGEs had the potential to be sent out
	  of order from how they were specified in a set of
	  dialplan steps.

	  This change creates a serializer for sending outbound
	  MESSAGE requests on. This ensures that the MESSAGEs are
	  sent by Asterisk in the same order that they were sent
	  from the dialplan.

	  ASTERISK-24937 #close
	  Reported by Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/4579
	  ........

	  Merged revisions 433968 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-02 09:56 +0000 [6e5efe04bd]  Scott Griepentrog <sgriepentrog@digium.com>

	* pjsip: resolve compatibility problem with ast_sip_session

	  A change in r430179 inserted a variable near the top of a
	  structure caused a problem when running DPMA in a version
	  of Asterisk compiled across the change.  This patch moves
	  the new variable to the end of the structure, eliminating
	  the problem.

	  Review: https://reviewboard.asterisk.org/r/4574/
	  ........

	  Merged revisions 433944 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-02 05:38 +0000 [154ba47766]  Corey Farrell <git@cfware.com>

	* Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS.

	  DEBUG_CHAOS was marked as conflicting with MALLOC_DEBUG, but
	  for this to work correctly MALLOC_DEBUG must also be marked
	  as conflicting with DEBUG_CHAOS.

	  Review: https://reviewboard.asterisk.org/r/4557/
	  ........

	  Merged revisions 433923 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-01 11:30 +0000 [a217d2d1db]  Ashley Sanders <asanders@digium.com>

	* stasis: set a channel variable on websocket disconnect error

	  Resolve compile errors caused by r433863 by fixing the
	  documentation xml to comply with the schema.
	  ........

	  Merged revisions 433888 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-01 11:27 +0000 [39824e3d01]  Joshua Colp <jcolp@digium.com>

	* dns: Add support for SRV record parsing and sorting.

	  This change adds support for parsing SRV records and consuming their values
	  in an easy fashion. It also adds automatic sorting of SRV records according
	  to RFC 2782.

	  Tests have also been included which cover parsing, sorting, and off-nominal
	  cases where the record is corrupted.

	  ASTERISK-24931 #close
	  Reported by: Joshua Colp

	  Review: https://reviewboard.asterisk.org/r/4528/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-04-01 08:35 +0000 [da13d15425]  Mark Michelson <mmichelson@digium.com>

	* stasis: set a channel variable on websocket disconnect error

	  Resolve compile errors caused by r433839 by included the missing
	  header file, pbx.h.
	  ........

	  Merged revisions 433863 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-31 17:49 +0000 [06578ef407]  Ashley Sanders <asanders@digium.com>

	* stasis: set a channel variable on websocket disconnect error

	  When an error occurs while writing to a web socket, the web socket is
	  disconnected and the event is logged. A side-effect of this, however, is that
	  any application on the other side waiting for a response from Stasis is left
	  hanging indefinitely (as there is no mechanism presently available for
	  notifying interested parties about web socket error states in Stasis).

	  To remedy this scenario, this patch introduces a new channel variable:
	  STASISSTATUS.

	  The possible values for STASISSTATUS are:
	  SUCCESS         - The channel has exited Stasis without any failures
	  FAILED          - Something caused Stasis to croak. Some (not all) possible
	                    reasons for this:
	                      - The app registry is not instantiated;
	                      - The app requested is not registered;
	                      - The app requested is not active;
	                      - Stasis couldn't send a start message

	  ASTERISK-24802
	  Reported By: Kevin Harwell
	  Review: https://reviewboard.asterisk.org/r/4519/
	  ........

	  Merged revisions 433839 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-31 12:04 +0000 [2d28fa678e]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip: Fix expression in unit test /channels/chan_sip/test_sip_rtpqos.

	  Fix misplaced parentheses in original fabs() expression.
	  ........

	  Merged revisions 433816 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433817 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433818 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-31 06:55 +0000 [076fc12afb]  Corey Farrell <git@cfware.com>

	* Blocked revisions 433795

	  ........
	  Re-add _ast_mem_backtrace_buffer variable for ABI compatibility.

	  Modules built prior to commit of r4502 expect to link at runtime
	  to the variable _ast_mem_backtrace_buffer.  This change re-adds
	  the variable to the C file only.

	  Review: https://reviewboard.asterisk.org/r/4558/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433796 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-30 06:43 +0000 [8d12288d8a]  Corey Farrell <git@cfware.com>

	* Fix an ABI compatibility issue with ast_log_safe for modules.

	  Binary modules are sometimes built against the latest release of
	  Asterisk in each branch, and need to be compatible with all
	  releases of that branch.  This change ensures that utils.h only
	  uses ast_log_safe from the core.  For modules and utilities ast_log
	  is used instead.

	  Review: https://reviewboard.asterisk.org/r/4548/
	  ........

	  Merged revisions 433772 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433773 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433774 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-29 21:45 +0000 [7bc2345fb1]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix -Wabsolute-value warnings

	  This patch fixes several warnings caught by clang - in this case, usage of the
	  abs function on non-integer values. This patch uses labs and fabs, as
	  appropriate, in the various affected files.

	  Review: https://reviewboard.asterisk.org/r/4525

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4525.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433750 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-29 21:39 +0000 [ce59fabd5c]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix invalid enum conversion

	  This patch fixes some invalid enum conversion warnings caught by clang. In
	  particular:
	  * chan_sip: Several functions mixed usage of the st_refresher_param
	    enum and st_refresher enum. This patch corrects the functions to use the
	    right enum.
	  * chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state.
	  * strings: Fixed incorrect usage of AO2 flags with strings container.
	  * res_stasis: Change a return enumeration to stasis_app_user_event_res.

	  Review: https://reviewboard.asterisk.org/r/4535

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4535.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433747 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-29 21:29 +0000 [61577cbee6]  Matt Jordan <mjordan@digium.com>

	* main/stdtime/localtime: Fix warning introduced in r433720

	  The patch in r433720 caused a warning to be kicked back by gcc. It occurred
	  due to this check in unistd.h:

	      if (__nbytes > __bos0 (__buf))
	          return __read_chk_warn (__fd, __buf, __nbytes, __bos0 (__buf));

	  That is, if __nbytes is greater than the result of GCC's built-in object size
	  for the struct, we'll kick back a warning.

	  As it turns out, this is because there is an error in the code in the patch.
	  We are passing the address of the pointer to the struct, not iev, which is a
	  pointer to the struct. Hence, the number of bytes is probably going to be lot
	  larger than the number of bytes that make up a pointer! This patch changes
	  the code just read from the pointer to the struct - which fixes the warning.

	  ASTERISK-24917
	  ........

	  Merged revisions 433743 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433744 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433745 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-29 20:57 +0000 [072734692e]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Ignore -Wunused-command-line-argument

	  Asterisk's build system has a tendency to pass include directives for libraries
	  to everything compiled within a particular group of source files. This means
	  we pass the header for libxml2 to things that don't necessarily need it. As a
	  result, we ignore this particular warning.

	  Review: https://reviewboard.asterisk.org/r/4545/

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4545.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433720 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433721 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433722 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-29 20:53 +0000 [1cf949c489]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix warning for -Wgnu-variable-sized-type-not-at-end

	  This patch fixes a warning caught by clang, wherein a variable sized struct is
	  not located at the end of a struct. While the code in question actually
	  expected this, this is a good warning to watch for. Hence, this patch refactors
	  the code in question to not have two variable length elements in the same
	  struct.

	  Review: https://reviewboard.asterisk.org/r/4530/

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4530.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433717 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433718 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433719 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-28 07:56 +0000 [d2776d4d45]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix a variety of "unused" warnings

	  This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
	  errors caught by clang. Specifically:

	  * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
	                      qsmp_cmd_usage[]
	  * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
	  * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel"
	  * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
	  * funcs/func_env.c:729: Fixed ast_str_append_substr.
	  * main/editline/np/strlcat.c: removed unused rcsid variable
	  * main/editline/np/strlcpy.c: removed unused rcsid variable
	  * main/security_events.c: removed unused TIMESTAMP_STR_LEN
	  * utils/conf2ael.c: removed unused cfextension_states
	  * utils/extconf.c: removed unused cfextension_states

	  Review: https://reviewboard.asterisk.org/r/4526

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4526.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433694 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-28 07:48 +0000 [cb7b6bc4be]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix -Wself-assign

	  Assigning a variable to itself isn't super useful. However, the WAV format
	  modules make use of this in order to perform byte endian checks. This patch
	  works around the warning by only performing the self assignment if we are
	  going to do more than just assign it to ourselves. Which is odd, but true.

	  Review: https://reviewboard.asterisk.org/r/4544/

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4544.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433690 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433691 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433692 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-28 07:41 +0000 [e9520dbe0d]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix -Wparantheses-equality warnings

	  Clang will treat ((a == b)) as a warning, as it reasonably expects that the
	  developer may have intended to write (a == b) or ((a = b)). This patch cleans
	  up all instances where equality, not assignment, was intended between two
	  parantheses.

	  Review: https://reviewboard.asterisk.org/r/4531/

	  ASTERISK-24917
	  Repoted by: dkdegroot
	  patches:
	    rb4531.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433688 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-28 07:33 +0000 [fd50e5bfb5]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix -Wbitfield-constant-conversion warning

	  In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by
	  clang, as it will truncate the -1 to a 1 implicitly.

	  Instead, we just assign the value a '1'.

	  Review: https://reviewboard.asterisk.org/r/4537/

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4537.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433683 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433684 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433686 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-28 07:32 +0000 [c747b3b12a]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix -Winitializer-overrides

	  This patch fixes clange compiler warnings for initializer overrides.
	  Specifically:

	  res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration
	  value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing
	  those enum values, we therefore initialize the value twice to two different
	  values, "tlsv1" and "default". This patch changes it to just initialize
	  the index in the array to "tlsv1".

	  Review: https://reviewboard.asterisk.org/r/4539/

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4539.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433682 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433685 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-28 07:20 +0000 [d6173cd1d0]  Diederik de Groot <dkgroot@talon.nl> (License 6600)

	* clang compiler warnings: Fix -Wunused-function; make inline function static

	  This patch fixes clang compilers warnings for unused functions. Specifically:
	   * channels/chan_iax2: removed user_ref function
	   * main/dsp.c: removed goertzel_update function
	   * main/config.c: made variable_list_switch static

	  Review: https://reviewboard.asterisk.org/r/4527

	  ASTERISK-24917
	  Reported by: dkdegroot
	  patches:
	    rb4527.patch submitted by dkdegroot (License 6600)
	  ........

	  Merged revisions 433678 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433680 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433681 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 17:26 +0000 [b56592e3ae]  Jonathan Rose <jrose@digium.com>

	* SAC: Add conferencing extensions and configuration

	  Review: https://reviewboard.asterisk.org/r/4504/
	  ........

	  Merged revisions 433656 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433657 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 16:21 +0000 [c21e2e45a8]  Rusty Newton <rnewton@digium.com>

	* configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2

	  Example configuration files for a "basic PBX" deployment for the fictitious
	  Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/
	  and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company

	  Patch 4488 includes all functionality needed for SAC's outside connectivity
	  and some externally accessed features, as well as outbound dialing.

	  Reported by: Malcolm Davenport
	  Tested by: Rusty Newton

	  Review: https://reviewboard.asterisk.org/r/4488/
	  ........

	  Merged revisions 433624 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 16:06 +0000 [2659e48d9d]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage.

	  * Converted the contact_autoexpire container to use the ao2 template hash
	  and cmp functions.  Also made use the OBJ_SEARCH_xxx names instead of the
	  deprecated names.

	  * Eliminates several unnecessary uses of RAII_VAR().

	  Review: https://reviewboard.asterisk.org/r/4524/
	  ........

	  Merged revisions 433622 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 15:46 +0000 [0b62e41654]  Mark Michelson <mmichelson@digium.com>

	* Add stateful PJSIP response API call, and use it for out-of-dialog responses.

	  Asterisk had an issue where retransmissions of MESSAGE requests resulted in
	  Asterisk processing the retransmission as if it were a new MESSAGE request.

	  This patch fixes the issue by creating a transaction in PJSIP on the incoming
	  request. This way, if a retransmission arrives, the PJSIP transaction layer
	  will resend the response and Asterisk will not ever see the retransmission.

	  ASTERISK-24920 #close
	  Reported by Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/4532/
	  ........

	  Merged revisions 433619 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433620 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 15:23 +0000 [a18da4eaf2]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_registrar_expire.c: Cleanup scheduler leaks on unload/shutdown.

	  Contact expiration object refs were leaked when the module was unloaded.

	  * Made empty the scheduler of entries before destroying it to release the
	  object ref held by the scheduler entry.

	  Review: https://reviewboard.asterisk.org/r/4523/
	  ........

	  Merged revisions 433596 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 12:58 +0000 [cb1c639817]  Richard Mudgett <rmudgett@digium.com>

	* Add missing file.  ASTERISK-24781

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 09:41 +0000 [a024af1156]  Justin T. Gibbs <gibbs@scsiguy.org> (License 6692)

	* res/res_timing_kqueue: Update the module to conform to current timer API

	  This patch updates the kqueue timing module to conform to current timer API.

	  This fixes issues with using the kqueue timing source on Asterisk 13 on
	  FreeBSD 10. These issues include:

	  - Remove support for kevent64().  The values used to support Asterisk timers
	    fit within 32bits and so can be handled on all platforms via kevent().

	  - Provide debug logging for, but do not track, unacked events.  This matches
	    the behavior of all other timer implementations.

	  - Implement continuous mode by triggering and leaving active, a user event.
	    This ensures that the file descriptor for the timer returns immediately from
	    poll(), without placing the load of a high speed timer on the kernel.

	  - In kqueue_timer_get_max_rate(), don't overstate the capability of the timer.
	    On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer
	    type kqueue supports for timers.

	  - In kqueue_timer_get_event(), assume the caller woke up from poll() and just
	    return the mode the timer is currently in. This matches all other timer
	    implementations.

	  - Adjust the test code now that unacked events are not tracked.

	  Review: https://reviewboard.asterisk.org/r/4465/

	  ASTERISK-24857 #close
	  Reported by: scsiguy
	  Tested by: Ed Hynan
	  patches:
	    rb4465.patch submitted by scsiguy (License 6692)
	  ........

	  Merged revisions 433574 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433575 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 07:27 +0000 [10458d2878]  Corey Farrell <git@cfware.com>

	* Fix link error for utils/aelparse.

	  Use the standard ast_log instead of ast_log_safe for STANDALONE programs.

	  Review: https://reviewboard.asterisk.org/r/4538/
	  ........

	  Merged revisions 433549 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433550 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-27 02:12 +0000 [28e3bd0af7]  Corey Farrell <git@cfware.com>

	* Improved and portable ast_log recursion avoidance

	  This introduces a new logger routine ast_log_safe.  This routine should be
	  used for all error messages in code that can be run as a result of ast_log.
	  ast_log_safe does nothing if run recursively.  All error logging in
	  astobj2.c, strings.c and utils.h have been switched to ast_log_safe.

	  This required adding support for raw threadstorage.  This provides direct
	  access to the void* pointer in threadstorage.  In ast_log_safe, NULL is used
	  to signify that this thread is not already running ast_log_safe, (void*)1 when
	  it is already running.  This was done since it's critical that ast_log_safe
	  do nothing that could log during recursion checking.

	  ASTERISK-24155 #close
	  Reported by: Timo Teräs
	  Review: https://reviewboard.asterisk.org/r/4502/
	  ........

	  Merged revisions 433522 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433523 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433524 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-26 18:09 +0000 [554eb74516]  Corey Farrell <git@cfware.com>

	* Fix compile errors caused by r4500 / r4501.

	  * Add ast_register_cleanup to utils/clicompat.c to deal with
	    any utils that copy sources from main.
	  * Asterisk 13+: remove unused variables from core_local.c.

	  Review: https://reviewboard.asterisk.org/r/4534/
	  ........

	  Merged revisions 433499 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433500 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433501 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-26 17:24 +0000 [3ddd92902a]  Corey Farrell <git@cfware.com>

	* Replace most uses of ast_register_atexit with ast_register_cleanup.

	  Since 'core stop now' and 'core restart now' do not stop modules,
	  it is unsafe for most of the core to run cleanups.  Originally all
	  cleanups used ast_register_atexit, and were only changed when it
	  was shown to be unsafe.  ast_register_atexit is now used only when
	  absolutely required to prevent corruption and close child processes.

	  Exceptions that need to use ast_register_atexit:
	  * CDR: Flush records.
	  * res_musiconhold: Kill external applications.
	  * AstDB: Close the DB.
	  * canary_exit: Kill canary process.

	  ASTERISK-24142 #close
	  Reported by: David Brillert

	  ASTERISK-24683 #close
	  Reported by: Peter Katzmann

	  ASTERISK-24805 #close
	  Reported by: Badalian Vyacheslav

	  ASTERISK-24881 #close
	  Reported by: Corey Farrell

	  Review: https://reviewboard.asterisk.org/r/4500/
	  Review: https://reviewboard.asterisk.org/r/4501/
	  ........

	  Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-26 12:47 +0000 [d7fc85e69d]  Corey Farrell <git@cfware.com>

	* res_pjsip: Enable unload of all modules at shutdown.

	  * Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
	    caused by running PJSIP functions from non-PJSIP threads.
	  * Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
	    crashes in some cases.  In theory pj_shutdown() should take care of this.
	  * Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
	    shutdown.
	  * Resolve leaked config global in res_pjsip_notify.
	  * Unregister pubsub pjsip service module.
	  * Implement cleanup for res_pjsip_session.

	  ASTERISK-24731 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4498/
	  ........

	  Merged revisions 433469 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-26 12:13 +0000 [ab674f67b5]  Kevin Harwell <kharwell@digium.com>

	* app_confbridge: file playback blocks dtmf

	  Attempting to execute DTMF in a confbridge while file playback (prompt,
	  announcement, etc) is occurring is not allowed. You have to wait until
	  the sound file has completed before entering DTMF. This patch fixes it
	  so that app_confbridge now monitors for dtmf key presses during menu
	  driven file playback. If a key is pressed playback stops and it executes
	  the matched menu option.

	  ASTERISK-24864 #close
	  Reported by: Steve Pitts
	  Review: https://reviewboard.asterisk.org/r/4510/
	  ........

	  Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433446 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433447 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-25 13:37 +0000 [e953d15223]  Richard Mudgett <rmudgett@digium.com>

	* A couple minor cleanup tweaks.

	  * In res/res_sorcery_realtime.c: Broke long line.

	  * In main/bucket.c: Eliminated unnecessary NULL check as
	  ast_sorcery_unref() is NULL tolerant and set the global object to NULL
	  after unref in the system shutdown bucket_cleanup().
	  ........

	  Merged revisions 433420 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433421 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-25 10:31 +0000 [47156aab92]  Simon Arlott (License 5756)

	* res_xmpp: Buddies are always auto-registered when processing the roster

	  Due to a quirk in the configuration handling of res_xmpp, the 'autoregister'
	  setting was never actually processed. This was due to not properly copying
	  over the global settings to the client settings when applying the
	  configuration to the run-time object.

	  Review: https://reviewboard.asterisk.org/r/4496/

	  ASTERISK-14233
	  ASTERISK-24780 #close
	  Reported by: Simon Arlott
	  patches:
	    asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756)
	  ........

	  Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433396 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-25 07:32 +0000 [abf3e40902]  Joshua Colp <jcolp@digium.com>

	* dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests.

	  This change adds an abstracted core DNS API which resembles the API described
	  here[1]. The API provides a pluggable mechanism for resolvers and also a
	  consistent view for records. Both synchronous and asynchronous queries are
	  supported.

	  This change also adds a res_resolver_unbound module which uses the libunbound
	  library to provide resolution.

	  Unit tests have also been written for all of the above to confirm the API and
	  functionality.

	  ASTERISK-24834 #close
	  Reported by: Matt Jordan

	  ASTERISK-24836 #close
	  Reported by: Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4474/
	  Review: https://reviewboard.asterisk.org/r/4512/

	  [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-24 14:41 +0000 [4c2fc5b811]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.

	  Incoming PJSIP call legs that have not been answered yet send unnecessary
	  "180 Ringing" or "183 Progress" messages every time a connected line
	  update happens.  If the outgoing channel is also PJSIP then the incoming
	  channel will always send a "180 Ringing" or "183 Progress" message when
	  the outgoing channel sends the INVITE.

	  Consequences of these unnecessary messages:

	  * The caller can start hearing ringback before the far end even gets the
	  call.

	  * Many phones tend to grab the first connected line information and refuse
	  to update the display if it changes.  The first information is not likely
	  to be correct if the call goes to an endpoint not under the control of the
	  first Asterisk box.

	  When connected line first went into Asterisk in v1.8, chan_sip received an
	  undocumented option "rpid_immediate" that defaults to disabled.  When
	  enabled, the option immediately passes connected line update information
	  to the caller in "180 Ringing" or "183 Progress" messages as described
	  above.

	  * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
	  "183 Progress" messages.  The default is "no" to disable sending the
	  unnecessary messages.

	  ASTERISK-24781 #close
	  Reported by: Richard Mudgett

	  Review: https://reviewboard.asterisk.org/r/4473/
	  ........

	  Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-22 19:05 +0000 [60f01520e7]  snuffy <snuffy22@gmail.com> (License 5024)

	* Fix compilations errors on 64-bit OpenBSD systems

	  In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to
	  (long) when printing members of certain time structs.

	  Review: https://reviewboard.asterisk.org/r/4507

	  ASTERISK-24879 #close
	  Reported by: snuffy
	  Tested by: snuffy
	  patches:
	    openbsd-time64.diff uploaded by snuffy (License 5024)
	  ........

	  Merged revisions 433268 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433269 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433270 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-22 18:11 +0000 [66670f02e6]  snuffy <snuffy22@gmail.com> (License 5024)

	* Fix compilation issues for OpenBSD

	  This patch addresses compilation issues for OpenBSD. Specifically, it
	  addresses:
	   * It allows including <sys/vmmeter.h> in asterisk.c
	   * Provides a needed (size_t) cast in xmldoc.c

	  In 13+, it also addresses a conditional inclusion in loader.c.

	  Review: https://reviewboard.asterisk.org/r/4506

	  ASTERISK-24880 #close
	  Reported by: snuffy
	  Tested by: snuffy
	  patches:
	    misc-openbsd.diff uploaded by snuffy (License 5024)
	  ........

	  Merged revisions 433245 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433247 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-20 14:54 +0000 [7e097bce86]  Richard Mudgett <rmudgett@digium.com>

	* Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks.

	  Valgrind found some memory leaks associated with
	  ast_pjsip_rdata_get_endpoint().  The leaks would manifest when sending
	  responses to OPTIONS requests, processing MESSAGE requests, and
	  res_pjsip supplements implementing the incoming_request callback.

	  * Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in
	  res/res_pjsip.c:supplement_on_rx_request(),
	  res/res_pjsip/pjsip_options.c:send_options_response(),
	  res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and
	  res/res_pjsip_messaging.c:send_response().

	  * Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in
	  res/res_pjsip_nat.c:nat_on_rx_message().

	  * Fixed inconsistent but benign return value in
	  res/res_pjsip/pjsip_options.c:options_on_rx_request().

	  Review: https://reviewboard.asterisk.org/r/4511/
	  ........

	  Merged revisions 433222 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-20 13:27 +0000 [148e8799fe]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak respectively.

	  Valgrind found a memory leak and invalid access.

	  * Fix invalid access by sscanf() being fed a non-nul terminated string of
	  digits in res/res_pjsip_sdp_rtp.c:get_codecs().

	  * Fix memory leak in main/sorcery.c:sorcery_object_field_destructor().

	  * Fix potential NULL pointer dereference in
	  main/xmldoc.c:xmldoc_get_syntax_config_option().

	  Review: https://reviewboard.asterisk.org/r/4513/
	  ........

	  Merged revisions 433199 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433200 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-19 14:20 +0000 [627cc16a8d]  Matt Jordan <mjordan@digium.com>

	* funcs/func_env: Fix regression caused in FILE read operation

	  When r432935 was merged, it did correctly fix a situation where a FILE read
	  operation on the middle of a file buffer would not read the requested length
	  in the parameters passed to the FILE function. Unfortunately, it would also
	  allow the FILE function to append more bytes than what was available in the
	  buffer if the length exceeded the end of the buffer length.

	  This patch takes the minimum of the remaining bytes in the buffer along with
	  the calculated length to append provided by the original patch, and uses
	  that as the length to append in the return result. This patch also updates
	  the unit tests with the scenarios that were originally pointed out in
	  ASTERISK-21765 that the original implementation treated incorrectly.

	  ASTERISK-21765
	  ........

	  Merged revisions 433173 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433174 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433175 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-19 10:27 +0000 [79a81fed59]  Kevin Harwell <kharwell@digium.com>

	* alemebic scripts: endpoint identifier order option

	  The script was added in 13, but when committed to trunk it caused a branch to
	  occur due to some trunk only alemebic changes. This fixes it so that the new
	  'add_pjsip_endpoint_identifier_order script points to the correct down revision.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433152 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-19 05:21 +0000 [3aa0a869c2]  Corey Farrell <git@cfware.com>

	* logger: Apply default console logging when configuration cannot be loaded.

	  When logger.conf is missing or invalid enable console logging and display
	  an error message.

	  ASTERISK-24817 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4497/
	  ........

	  Merged revisions 433122 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433126 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433130 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-19 04:57 +0000 [d486659502]  Corey Farrell <git@cfware.com>

	* chan_sip: Simplify dialog/peer references, improve REF_DEBUG output.

	  * Replace functions for ref/undef of dialogs and peers with macro's
	    to call ao2_t_bump/ao2_t_cleanup.
	  * Enable passthough of REF_DEBUG caller information to sip_alloc and
	    find_call.

	  ASTERISK-24882 #close 
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4189/
	  ........

	  Merged revisions 433115 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-19 04:46 +0000 [2c83ac4364]  Corey Farrell <git@cfware.com>

	* chan_sip: Fix dialog reference leaked to scheduler for reinvite_timeout.

	  Release the scheduler reference to the dialog for reinvite timeout during
	  dialog_unlink_all.

	  ASTERISK-24876 #close 
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4491/
	  ........

	  Merged revisions 433112 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433113 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433114 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-17 21:42 +0000 [e0ea490a11]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session: Fix off-nominal extra unref of session.
	  ........

	  Merged revisions 433088 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-17 17:15 +0000 [8c65c9167e]  Scott Griepentrog <sgriepentrog@digium.com>

	* Various: bugfixes found via chaos

	  Using DEBUG_CHAOS several instances of a null
	  pointer crash, and one uninitialized variable
	  were uncovered and fixed.  Also added details
	  on why Asterisk failed to initialize.

	  Review: https://reviewboard.asterisk.org/r/4468/
	  ........

	  Merged revisions 433064 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-17 17:03 +0000 [f25b265329]  Scott Griepentrog <sgriepentrog@digium.com>

	* core: Introduce chaos into memory allocations

	  Locate potential crashes by exercising seldom
	  used code paths.  This patch introduces a new
	  define DEBUG_CHAOS, and mechanism to randomly
	  return an error condition from functions that
	  will seldom do so.  Functions that handle the
	  allocation of memory get the first treatment.

	  Review: https://reviewboard.asterisk.org/r/4463/
	  ........

	  Merged revisions 433060 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-17 17:03 +0000 [62cf2a2c02]  Scott Griepentrog <sgriepentrog@digium.com>

	* Reverting accidental ci of wrong change in r433061


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433062 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-17 17:00 +0000 [cb6c7eecfd]  Scott Griepentrog <sgriepentrog@digium.com>

	* various: cleanup issues found during leak hunt

	  In this collection of small patches to prevent
	  Valgrind errors are: fixes for reference leaks
	  in config hooks, evaluating a parameter beyond
	  bounds, and accessing a structure after a lock
	  where it could have been already free'd.

	  Review: https://reviewboard.asterisk.org/r/4407/
	  ........

	  Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433061 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-17 16:52 +0000 [c41dd32b94]  Richard Mudgett <rmudgett@digium.com>

	* Audit ast_sockaddr_resolve() usage for memory leaks.

	  Valgrind found some memory leaks associated with ast_sockaddr_resolve().
	  Most of the leaks had already been fixed by earlier memory leak hunt
	  patches.  This patch performs an audit of ast_sockaddr_resolve() and found
	  one more.

	  * Fix ast_sockaddr_resolve() memory leak in
	  apps/app_externalivr.c:app_exec().

	  * Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs
	  parameter for safety so the pointer will never be uninitialized on return.
	  The same goes for res/res_pjsip_acl.c:extract_contact_addr().

	  * Made functions that call ast_sockaddr_resolve() with RAII_VAR()
	  controlling the addrs variable use ast_free instead of ast_free_ptr to
	  provide better MALLOC_DEBUG information.

	  Review: https://reviewboard.asterisk.org/r/4509/
	  ........

	  Merged revisions 433056 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 433057 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-17 13:35 +0000 [803a916334]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: Allow configuration of endpoint identifier query order

	  Updated some documentation stating that endpoint identifiers registered without
	  a name are place at the front of the lookup list. Also renamed register method
	  'ast_sip_register_endpoint_identifier_by_name' to
	  'ast_sip_register_endpoint_identifier_with_name'

	  ASTERISK-24840
	  Reported by: Mark Michelson
	  ........

	  Merged revisions 433031 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-17 13:22 +0000 [aef7278af6]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: Allow configuration of endpoint identifier query order

	  This patch fixes previously reverted code that caused binary incompatibility
	  problems with some modules. And like the original patch it makes sure that
	  no matter what order the endpoint identifier modules were loaded, priority is
	  given based on the ones specified in the new global 'endpoint_identifier_order'
	  option.

	  ASTERISK-24840
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4489/
	  ........

	  Merged revisions 433028 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-17 11:11 +0000 [259e833e88]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Add reason comment.
	  ........

	  Merged revisions 433005 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-13 21:29 +0000 [e89f83b3ad]  Matt Jordan <mjordan@digium.com>

	* main/frame: Don't report empty disallow values as an error

	  In realtime, it is normal to have a database with both 'allow' and 'disallow'
	  columns in the schema. It is perfectly valid to have an 'allow' value of
	  '!all,g722,ulaw,alaw' and no 'disallow' value. Unlike in static conf files,
	  you can't *not* provide the disallow value. Thus, the empty disallow value
	  causes a spurious WARNING message, which is kind of annoying.

	  This patch makes it so that a 'disallow' value with no ... value ... is
	  ignored. Granted, you can still screw this up as well, as technically
	  specifying 'disallow=all,!ulaw' allows only ulaw, and then you would have no
	  'allow' value in your database. But really, why would you do that? WHY?

	  ASTERISK-16779 #close
	  Reported by: Atis Lezdins
	  ........

	  Merged revisions 432970 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-03-13 21:01 +0000 [0d52907d2b]  Joshua Colp <jcolp@digium.com>

	* func_curl: Don't hold exclusive lock when performing HTTP request.

	  This code originally kept a lock held when performing the HTTP
	  request to ensure that the options provided to curl remain valid.
	  This doesn't seem to be necessary these days and holding the lock
	  caused requests to happen sequentially instead of in parallel.

	  ASTERISK-18708 #close
	  Reported by: Dave Cabot
	  ........

	  Merged revisions 432948 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-03-13 20:53 +0000 [ac1214d9d4]  Jan Juergens (License 6538)

	* apps/app_sms: Add an option to prevent SMS content from being logged

	  In some countries, privacy laws specify that SMS content cannot be saved by a
	  provider. This patch adds a new option to the SMS application, 'n', which
	  prevents the SMS content from being written to the SMS log.

	  ASTERISK-22591 #close
	  Reported by: Jan Juergens
	  patches:
	    DisableSmsContentLoggingByParam.patch uploaded by Jan Juergens (License 6538)


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-13 20:37 +0000 [b3fa35786f]  Joshua Colp <jcolp@digium.com>

	* core: Fix tab completion of "core set debug channel" CLI command.

	  The "core set debug channel" CLI command mistakenly had source filenames
	  added to its tab completion. This occurred because the CLI generator fell back
	  to the "core set debug" command which permits setting debug at a source
	  filename level.

	  ASTERISK-21038 #close
	  Reported by: Richard Kenner
	  ........

	  Merged revisions 432944 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-03-13 20:22 +0000 [b4cc056067]  Di-Shi Sun (License 5076)

	* FILE: fix retrieval of file contents when offset is specified

	  The loop that reads in a file was not correctly using the offset when
	  determining what bytes to append to the output. This patch corrects
	  the logic such that the correct portion of the file is extracted when an
	  offset is specified.

	  ASTERISK-21765
	  Reported by: John Zhong
	  Tested by: Matt Jordan, Di-Shi Sun
	  patches:
	    file_read_390821.patch uploaded by Di-Shi Sun (License 5076)
	  ........

	  Merged revisions 432935 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432938 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-13 19:24 +0000 [dc752f515b]  Matt Jordan <mjordan@digium.com>

	* apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentation

	  This patch corrects the documentation for the AMD application. Specifically:
	  * It documents the maximum_word_length option, which limits the maximum allowed
	    length of a single utterance.
	  * It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH
	    was documented as MAXWORDS, while MAXWORDS was undocumented.

	  Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues.

	  ASTERISK-19470 #close
	  Reported by: Frank DiGennaro
	  ........

	  Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-03-13 12:06 +0000 [c52adca396]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip: AMI action PJSIPShowEndpoint closes AMI connection on error.

	  Also fixed similar problem with AMI action PJSIPShowEndpoints.

	  ASTERISK-24872 #close
	  Reported by: Dmitriy Serov

	  Review: https://reviewboard.asterisk.org/r/4487/
	  ........

	  Merged revisions 432894 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-13 11:37 +0000 [636d82f4d8]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent.

	  The res_pjsip modules were manually checking both name and number
	  presentation values when there is a function that determines the combined
	  presentation for a party ID struct.  The function takes into account if
	  the name or number components are valid while the manual code rarely
	  checked if the data was even valid.

	  * Made use ast_party_id_presentation() rather than manually checking party
	  ID presentation values.

	  * Ensure that set_id_from_pai() and set_id_from_rpid() will not return
	  presentation values other than what is pulled out of the SIP headers.  It
	  is best if the code doesn't assume that AST_PRES_ALLOWED and
	  AST_PRES_USER_NUMBER_UNSCREENED are zero.

	  * Fixed copy paste error in add_privacy_params() dealing with RPID
	  privacy.

	  * Pulled the id->number.valid test from add_privacy_header() and
	  add_privacy_params() up into the parent function add_id_headers() to skip
	  adding PAI/RPID headers earlier.

	  * Made update_connected_line_information() not send out connected line
	  updates if the connected line number is invalid.  Lower level code would
	  not add the party ID information and thus the sent message would be
	  unnecessary.

	  * Eliminated RAII_VAR usage in send_direct_media_request().

	  Review: https://reviewboard.asterisk.org/r/4472/
	  ........

	  Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-13 09:55 +0000 [d42c6adb1a]  Kevin Harwell <kharwell@digium.com>

	* Revert - res_pjsip: Allow configuration of endpoint identifier query order

	  Due to a break in binary compatibility with some other modules these changes
	  are being reverted until the issue can be resolved.

	  ASTERISK-24840
	  Reported by: Mark Michelson
	  ........

	  Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-12 21:10 +0000 [f2c21ead1f]  Corey Farrell <git@cfware.com>

	* Logger: Fix MALLOC_DEBUG build error.

	  Revision 432834 introduced a build error when MALLOC_DEBUG
	  is used.  Switch callid threadstorage to simple
	  AST_THREADSTORAGE since we no longer need custom cleanup.

	  Reported by: Corey Farrell


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432851 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-12 20:12 +0000 [c08fd275bf]  Corey Farrell <git@cfware.com>

	* Logger: Convert 'struct ast_callid' to unsigned int.

	  Switch logger callid's from AO2 objects to simple integers.
	  This helps in two ways.  Copying integers is faster than
	  referencing AO2 objects, so this will result in a small
	  reduction in logger overhead.  This also erases the possibility
	  of an infinate loop caused by an invalid callid in
	  threadstorage.

	  ASTERISK-24833 #comment Committed callid conversion to trunk. 
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4466/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-12 07:58 +0000 [38ee441ea7]  Matt Jordan <mjordan@digium.com>

	* main/audiohook: Update internal sample rate on reads

	  When an audiohook is created (which is used by the various Spy applications
	  and Snoop channel in Asterisk 13+), it initially is given a sample rate of
	  8kHz. It is expected, however, that this rate may change based on the media
	  that passes through the audiohook. However, the read/write operations on the
	  audiohook behave very differently.

	  When a frame is written to the audiohook, the format of the frame is checked
	  against the internal sample rate. If the rate of the format does not match
	  the internal sample rate, the internal sample rate is updated and a new SLIN
	  format is chosen based on that sample rate. This works just fine.

	  When a frame is read, however, we do something quite different. If the format
	  rate matches the internal sample rate, all is fine. However, if the rates
	  don't match, the audiohook attempts to "fix up" the number of samples that
	  were requested. This can result in some seriously large number of samples
	  being requested from the read/write factories.

	  Consider the worst case - 192kHz SLIN. If we attempt to read 20ms worth of
	  audio produced at that rate, we'd request 3840 samples (192000 / (1000 / 20)).
	  However, if the audiohook is still expecting an internal sample rate of 8000,
	  we'll attempt to "fix up" the requested samples to:

	    samples_converted = samples * (ast_format_get_sample_rate(format) /
	                                   (float) audiohook->hook_internal_samp_rate);

	    which is:

	    92160 = 3840 * (192000 / 8000)

	  This results in us attempting to read 92160 samples from our factories, as
	  opposed to the 3840 that we actually wanted. On a 64-bit machine, this
	  miraculously survives - despite allocating up to two buffers of length 92160
	  on the stack. The 32-bit machines aren't quite so lucky. Even in the case where
	  this works, we will either (a) get way more samples than we wanted; or (b) get
	  about 3840 samples, assuming the timing is pretty good on the machine.

	  Either way, the calculation being performed is wrong, based on the API users
	  expectations.

	  My first inclination was to allocate the buffers on the heap. As it is,
	  however, there's at least two drawbacks with doing this:
	  (1) It's a bit complicated, as the size of the buffers may change during the
	      lifetime of the audiohook (ew).
	  (2) The stack is faster (yay); the heap is slower (boo).

	  Since our calculation is flat out wrong in the first place, this patch fixes
	  this issue by instead updating the internal sample rate based on the format
	  passed into the read operation. This causes us to read the correct number of
	  samples, and has the added benefit of setting the audihook with the right
	  SLIN format.

	  Note that this issue was caught by the Asterisk Test Suite as a result of
	  r432195 in the 13 branch. Because this issue is also theoretically possible
	  in Asterisk 11, the change is being made here as well.

	  Review: https://reviewboard.asterisk.org/r/4475/
	  ........

	  Merged revisions 432810 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432811 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432812 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-12 07:40 +0000 [29304d10a0]  Diederik de Groot (License 6600)

	* Add support for the clang compiler; update RAII_VAR to use BlocksRuntime

	  RAII_VAR, which is used extensively in Asterisk to manage reference counted
	  resources, uses a GCC extension to automatically invoke a cleanup function
	  when a variable loses scope. While this functionality is incredibly useful
	  and has prevented a large number of memory leaks, it also prevents Asterisk
	  from being compiled with clang.

	  This patch updates the RAII_VAR macro such that it can be compiled with clang.
	  It makes use of the BlocksRuntime, which allows for a closure to be created
	  that performs the actual cleanup.

	  Note that this does not attempt to address the numerous warnings that the clang
	  compiler catches in Asterisk.

	  Much thanks for this patch goes to:
	  * The folks on StackOverflow who asked this question and Leushenko for
	    providing the answer that formed the basis of this code:
	    http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang
	  * Diederik de Groot, who has been extremely patient in working on getting this
	    patch into Asterisk.

	  Review: https://reviewboard.asterisk.org/r/4370/

	  ASTERISK-24133
	  ASTERISK-23666
	  ASTERISK-20399
	  ASTERISK-20850 #close
	  Reported by: Diederik de Groot
	  patches:
	    RAII_CLANG.patch uploaded by Diederik de Groot (License 6600)
	  ........

	  Merged revisions 432807 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-03-11 11:39 +0000 [4115e327ac]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Move internal init/destroy prototypes to private header file.

	  Done as a separate commit from a finding in
	  https://reviewboard.asterisk.org/r/4467/
	  ........

	  Merged revisions 432787 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-11 10:26 +0000 [89b65f5dda]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Fix pjsip.conf type=global object default value handling.

	  When a type=global section is not defined in pjsip.conf the global
	  defaults are not applied.  As a result the mandatory Max-Forwards header
	  is not added to SIP messages for res_pjsip/chan_pjsip.

	  The handling of pjsip.conf type=global objects has several problems:

	  1) If the global object is missing the defaults are not applied.

	  2) If the global object is missing the default_outbound_endpoint's default
	  value is not returned by ast_sip_global_default_outbound_endpoint().

	  3) Defines are needed so default values only need to be changed in one
	  place.

	  * Added a sorcery instance observer callback to check if there were any
	  type=global sections loaded.  If there were more than one then issue an
	  error message.  If there were none then apply the global defaults.

	  * Fixed ast_sip_global_default_outbound_endpoint() to return the
	  documented default when no type=global object is defined.

	  * Made defines for the global default values.

	  * Increased the default_useragent[] size because SVN version strings can
	  get lengthy and 128 characters may not be enough.

	  * Fixed an off-nominal code path ref leak in global_alloc() if the string
	  fields fail to initialize.

	  * Eliminated RAII_VAR in get_global_cfg() and
	  ast_sip_global_default_outbound_endpoint().

	  ASTERISK-24807 #close
	  Reported by: Anatoli

	  Review: https://reviewboard.asterisk.org/r/4467/
	  ........

	  Merged revisions 432766 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432767 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-11 10:22 +0000 [185d2e082a]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Fixed invalid empty Server and User-Agent SIP headers.

	  Setting pjsip.conf useragent to an empty string results in an empty SIP
	  header being sent.

	  * Made not add an empty SIP header item to the global SIP headers list.

	  Review: https://reviewboard.asterisk.org/r/4467/
	  ........

	  Merged revisions 432764 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-10 18:09 +0000 [2889f074a0]  Joshua Colp <jcolp@digium.com>

	* core: Don't create snapshots with locks.

	  Snapshots are immutable and are never changed. Allocating them
	  with a lock is wasteful.

	  Review: https://reviewboard.asterisk.org/r/4469/
	  ........

	  Merged revisions 432742 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432743 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-10 16:33 +0000 [15d266bf85]  Javier Acosta (License 6690)

	* res/res_config_odbc: Fix improper escaping of backslashes with MySQL

	  When escaping backslashes with MySQL, the proper way to escape the characters
	  in a LIKE clause is to escape the '\' four times, i.e., '\\\\'. To quote the
	  MySQL manual:

	  "Because MySQL uses C escape syntax in strings (for example, “\n” to represent
	  a newline character), you must double any “\” that you use in LIKE strings.
	  For example, to search for “\n”, specify it as “\\n”. To search for “\”,
	  specify it as “\\\\”; this is because the backslashes are stripped once by the
	  parser and again when the pattern match is made, leaving a single backslash to
	  be matched against."

	  ASTERISK-24808 #close
	  Reported by: Javier Acosta
	  patches:
	    res_config_odbc.diff uploaded by Javier Acosta (License 6690)
	  ........

	  Merged revisions 432720 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-03-10 13:13 +0000 [ab6e2c93f3]  Graham Barnett (License 6685)

	* app_voicemail: Fix crash with IMAP backends when greetings aren't present

	  When an IMAP backend is in use and greetings are set to be used, but aren't
	  present for a user in their IMAP folder, Asterisk will crash. This occurs
	  due to the mailstream being set to the 'greetings' folder and being left
	  in that particular state, regardless of the success/failure of the attempt
	  to access the folder the mailstream points to. Later access of the mailstream
	  assumes that it points to the 'INBOX' (or some other folder), resulting in
	  either a crash (if the greetings folder didn't exist and the mailstream is
	  invalid) or an inability to read messages from the 'INBOX' folder.

	  This patch restores the mailstream to its correct state after accessing the
	  greetings. This fixes the crash, and sets the mailstream to the state that
	  VoiceMailMain expects.

	  Note that while ASTERISK-23390 also contained a patch for this issue, the
	  patch on ASTERISK-24786 is the one being merged here.

	  Review: https://reviewboard.asterisk.org/r/4459/

	  ASTERISK-23390 #close
	  Reported by: Ben Smithurst

	  ASTERISK-24786 #close
	  Reported by: Graham Barnett
	  Tested by: Graham Barnett
	  patches:
	    app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett (License 6685)
	  ........

	  Merged revisions 432695 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432696 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432697 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-10 13:05 +0000 [79e9b37ad0]  Ed Hynan (Licnese 6680)

	* localtime: Fix file descriptor leak on kqueue(2) systems

	  The localtime management in the Asterisk core contains a thread that watches
	  for changes in the local timezone. On systems where the directory containing
	  /etc/localtime is modified frequently, the thread monitoring the changes will
	  be woken up to determine if any changes in timezone have occurred. When using
	  kqueue(2), this can cause a leak of file descriptors due to some improper
	  management of resources.

	  This patch updates the kqueue(2) handling in localtime, such that is no longer
	  leaks resources.

	  Review: https://reviewboard.asterisk.org/r/4450/

	  ASTERISK-24739 #close
	  Reported by: Ed Hynan
	  patches:
	    11.15.0-u.diff uploaded by Ed Hynan (Licnese 6680)
	    11.7.0-u.diff uploaded by Ed Hynan (License 6680)
	    svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License 6680)
	  ........

	  Merged revisions 432691 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432693 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-10 11:08 +0000 [e7ee83ea90]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER.

	  A race condition happened between initiating a transfer and requesting
	  that a dialog termination be delayed.  Occasionally, the transferrer
	  channels would exit the bridge and hangup before the dialog termination
	  delay was requested.

	  * Made request dialog termination delay before initiating the transfer
	  action.  If the transfer fails then cancel the delayed dialog termination
	  request.

	  ASTERISK-24755 #close
	  Reported by: John Bigelow

	  Review: https://reviewboard.asterisk.org/r/4460/
	  ........

	  Merged revisions 432668 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432669 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-09 11:13 +0000 [1ce529d30e]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: allow configuration of endpoint identifier query order

	  It's possible to have a scenario that will create a conflict between endpoint
	  identifiers. For instance an incoming call could be identified by two different
	  endpoint identifiers and the one chosen depended upon which identifier module
	  loaded first. This of course causes problems when, for example, the incoming
	  call is expected to be identified by username, but instead is identified by ip.
	  This patch adds a new 'global' option to res_pjsip called
	  'endpoint_identifier_order'. It is a comma separated list of endpoint
	  identifier names that specifies the order by which identifiers are processed
	  and checked.

	  ASTERISK-24840 #close
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4455/
	  ........

	  Merged revisions 432638 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-07 19:47 +0000 [a5f80f1781]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Fix wrongful use of USE_PJPROJECT define.

	  As pjproject is now used as a shared library a different define,
	  HAVE_PJPROJECT, is used to specify if pjproject is present.

	  ASTERISK-24830 #close
	  Reported by: Stefan Engström
	  ........

	  Merged revisions 432614 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-06 16:59 +0000 [affcf1d766]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_refer: Make safely get the context for a blind transfer.

	  Made safely get the TRANSFER_CONTEXT channel value while the channel is
	  locked in refer_incoming_attended_request() and
	  refer_incoming_blind_request().  The pointer returned by
	  pbx_builtin_getvar_helper() is only valid while the channel is locked.
	  ........

	  Merged revisions 432594 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-06 16:18 +0000 [090ab1735b]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_refer: Made refer_attended_alloc() not create the ao2 object with a lock.

	  The lock is unused.
	  ........

	  Merged revisions 432574 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-03-06 15:38 +0000 [b85cb7ea1b]  Jonathan Rose <jrose@digium.com>

	* app: Add functions to swap voicemail function table for testing purposes
	  ........

	  Merged revisions 432556 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432573 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-06 14:24 +0000 [c7cc1b3059]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi/sig_analog: Fix distinctive ring detection to suck less.

	  The distinctive ring feature interferes with detecting Caller ID and
	  appears to have been broken for years.  What happens is if you have a
	  ring-ring cadence as used in the UK you get too many DAHDI events for the
	  distinctive ring pattern array and Caller ID detection is aborted.  I
	  think when Zapata/DAHDI added the ring begin event it broke distinctive
	  ring.  More events happen than before and the code does no filtering of
	  which event times are recorded in the pattern array.

	  * Made distinctive ring only record the ringt count when the ring ends
	  instead of on just any DAHDI event.  Distinctive ring can be ring,
	  ring-ring, ring-ring-ring, or different ring durations for the up to three
	  rings.

	  * Fixed the distinctive ring detection enable (chan_dahdi.conf option
	  usedistinctiveringdetection) to be per port instead of somewhat per port
	  and somewhat global.  This has been broken since v1.8.

	  * Fixed using the default distinctive ring context when the detected
	  pattern does not match any configured dringX patterns.  The default
	  context did not get set when the previous call was a matched distinctive
	  ring pattern and the current call is not matched.  This has been broken
	  since v1.8.

	  * Made distinctive ring have no effect on Caller ID detection when it is
	  disabled.  Caller ID detection just monitors for 10 seconds before giving
	  up.

	  * Fixed leak of struct callerid_state memory when a polarity reversal
	  during Caller ID detection causes the incoming call to be aborted.

	  DAHDI-1143
	  AST-1545
	  ASTERISK-24825 #close
	  Reported by: Richard Mudgett

	  ASTERISK-17588
	  Reported by: Daniel Flounders

	  Review: https://reviewboard.asterisk.org/r/4444/
	  ........

	  Merged revisions 432530 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432534 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-06 13:34 +0000 [f1ab2c5e8b]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip: Fix realtime locking inversion when poking a just built peer.

	  When a realtime peer is built it can cause a locking inversion when the
	  just built peer is poked.  If the CLI command "sip show channels" is
	  periodically executed then a deadlock can happen because of the locking
	  inversion.

	  * Push the peer poke off onto the scheduler thread to avoid the locking
	  inversion of the just built realtime peer.

	  AST-1540
	  ASTERISK-24838 #close
	  Reported by: Richard Mudgett

	  Review: https://reviewboard.asterisk.org/r/4454/
	  ........

	  Merged revisions 432526 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432528 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432529 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-05 10:40 +0000 [5c3e33b3ca]  gtjoseph <george.joseph@fairview5.com>

	* app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE.

	  There is a leftover "assert" in app_voicemail/__messagecount that references 
	  variables that don't exist.  This causes the compile to fail when 
	  --enable-dev-mode and IMAP_STORAGE are selected.

	  This patch removes the assert.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4461/
	  ........

	  Merged revisions 432484 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432486 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-04 12:55 +0000 [41ba8fd7c0]  Matt Jordan <mjordan@digium.com>

	* translate: Prevent invalid memory accesses on fast shutdown

	  When a 'core restart now' or 'core stop now' is executed and a channel is
	  currently in a media operation, the translator matrix can be destroyed while a
	  channel is currently blocked on getting the best translation choice
	  (see ast_translator_best_choice). When the channel gets the mutex, the
	  translation matrix now has invalid memory, and Asterisk crashes.

	  This patch does two things:
	  (1) We now only clean up the translation matrix on a graceful shutdown. In that
	      case, there are no channels, and so there is no risk of this occurring.
	  (2) We also now set the __matrix and __indextable to NULL. In some initial
	      backtraces when this occurred, it looked as if there was a memory corruption
	      occurring, and it wasn't until we determined that something had restarted
	      Asterisk that the issue became clear. By setting these to NULL on shutdown,
	      it becomes a bit easier to determine why a crash is occurring.

	  Note that we could litter the code with NULL checks on the __matrix, but the
	  act of making the translation matrix cleaned up on shutdown should preclude
	  this issue from occurring in the first place, and this part of the code needs
	  to be as fast as possible.

	  Review: https://reviewboard.asterisk.org/r/4457/
	  ........

	  Merged revisions 432453 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432455 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-03-02 13:15 +0000 [278ea2f468]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_sdp_rtp: Revert portion of r432195

	  Unfortunately, while initial testing with ConfBridge did not reproduce the
	  audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing
	  did show that bridge_softmix and/or ConfBridge has a severe problem bridging
	  two or more participants at different sampling rates. Sometimes, it even picks
	  odd sampling rates that cause hideous audio problems.

	  This patch backs out the offending portion of the code until the issues in
	  the affected bridging modules can be more properly analyzed.

	  ASTERISK-24841
	  ........

	  Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432425 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-27 12:31 +0000 [9e841e4fb6]  Richard Mudgett <rmudgett@digium.com>

	* ARI: Fix crash if integer values used in JSON payload 'variables' object.

	  Sending the following ARI commands caused Asterisk to crash if the JSON
	  body 'variables' object passes values of types other than strings.

	  POST /ari/channels
	  POST /ari/channels/{channelid}
	  PUT /ari/endpoints/sendMessage
	  PUT /ari/endpoints/{tech}/{resource}/sendMessage

	  * Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),
	  ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and
	  ast_ari_endpoints_send_message_to_endpoint().

	  ASTERISK-24751 #close
	  Reported by:  jeffrey putnam

	  Review: https://reviewboard.asterisk.org/r/4447/
	  ........

	  Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432405 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-26 12:53 +0000 [d79670b269]  Scott Griepentrog <sgriepentrog@digium.com>

	* Dial API: add self destruct option when complete

	  This patch adds a self-destruction option to the
	  dial api.  The usefulness of this is mostly when
	  using async mode to spawn a separate thread used
	  to handle the new call, while the calling thread
	  is allowed to go on about other business.

	  The only alternative to this option would be the
	  calling thread spawning a new thread, or hanging
	  around itself waiting to destroy the dial struct
	  after completion.

	  Example of use (minus error checking):

	    struct ast_dial *dial = ast_dial_create();

	    ast_dial_append(dial, "PJSIP", "200", NULL);

	    ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, "Echo");
	    ast_dial_option_global_enable(dial, AST_DIAL_OPTION_SELF_DESTROY, NULL);

	    ast_dial_run(dial, NULL, 1);

	  The dial_run call will return almost immediately
	  after spawning the new thread to run and monitor
	  the dial.  If the call is answered, it is placed
	  into the echo app.  When completed, it will call
	  ast_dial_destroy() on the dial structure.

	  Note that any allocations made to pass values to
	  ast_dial_set_user_data() or dial options must be
	  free'd in a state callback function on any of:
	    AST_DIAL_RESULT_UNASWERED,
	    AST_DIAL_RESULT_ANSWERED,
	    AST_DIAL_RESULT_HANGUP, or 
	    AST_DIAL_RESULT_TIMEOUT.

	  Review: https://reviewboard.asterisk.org/r/4443/
	  ........

	  Merged revisions 432385 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432386 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-26 11:12 +0000 [d04fbb0f9d]  Kevin Harwell <kharwell@digium.com>

	* app_chanspy, channel: fix frame leaks

	  Fixed a couple of frame leaks that were found during testing.

	  ASTERISK-24828 #close
	  Reported by: John Hardin
	  Review: https://reviewboard.asterisk.org/r/4445/
	  ........

	  Merged revisions 432362 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432363 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432364 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-25 22:58 +0000 [8a16c2f0c2]  Matt Jordan <mjordan@digium.com>

	* make: Remove 'res_features' from libraries to link against with cygwin/mingw32

	  Both the apps and channels Makefiles still listed 'res_features' as modules to
	  link against when compiling for cygwin or mingw32. This module hasn't existed
	  for quite some time.

	  ASTERISK-18105 #close
	  Reported by: feyfre
	  ........

	  Merged revisions 432341 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-25 21:03 +0000 [3725173b9e]  Makoto Dei (License 5027)

	* channels/chan_sip: Don't send a BYE after final response when PBX thread fails

	  When Asterisk fails to start a PBX thread for a new channel - for example, when
	  the maxcalls setting in asterisk.conf is exceeded - we currently send a final
	  response, and then attempt to send a BYE request to the UA. Since that's all
	  sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt
	  such that we don't get stuck sending BYE requests to something that does not
	  want it.

	  Note that this patch is a slight modification of the one on ASTERISK-15434.
	  For clarity, it explicitly calls sipalreadygone with the calls to transmit a
	  final response.

	  ASTERISK-21845
	  ASTERISK-15434 #close
	  Reported by: Makoto Dei
	  Tested by: Matt Jordan
	  patches:
	    sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027)
	  ........

	  Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-25 17:49 +0000 [e484140aed]  Rusty Newton <rnewton@digium.com>

	* configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 1

	  Example configuration files for a "basic PBX" deployment for the fictitious
	  Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4379/
	  and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company

	  Reported by: Malcolm Davenport
	  Tested by: Rusty Newton

	  Review: https://reviewboard.asterisk.org/r/4379/
	  ........

	  Merged revisions 432301 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432302 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-25 17:09 +0000 [ced84d7e62]  Matt Jordan <mjordan@digium.com>

	* configure: Promote SQLite3 "not installed" warning to error

	  Since Asterisk won't build without the library, not having it is definitely
	  an error. Thanks to Kyle Kurz for pointing this out.
	  ........

	  Merged revisions 432280 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432281 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-25 17:05 +0000 [4b63da7f7d]  Matt Jordan <mjordan@digium.com>

	* channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario

	  When we receive an SDP as part of an offer/answer for a peer/friend has been
	  configured to require encryption, and that SDP offer/answer failed to provide
	  acceptable crypto attributes, we currently issue a WARNING that uses the phrase
	  "we" and "requested". In this case, both of those terms are ambiguous - the
	  user will probably think "we" is Asterisk (it most likely isn't) and it may
	  not be a "request", so much as an SDP that was received in some fashion.

	  This patch makes the WARNING messages slightly less bad and a bit more
	  accurate as well.

	  ASTERISK-23214 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 432277 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-25 15:42 +0000 [d68012d1a3]  Olle Johansson <oej@edvina.net> (License 5267)

	* channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI

	  Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would
	  be rejected if those crypto attributes contained either a key lifetime or a
	  MKI parameter. While from a theoretical point of view this was defensible -
	  Asterisk does not support key lifetimes or multiple crypto keys - from a
	  practical point of view, this is quite a problem. A large number of endpoints
	  offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually
	  have to support anything more than a single key or refresh the key.
	  In reality, this is (so far as we've seen) always the case.

	  This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8
	  branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters
	  in the following fashion:

	  > The Lingon branch now handle lifetime and MKI parameters.
	  >
	  > We only accept lifetimes up to max for the crypto and higher than 10 hours
	  > for packetization of 20 ms (50 pps).
	  >
	  > We only handle MKI with index 1.
	  >
	  > We do not really bother with counting packets and reinviting at end of
	  > lifetime, so the min of 10 hours kind of takes care of most calls. If there
	  > are longer ones, we rely on the other side for re-invites.
	  >
	  > It's still not perfect, but I personally think this is an improvement. A
	  > configuration option for minimum lifetime accepted could be added.

	  When the patch was ported forward, I decided against adding a configuration
	  option as Olle's handling was more than sufficient for every case I've seen
	  come through the issue tracker or through interoperability testing. We can
	  revisit that decision if it proves to be false.

	  A few small other tweaks were made to the surrounding code to reduce
	  indentation and provide better type safety for the 'tag' parameter.

	  Review: https://reviewboard.asterisk.org/r/4419/
	  Review: https://reviewboard.asterisk.org/r/4418/

	  ASTERISK-17721 #close
	  Reported by: Terry Wilson

	  ASTERISK-17899 #close
	  Reported by: Dwayne Hubbard
	  patches:
	    lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267)

	  ASTERISK-20233
	  Reported by: tootai

	  ASTERISK-22748
	  Reported by: Alejandro Mejia
	  ........

	  Merged revisions 432239 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-25 14:47 +0000 [ff642289f4]  David M. Lee <dlee@digium.com>

	* Increase WebSocket frame size and improve large read handling

	  Some WebSocket applications, like [chan_respoke][], require a larger
	  frame size than the default 8k; this patch bumps the default to 16k.
	  This patch also fixes some problems exacerbated by large frames.

	  The sanity counter was decremented on every fread attempt in
	  ws_safe_read(), regardless of whether data was read from the socket or
	  not. For large frames, this could result in loss of sanity prior to
	  reading the entire frame. (16k frame / 1448 bytes per segment = 12
	  segments).

	  This patch changes the sanity counter so that it only decrements when
	  fread() doesn't read any bytes. This more closely matches the original
	  intention of ws_safe_read(), given that the error message is
	  "Websocket seems unresponsive".

	  This patch also properly logs EOF conditions, so disconnects are no
	  longer confused with unresponsive connections.

	   [chan_respoke]: https://github.com/respoke/chan_respoke

	  Review: https://reviewboard.asterisk.org/r/4431/
	  ........

	  Merged revisions 432236 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432237 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-24 17:00 +0000 [57525c3cf2]  Richard Mudgett <rmudgett@digium.com>

	* config.h: Use real parameter names for ast_variable_new() define.

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432220 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-24 16:14 +0000 [8574c4d197]  Matt Jordan <mjordan@digium.com>

	* channels/chan_sip: Fix crash when transmitting packet after thread shutdown

	  When the monitor thread is stopped, its pthread ID is set to a specific value
	  (AST_PTHREADT_STOP) so that later portions of the code can determine whether
	  or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit
	  failed to check for that value, checking instead only for AST_PTHREAD_STOP.
	  Passing the invalid yet very specific value to pthread_kill causes a crash.

	  This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that
	  it doesn't attempt to poke the thread if the thread has already been stopped.

	  ASTERISK-24800 #close
	  Reported by: JoshE
	  ........

	  Merged revisions 432198 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432199 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-24 16:00 +0000 [a528dfc9a7]  Matt Jordan <mjordan@digium.com>

	* ARI/PJSIP: Apply requesting channel's format cap to created channels

	  This patch addresses the following problems:
	  * ari/resource_channels: In ARI, we currently create a format capability
	    structure of SLIN and apply it to the new channel being created. This was
	    originally done when the PBX core was used to create the channel, as there
	    was a condition where a newly created channel could be created without any
	    formats. Unfortunately, now that the Dial API is being used, this has two
	    drawbacks:
	    (a) SLIN, while it will ensure audio will flows, can cause a lot of
	        needless transcodings to occur, particularly when a Local channel is
	        created to the dialplan. When no format capabilities are available, the
	        Dial API handles this better by handing all audio formats to the requsted
	        channels. As such, we defer to that API to provide the format
	        capabilities.
	    (b) If a channel (requester) is causing this channel to be created, we
	        currently don't use its format capabilities as we are passing in our own.
	        However, the Dial API will use the requester channel's formats if none
	        are passed into it, and the requester channel exists and has format
	        capabilities. This is the "best" scenario, as it is the most likely to
	        create a media path that minimizes transcoding.
	    Fixing this simply entails removing the providing of the format capabilities
	    structure to the Dial API.

	  * chan_pjsip: Rather than blindly picking the first format in the format
	    capability structure - which actually *can* be a video or text format - we
	    select an audio format, and only pick the first format if that fails. That
	    minimizes the weird scenario where we attempt to transcode between video/audio.

	  * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure.
	    Since ast_request already limits us down to one format capability once the
	    format capabilities are passed along, there's no reason to squelch it here.

	  * channel: Fixed a comment. The reason we have to minimize our requested
	    format capabilities down to a single format is due to Asterisk's inability
	    to convey the format to be used back "up" a channel chain. Consider the
	    following:

	      PJSIP/A => L;1 <=> L;2 => PJSIP/B
	      g,u,a     g,u,a    g,u,a      u

	    That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials
	    PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local
	    channel has inherited those format capabilities down the line; PJSIP/B
	    supports only ulaw. According to these format capabilities, ulaw is
	    acceptable and should be selected across all the channels, and no
	    transcoding should occur. However, there is no way to convey this: when L;2
	    and PJSIP/B are put into a bridge, we will select ulaw, but that is not
	    conveyed to PJSIP/A and L;1. Thus, we end up with:

	      PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
	        g          g   X   u        u

	    Which causes g722 to be written to PJSIP/B.

	    Even if we can convey the 'ulaw' choice back up the chain (which through
	    some severe hacking in Local channels was accomplished), such that the chain
	    looks like:

	      PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
	        u          u       u         u

	    We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back
	    with only 'ulaw'. This results in all the channel structures being set up
	    correctly, but PJSIP/A *still* sending g722 and causing the chain to fall
	    apart.

	    There's a lot of difficulty just in setting this up, as there are numerous
	    race conditions in the act of bridging, and no clean mechanism to pass the
	    selected format backwards down an established channel chain. As such, the
	    best that can be done at this point in time is clarifying the comment.

	  Review: https://reviewboard.asterisk.org/r/4434/

	  ASTERISK-24812 #close
	  Reported by: Matt Jordan
	  ........

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2015-02-24 12:38 +0000 [91733b5d15]  Kevin Harwell <kharwell@digium.com>

	* bridge_softmix: G.729 codec license held

	  When more than one call using the same codec type enters into a softmix bridge
	  and no audio is present for a channel the bridge optimizes the out frame by
	  using the same one for all channels with the same codec type. Unfortunately,
	  when that number (channels with same codec type) dropped to <= 1 the codec
	  was not dereferenced. At least not until all parties left the bridge. Thus in
	  the case of G.729 the license was not released. This patch ensures that the
	  codec is dereferenced immediately when the optimization no longer applies.

	  ASTERISK-24797 #close
	  Reported by: Luke Hulsey
	  Review: https://reviewboard.asterisk.org/r/4429/
	  ........

	  Merged revisions 432174 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-21 14:48 +0000 [bedf51b2ce]  Joshua Colp <jcolp@digium.com>

	* res_ari_channels: Return a 404 response when a requested channel variable does not exist.

	  This change makes it so that if a channel variable is requested and it does not exist
	  a 404 response will be returned instead of an allocation failed response. This makes
	  it easier to debug and figure out what is going on for a user.

	  ASTERISK-24677 #close
	  Reported by: Joshua Colp
	  ........

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2015-02-21 13:28 +0000 [87b7060f36]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_registrar: Add Expires header to 200 OK if present in REGISTER.

	  Some implementations don't pay attention to the expires for individual contacts.
	  In this case they may consider the lack of an Expires header in the 200 OK as
	  unregistered. This change makes it so if an Expires header is present in the REGISTER
	  we will add one in the 200 OK.

	  ASTERISK-24785 #close
	  Reported by: Ross Beer
	  ........

	  Merged revisions 432136 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-21 12:53 +0000 [283bb15c16]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid.

	  ASTERISK-24499 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 432118 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-21 11:36 +0000 [b3c1ad5d73]  Graham Barnett (License 6685)

	* apps/app_voicemail: Demote an ERROR message to a WARNING message

	  When using IMAP voicemail with FreePBX, you will often get ERROR messages
	  complaining about not being able to find a mailbox. This is due to how FreePBX
	  handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this
	  a configuration error, as in any other system it would be indicative of
	  someone misconfiguring their system.

	  Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch
	  demotes the message so that system administrators can hopefully reduce some
	  of the noise in their log files.

	  Note that in the original patch this was made into a NOTICE, but that's a
	  too forgiving.

	  ASTERISK-24790 #close
	  Reported by: Graham Barnett
	  patches:
	    app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685)
	  ........

	  Merged revisions 432098 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-21 08:06 +0000 [2ea7ccbf70]  Joshua Colp <jcolp@digium.com>

	* http: Add missing html tag to 'httpstatus' functionality.

	  ASTERISK-24724 #close
	  Reported by: Ashley Sanders
	  ........

	  Merged revisions 432078 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-20 20:58 +0000 [e66b874f5d]  Corey Farrell <git@cfware.com>

	* Allow shutdown to unload modules that register bucket scheme's or codec's.

	  * Change __ast_module_shutdown_ref to be NULL safe (11+).
	  * Allow modules that call ast_bucket_scheme_register or ast_codec_register
	    to be unloaded during graceful shutdown only (13+ only).

	  ASTERISK-24796 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4428/
	  ........

	  Merged revisions 432058 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 432059 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-20 20:51 +0000 [bb71672a47]  Corey Farrell <git@cfware.com>

	* main/asterisk.c: Reverse #if statement in listener() to fix code folding.

	  listener() opens the same code block in two places (#if and #else).  This
	  confuses some folding editors causing it to think that an extra code block
	  was opened.  Folding in 'geany' causes all code after listener() to be
	  folded as if it were part of that procedure.

	  ASTERISK-24813 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4435/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432057 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-20 20:47 +0000 [ce50fa314a]  Corey Farrell <git@cfware.com>

	* asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64-bit integers.

	  Add a couple of missing closing brackets / parenthesis.

	  ASTERISK-24814 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4436/
	  ........

	  Merged revisions 432054 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-20 11:55 +0000 [bb06603d5f]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi/sig_analog: Put log message strings on one line.

	  With the log messages on one line, you can search for the log message seen
	  in the log and expect to find it.
	  ........

	  Merged revisions 432032 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-20 11:53 +0000 [340818ad12]  Matt Hoskins (license 6688)

	* ASTERISK-24811: Add ast_sorcery_apply_config() to res_pjsip_publish_asterisk.

	  Matt Hoskins reported that res_pjsip_publish_asterisk wouldn't pull config from 
	  realtime.  Turns out it was just missing a call ast_sorcery_apply_config().

	  res_pjsip_acl was missing it as well, so I added it.  The other pjsip modules 
	  looked OK.

	  ASTERISK-24811 #close
	  Reported-by: Matt Hoskins
	  Tested-by: George Joseph
	  Tested-by: Matt Hoskins
	  patches:
	  	res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins (license 6688)

	  Review: https://reviewboard.asterisk.org/r/4433/
	  ........

	  Merged revisions 432033 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-20 09:47 +0000 [4dab71831f]  Graham Barnett (License 6685)

	* apps/app_voicemail: Fix IMAP header compatibility issue with Microsoft Exchange

	  When interfacing with Microsoft Exchange, custom headers will be returned as
	  all lower case. Currently, the IMAP header code will fail to parse the returned
	  custom headers, as it will be performing a case sensitive comparison. This can
	  cause playback of messages to fail, as needed information - such as origtime -
	  will not be present.

	  This patch updates app_voicemail's header parsing code to perform a case
	  insensitive lookup for the requested custom headers. Since the headers are
	  specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be
	  unique in an IMAP message, this should cause no issues with other systems.

	  ASTERISK-24787 #close
	  Reported by: Graham Barnett
	  patches:
	    app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685)
	  ........

	  Merged revisions 432012 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-19 15:26 +0000 [05cc6d6d55]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi: Remove some dead code.
	  ........

	  Merged revisions 431992 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-19 12:26 +0000 [252aee4228]  Richard Mudgett <rmudgett@digium.com>

	* ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association.

	  Processing an AOC-E event that does not or no longer has a channel
	  association causes a crash.

	  The problem with posting AOC events to the channel topic is that AOC-E
	  events don't always have a channel association and posting the event to
	  the all channels topic is just wrong.  AOC-E events do however have their
	  own charging association method to refer to the agreement with the
	  charging entity.

	  * Changed the AOC events to post to the AMI manager topic instead of the
	  channel topics.  If a channel is associated with the event then channel
	  snapshot information is supplied with the AMI event.

	  * Eliminated RAII_VAR() usage in aoc_to_ami() and ast_aoc_manager_event().

	  This patch supercedes the patch on Review: https://reviewboard.asterisk.org/r/4427/

	  ASTERISK-22670 #close
	  Reported by: klaus3000

	  ASTERISK-24689 #close
	  Reported by: Marcel Manz

	  ASTERISK-24740 #close
	  Reported by: Panos Gkikakis

	  Review: https://reviewboard.asterisk.org/r/4430/
	  ........

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2015-02-19 11:37 +0000 [6992b2e8fa]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_refer: Handle INVITE with Replaces failure after answer.

	  * Fixed hangup handling of the session->channel after answer if the
	  ast_channel_move() or ast_bridge_impart() fails.  We are still the thread
	  controlling the session->channel so we need to call ast_hangup() to kill
	  the channel.

	  * Fixed debug messages in refer_incoming_invite_request() referencing
	  incorrect channnels on success.  Code comments now say why the
	  session->channel cannot be used.

	  Review: https://reviewboard.asterisk.org/r/4422/
	  ........

	  Merged revisions 431956 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-19 09:28 +0000 [e3fd826cdb]  Alexander Traud (License 6520)

	* tcptls: Handle new OpenSSL compile time option to disable SSLv3

	  Some distributions are going to disable SSLv3 at compile time. This option can
	  be checked using the directive OPENSSL_NO_SSL3_METHOD. This patch updates the
	  TCP/TLS handling in Asterisk to look for that directive before attempting to
	  use the SSLv3 specific methods.

	  ASTERISK-24799 #close
	  Reported by: Alexander Traud
	  patches:
	    no-ssl3-method.patch uploaded by Alexander Traud (License 6520)
	  ........

	  Merged revisions 431936 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-18 20:03 +0000 [a4774ceaa5]  Corey Farrell <git@cfware.com>

	* Create work around for scheduler leaks during shutdown.

	  * Added ast_sched_clean_by_callback for cleanup of scheduled events
	    that have not yet fired.
	  * Run all pending peercnt_remove_cb and replace_callno events in chan_iax2.
	    Cleanup of replace_callno events is only run 11, since it no longer
	    releases any references or allocations in 13+.

	  ASTERISK-24451 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4425/
	  ........

	  Merged revisions 431916 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-17 09:34 +0000 [09bfe4b208]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_refer: Fix crash from a REFER and BYE collision.

	  Analyzing a one-off crash on a busy system showed that processing a REFER
	  request had a NULL session channel pointer.  The only way I can think of
	  that could cause this is if an outgoing BYE transaction overlapped the
	  incoming REFER transaction in a collision.  Asterisk sends a BYE while the
	  phone sends a REFER to complete an attended transfer.

	  * Made check the session channel pointer before processing an incoming
	  REFER request in res_pjsip_refer.

	  * Fixed similar crash potential for res_pjsip supplement incoming request
	  processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE,
	  res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER
	  messages.

	  * Made res_pjsip_messaging respond to a message body too large with a 413
	  instead of ignoring it.

	  ASTERISK-24700 #close
	  Reported by: Zane Conkle

	  Review: https://reviewboard.asterisk.org/r/4417/
	  ........

	  Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-16 15:29 +0000 [d808eace5c]  Matt Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk: Fix crash in debug from RTCP reports without report block

	  When RTCP debugging was enabled, an RTCP report without a report block would
	  cause a crash. This was due to the verbose output not checking to see if the
	  report_block pointer was NULl before dereferencing it.

	  This patch adds the necessary check to prevent printing any verbose output
	  if the far side hasn't provided us the information they should have.

	  ASTERISK-24791 #close
	  Reported by: JoshE
	  Tested by: JoshE
	  ........

	  Merged revisions 431879 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-15 13:01 +0000 [55eb8fc068]  Joshua Colp <jcolp@digium.com>

	* pjsip: Remove "contact" type from pjsip.conf.sample

	  The "contact" object is not meant to be configured from the pjsip.conf
	  configuration file. It is meant to be created as a result of a registration
	  and stored elsewhere.

	  ASTERISK-24085 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 431860 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-15 12:00 +0000 [55709bc1f7]  Joshua Colp <jcolp@digium.com>

	* install_prereq: Tweak flags when configuring pjproject.

	  This change does two things:
	  1. Disables debugging so assertions which can return an error do,
	  instead of asserting.
	  2. Enables IPv6 support.

	  ASTERISK-24632 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 431843 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-15 11:43 +0000 [e78dd39885]  Joshua Colp <jcolp@digium.com>

	* res_sorcery_config: Improve object lookup times.

	  The res_sorcery_config module currently uses a fixed bucket
	  size of 53. This means that depending on the number of objects
	  you either end up with excess buckets or a lot of collisions.
	  Due to the way that res_sorcery_config is implemented it's actually
	  possible to make the bucket size dynamic based on the number of
	  objects. This is due to the fact that each loading of the config file
	  produces a new container and does not modify the existing one.
	  This change uses the number of expected objects and finds a prime
	  number near it. In practice depending on the number of objects this
	  can speed up lookups anywhere from 2X to 15X. This change also removes
	  the lock from the container as it is not needed.

	  Review: https://reviewboard.asterisk.org/r/4423/
	  ........

	  Merged revisions 431841 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-15 10:01 +0000 [e6fe69b76c]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Add "pjsip show version" CLI command.

	  When debugging things it can be useful to know absolutely what
	  version of pjproject res_pjsip is running against. This change
	  adds a "pjsip show version" CLI command which can be used to
	  query for this.

	  ASTERISK-24685 #close
	  Reported by: Joshua Colp

	  Review: https://reviewboard.asterisk.org/r/4424/
	  ........

	  Merged revisions 431824 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431825 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-15 06:41 +0000 [17f9e0cacc]  Matthias Urlichs (license 5508)

	* res_timing_pthread: Fix leaky pipes.

	  During some refactoring the way private information for timers
	  was stored was changed. As a result of this the action which normally
	  removed the timer upon closure in res_timing_pthread was also removed
	  causing the timer to remain after it should using up resources.
	  This change ensures that the timer is removed upon closure.

	  ASTERISK-24768 #close
	  Reported by: Matthias Urlichs
	  patches:
	   timer.patch submitted by Matthias Urlichs (license 5508)
	  ........

	  Merged revisions 431807 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-14 18:33 +0000 [d1bd8b091b]  Matt Jordan <mjordan@digium.com>

	* apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes

	  The Test Event for MIXMONITOR_END - which signals that a MixMonitor has
	  completed - technically fired before the filestream was closed. If a test
	  used this to trigger a condition to verify that the file was written, it
	  could result in a race condition where the file size would not be what the
	  test expected.

	  Luckily, no tests were using this (although they should have been). Since the
	  test event needed to be moved after the point where the MixMonitor autochan has
	  been destroyed, the test event no longer emits the channel name. Luckily,
	  nothing needs it.
	  ........

	  Merged revisions 431788 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 431789 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-14 13:46 +0000 [455a98a2f8]  Joshua Colp <jcolp@digium.com>

	* sorcery: Output an error message if a wizard is specified for an object type and it isn't found.

	  ASTERISK-24612 #close
	  Reported by: Joshua Colp
	  ........

	  Merged revisions 431771 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-14 12:31 +0000 [fae6bf8ace]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_exten_state: Improve log message when a subscription is attempted to a non-existent extension.

	  ASTERISK-24716 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 431754 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-14 12:21 +0000 [cc96e4a7ef]  Joshua Colp <jcolp@digium.com>

	* Multiple revisions 431751-431752

	  ........
	    r431751 | file | 2015-02-14 14:19:07 -0400 (Sat, 14 Feb 2015) | 5 lines
	    
	    chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type.
	    
	    ASTERISK-24771 #close
	    Reported by: Niklas Larsson
	  ........
	    r431752 | file | 2015-02-14 14:20:27 -0400 (Sat, 14 Feb 2015) | 2 lines
	    
	    'information' ends with an 'n'.
	  ........

	  Merged revisions 431751-431752 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-13 11:24 +0000 [f00ebf0a2d]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_session: Fix double re-INVITE collision crash.

	  A multi-asterisk box setup with direct media enabled would occasionally
	  crash when two re-INVITE collisions on a call leg happen in a row.

	  The re-INVITE logic only had one timer struct to defer the re-INVITE.
	  When the second collision happens the timer struct is overwritten and put
	  into the timer heap again.  Resources for the first timer are leaked and
	  the heap has two positions occupied by the same timer struct.  Now the
	  heap ordering is potentially corrupted, the timer will fire twice, and any
	  resources allocated for the second timer will be released twice.

	  * The solution is to put the collided re-INVITE into the delayed requests
	  queue with all the other delayed requests and cherry pick the next request
	  that can come off the queue when an event happens.

	  * Changed to put delayed BYE requests at the head of the delayed queue.
	  There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE
	  has been requested.

	  * Made the start of a BYE request flush the delayed requests queue to
	  prevent a delayed request from overlapping the BYE transaction.  I saw a
	  few cases where a delayed re-INVITE got started after the BYE transaction
	  started.

	  * Changed the delayed_request struct to use an enum instead of a string
	  for the request method.  Cherry picking the queue is easier with an enum
	  than string comparisons and the compiler can warn if a switch statement
	  does not cover all defined enum values.

	  * Improved the debug output to give more information.  It helps to know
	  which channel is involved with an endpoint.  Trunks can have many channels
	  associated with the endpoint at the same time.

	  ASTERISK-24727 #close
	  Reported by: Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/4414/
	  ........

	  Merged revisions 431734 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-12 14:34 +0000 [29f66b0429]  Matt Jordan <mjordan@digium.com>

	* ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app

	  This patch adds a new feature to ARI to redirect a channel to another server,
	  and fixes a few bugs in PJSIP's handling of the Transfer dialplan
	  application/ARI redirect capability.

	  *New Feature*
	  A new operation has been added to the ARI channels resource, redirect. With
	  this, a channel in a Stasis application can be redirected to another endpoint
	  of the same underlying channel technology.

	  *Bug fixes*
	  In the process of writing this new feature, two bugs were fixed in the PJSIP
	  stack:
	  (1) The existing .transfer channel callback had the limitation that it could
	      only transfer channels to a SIP URI, i.e., you had to pass
	      'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
	      still supported, it is somewhat unintuitive - particularly in a world full
	      of endpoints. As such, we now also support specifying the PJSIP endpoint to
	      transfer to.
	  (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
	      updating its Contact header. Alas, that resulted in the forwarding
	      destination set by the dialplan application/ARI resource/whatever being
	      rewritten with very incorrect information. Hence, we now don't bother
	      updating an outgoing response if it is a 302. Since this took a looong time
	      to find, some additional debug statements have been added to those modules
	      that update the Contact headers.

	  Review: https://reviewboard.asterisk.org/r/4316/

	  ASTERISK-24015 #close
	  Reported by: Private Name

	  ASTERISK-24703 #close
	  Reported by: Matt Jordan
	  ........

	  Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-11 12:03 +0000 [9d081ed06c]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: dtls_handler causes Asterisk to crash

	  There have been a couple of times where a crash occurred in the dtls_handler
	  section of the code for res_pjsip. Unfortunately, in working this issue the
	  problem was unable to be reproduced. After looking at the backtraces and
	  through the code the current best guess as to why this happened might be due
	  to a reentrance problem and the strtok function. So, the current fix is to
	  convert the strtok function into the reentrant version of the function,
	  strtok_r.

	  ASTERISK-24741 #close
	  Reported by: Zane Conkle
	  Review: https://reviewboard.asterisk.org/r/4409/
	  ........

	  Merged revisions 431698 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-11 11:45 +0000 [cc85e55d88]  Kevin Harwell <kharwell@digium.com>

	* ari_websockets: removed extra check on websocket session read

	  When merging the websocket timeout issue (ASTERISK-24701) an extra, almost
	  duplicate, check was left in the code that should not have been. This removes
	  it.

	  ASTERISK-24701 #close
	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/4412/
	  ........

	  Merged revisions 431693 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431695 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-11 11:39 +0000 [e2d3215b83]  Richard Mudgett <rmudgett@digium.com>

	* HTTP: Stop accepting requests on final system shutdown.

	  There are three CLI commands to stop and restart Asterisk each.

	  1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
	  New channels are prevented while the shutdown request is pending.

	  2) core stop/restart gracefully - Stop or restart Asterisk when there are
	  no calls remaining in the system.  New channels are prevented while the
	  shutdown request is pending.

	  3) core stop/restart when convenient - Stop or restart Asterisk when there
	  are no calls in the system.  New calls are not prevented while the
	  shutdown request is pending.

	  ARI has made stopping/restarting Asterisk more problematic.  While a
	  shutdown request is pending it is desirable to continue to process ARI
	  HTTP requests for current calls.  To handle the current calls while a
	  shutdown request is pending, a new committed to shutdown phase is needed
	  so ARI applications can deal with the calls until the system is fully
	  committed to shutdown.

	  * Added a new shutdown committed phase so ARI applications can deal with
	  calls until the final committed to shutdown phase is reached.

	  * Made refuse new HTTP requests when the system has reached the final
	  system shutdown phase.  Starting anything while the system is actively
	  releasing resources and unloading modules is not a good thing.

	  * Split the bridging framework shutdown to not cleanup the global bridging
	  containers when shutting down in a hurry.  This is similar to how other
	  modules prevent crashes on rapid system shutdown.

	  * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
	  ast_shutting_down().  You should not have to include channel.h just to
	  access these system functions.

	  ASTERISK-24752 #close
	  Reported by: Matthew Jordan

	  Review: https://reviewboard.asterisk.org/r/4399/
	  ........

	  Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-11 11:13 +0000 [5a17ed7a38]  Richard Miller (License 5685)

	* channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB

	  When a SIP device that has its registration stored in RealTime unregisters,
	  the entry for that device is updated with blank values, i.e., "", indicating
	  that it is no longer registered. Unfortunately, one of those values that is
	  'blanked' is the device's port. If the column type for the port is not a
	  string datatype (the recommended type is integer), an ODBC or database error
	  will be thrown. MariaDB does not coerce empty strings to a valid integer value.

	  This patch updates the query run from chan_sip such that it replaces the port
	  value with a value of '0', as opposed to a blank value. This is the value that
	  other database backends coerce the empty string ("") to already, and the
	  handling of reading a RealTime registration value from a backend already
	  anticipates receiving a port of '0' from the backends.

	  ASTERISK-24772 #close
	  Reported by: Richard Miller
	  patches:
	    chan_sip.diff uploaded by Richard Miller (License 5685)
	  ........

	  Merged revisions 431673 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 431674 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-11 11:03 +0000 [8cc50b1ebc]  Corey Farrell <git@cfware.com>

	* Enable REF_DEBUG for ast_module_ref / ast_module_unref.

	  Add ast_module_shutdown_ref for use by modules that can
	  only be unloaded during graceful shutdown.

	  When REF_DEBUG is enabled:
	  * Add an empty ao2 object to struct ast_module.
	  * Allocate ao2 object when the module is loaded.
	  * Perform an ao2_ref in each place where mod->usecount is manipulated.
	  * ao2_cleanup on module unload.

	  ASTERISK-24479 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4141/
	  ........

	  Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 431663 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-02-11 10:52 +0000 [137c4b0778]  Kevin Harwell <kharwell@digium.com>

	* res_http_websocket: websocket write timeout fails to fully disconnect

	  When writing to a websocket if a timeout occurred the underlying socket did not
	  get closed/disconnected. This patch makes sure the websocket gets disconnected
	  on a write timeout. Also a notice is logged stating that the websocket was
	  disconnected.

	  ASTERISK-24701 #close
	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/4412/
	  ........

	  Merged revisions 431669 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 431670 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-10 17:17 +0000 [49161d8df8]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard: Add ability to auto-create hints.

	  Looking at the Super Awesome Company sample reminded me that creating hints is 
	  just plain gruntwork.  So you can now have the pjsip conifg wizard auto-create 
	  them for you.

	  Specifying 'hint_exten' in the wizard will create 
	  'exten => <hint_exten>,hint/PJSIP/<wizard_id>'
	  in whatever is specified for 'hint_context'.

	  Specifying 'hint_application' in the wizard will create
	  'exten => <hint_exten>,1,<hint_application>'
	  in whatever is specified for 'hint_context'.

	  The default for 'hint_context' is the endpoint's context.
	  There's no default for 'hint_application'.  If not specified, no app is added.
	  There's no default for 'hint_exten'.  If not specified, neither the hint itself 
	  nor the application will be created.

	  Some may think this is the slippery slope to users.conf but hints are a basic 
	  necessity for phones unlike voicemail, manager, etc that users.conf creates.

	  Tested-by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/4383/
	  ........

	  Merged revisions 431643 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431644 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-08 21:12 +0000 [858e825568]  Ben Merrills (License 6678)

	* res/ari/resource_channels: Add missing 'no_answer' reason to DELETE /channels

	  One of the canonical reasons for hanging up a channel is because the far end
	  failed to answer - or because someone else answered, and we want to get rid of
	  this channel. This patch adds the missing value to the 'reason' query parameter
	  for the DELETE /channels operation.

	  Review: https://reviewboard.asterisk.org/r/4400

	  ASTERISK-24745 #close
	  Reported by: Ben Merrills
	  patches:
	    add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678)
	  ........

	  Merged revisions 431622 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-08 20:35 +0000 [17247daae6]  ibercom <ibercom123@gmail.com> (License 6599)

	* res/res_odbc: Remove unneeded queries when determining if a table exists

	  This patch modifies the ast_odbc_find_table function such that it only performs
	  a lookup of the requested table if the table is not already known. Prior to
	  this patch, a queries would be executed against the database even if the table
	  was already known and cached.

	  Review: https://reviewboard.asterisk.org/r/4405/

	  ASTERISK-24742 #close
	  Reported by: ibercom
	  patches:
	    patch.diff uploaded by ibercom (License 6599)
	  ........

	  Merged revisions 431617 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-02-08 11:24 +0000 [2ebe811d80]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_sdp_rtp: Fix leak of local ICE candidates when applying to SDP

	  When an SDP is created for an outgoing request/response, the ICE candidates
	  obtained from the RTP instance are currently leaked. This causes the ao2
	  container that holds the candidates to never properly be reclaimed when the
	  RTP instance is destroyed.

	  This patch properly decrements the ICE candidates' container if it is
	  successfully obtained.

	  ASTERISK-24769 #close
	  Reported by: Matt Jordan
	  ........

	  Merged revisions 431600 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431601 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-06 15:26 +0000 [7ca1a0da04]  Scott Griepentrog <sgriepentrog@digium.com>

	* various: cleanup issues found during leak hunt

	  In this collection of small patches to prevent
	  Valgrind errors are: fixes for reference leaks
	  in config hooks, evaluating a parameter beyond
	  bounds, and accessing a structure after a lock
	  where it could have been already free'd.

	  Review: https://reviewboard.asterisk.org/r/4407/
	  ........

	  Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-03 19:27 +0000 [a79c920aa1]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_keepalive: Don't crash if PJSIP module is not loaded.
	  ........

	  Merged revisions 431555 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-02-03 18:59 +0000 [03ce56d6c5]  Joshua Colp <jcolp@digium.com>

	* sorcery: Don't try to load object types which haven't been defined.

	  The act of defining wizards for an object type in sorcery.conf will
	  create a minimal object type. This can cause a problem when a module
	  has multiple sorcery instances (which all get the wizards from sorcery.conf
	  applied) but the sorcery instances do not all contain full information
	  about the object types. Upon loading errors will occur stating that
	  the objects can not be created. This is confusing and is actually
	  perfectly fine.

	  This change makes it so that only object types which have been fully
	  defined will be loaded.

	  ASTERISK-24748 #close
	  Reported by: Joshua Colp
	  ........

	  Merged revisions 431538 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-31 10:28 +0000 [14a57782a6]  Joshua Colp <jcolp@digium.com>

	* res_format_attr_h264: Fix crash when determining joint capability.

	  The res_format_attr_h264 module currently incorrectly attempts to
	  copy SPS and PPS information from the wrong attribute. This change
	  fixes that.

	  ASTERISK-24616 #close
	  Reported by: Yura Kocyuba

	  Review: https://reviewboard.asterisk.org/r/4392/
	  ........

	  Merged revisions 431521 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-30 11:49 +0000 [23bb5f6a73]  Richard Mudgett <rmudgett@digium.com>

	* app_agent_pool: Fix initial module load agent device state reporting.

	  When the app_agent_pool module initially loads there is a race condition
	  between the thread loading agents.conf and the device state internal
	  processing thread.  If the device state internal processing thread handles
	  the agent creation state updates before the thread that loaded agents.conf
	  registers the device state provider callback then the cached agent state
	  is "Invalid".  When a consumer module like app_queue asks for the agent state
	  it gets the cached "Invalid" state instead of the real state from the provider.

	  * Moved loading the agents.conf configuration to the last thing setup by
	  app_agent_pool in load_module().  Now the device state provider callback
	  is registered before the config is loaded so the agent creation state
	  updates are guaranteed to get the initial device state.

	  * Removed some now redundant config cleanup on error in load_config().

	  * Added lock protection when accessing the device state in
	  agent_pvt_devstate_get() and eliminated the RAII_VAR() usage.

	  ASTERISK-24737 #close
	  Reported by: Steve Pitts

	  Review: https://reviewboard.asterisk.org/r/4390/
	  ........

	  Merged revisions 431492 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431493 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-30 11:41 +0000 [5c9f1b3f51]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publish: eventually crashes when no response is ever received

	  When Asterisk attempts to send SIP outbound publish information and no response
	  is ever received (no 200 okay, 412, 423) the system eventually crashes. A
	  response is never received because the system Asterisk is attempting to send
	  publish information to is not available. The underlying pjsip framework attempts
	  to send publish information. After several attempts it calls back into the
	  Asterisk outbound publish code. At this point if the "client->queue" is empty
	  Asterisk attempts to schedule a refresh which utilizes "rdata" and since no
	  response was received the given "rdata" struture is NULL. Attempting to
	  dereference a NULL object of course results in a crash.

	  The fix here removes the dependency on rdata for schedule_publish_refresh.
	  Instead param->expiration is now passed to it as this is set to -1 if no
	  response is received. Also added a notification when no response is received.

	  ASTERISK-24635 #close
	  Reported by: Marco Paland
	  Review: https://reviewboard.asterisk.org/r/4384/
	  ........

	  Merged revisions 431490 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431491 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-30 11:21 +0000 [6a76740b83]  Ashley Sanders <asanders@digium.com>

	* HTTP: For httpd server, need option to define server name for security purposes

	  Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. In this version, [servername] is uncommented by default.

	  ASTERISK-24316 #close
	  Reported By: Andrew Nagy
	  Review: https://reviewboard.asterisk.org/r/4374/
	  ........

	  Merged revisions 431471 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-30 10:49 +0000 [bd0bdf1e41]  Mark Michelson <mmichelson@digium.com>

	* Fix some memory leaks.

	  These memory leaks were found and fixed by John Hardin. I'm just
	  committing them for him.

	  ASTERISK-24736 #close
	  Reported by Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/4389
	  ........

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2015-01-29 17:03 +0000 [388d691f34]  Scott Griepentrog <sgriepentrog@digium.com>

	* stasis transfer: fix stasis bridge push race part two

	  When swapping a Local channel in place of one already
	  in a bridge (to complete a bridge attended transfer),
	  the channel that was swapped out can actually be hung
	  up before the stasis bridge push callback executes on
	  the independant transfer thread.  This results in the
	  stasis app loop dropping out and removing the control
	  that has the the app name which the local replacement
	  channel needs so it can re-enter stasis.

	  To avoid this race condition a new push_peek callback
	  has been added, and called from the ast_bridge_impart
	  thread before it launches the independant thread that
	  will complete the transfer.  Now the stasis push_peek
	  callback can copy the stasis app name before the swap
	  channel can hang up.

	  ASTERISK-24649
	  Review: https://reviewboard.asterisk.org/r/4382/
	  ........

	  Merged revisions 431450 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-29 15:20 +0000 [f61c80a8f7]  Mark Michelson <mmichelson@digium.com>

	* Allow disabling of 100rel support on PJSIP endpoints.

	  Due to an inversion error, setting 100rel=no would not actually
	  change the current value of the setting (which defaulted to "yes").
	  With this fix, the inversion is corrected.
	  ........

	  Merged revisions 431420 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-29 15:02 +0000 [034798e37e]  Mark Michelson <mmichelson@digium.com>

	* Use SIPS URIs in Contact headers when appropriate.

	  RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
	  scenarios when we are required to use SIPS URIs in Contact
	  headers. Asterisk's non-compliance with this could actually
	  cause calls to get dropped when communicating with clients
	  that are strict about checking the Contact header.

	  Both of the SIP stacks in Asterisk suffered from this issue.
	  This changeset corrects the behavior in res_pjsip/chan_pjsip.c

	  Review: https://reviewboard.asterisk.org/r/4345
	  ........

	  Merged revisions 431426 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-29 14:54 +0000 [fe76d4829f]  Mark Michelson <mmichelson@digium.com>

	* Use SIPS URIs in Contact headers when appropriate.

	  RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
	  scenarios when we are required to use SIPS URIs in Contact
	  headers. Asterisk's non-compliance with this could actually
	  cause calls to get dropped when communicating with clients
	  that are strict about checking the Contact header.

	  Both of the SIP stacks in Asterisk suffered from this issue.
	  This changeset corrects the behavior in chan_sip.

	  ASTERISK-24646 #close
	  Reported by Stephan Eisvogel

	  Review: https://reviewboard.asterisk.org/r/4346
	  ........

	  Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-01-29 10:47 +0000 [8357ffab9c]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_exten_state: Reduce log clutter... change a WARNING to a VERBOSE/2

	  Reduce log clutter by changing the "Watcher for hint %s (removed|deactivated)"
	  message from WARNING to VERBOSE/2.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4387/
	  ........

	  Merged revisions 431403 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-29 06:09 +0000 [9893ba7ffb]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Fix DTLS when used with OpenSSL 1.0.1k

	  A recent security fix for OpenSSL broke DTLS negotiation for many
	  applications. This was caused by read ahead not being enabled when it
	  should be. While a commit has gone into OpenSSL to force read ahead
	  on for DTLS it may take some time for a release to be made and the
	  change to be present in distributions (if at all). As enabling read
	  ahead is a simple one line change this commit does that and fixes
	  the issue.

	  ASTERISK-24711 #close
	  Reported by: Jared Biel
	  ........

	  Merged revisions 431384 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-01-28 11:42 +0000 [b3ff43a4e8]  Mark Michelson <mmichelson@digium.com>

	* Fix file descriptor leak in RTP code.

	  SIP requests that offered codecs incompatible with configured values
	  could result in the allocation of RTP and RTCP ports that would not get
	  reclaimed later.

	  ASTERISK-24666 #close
	  Reported by Y Ateya

	  Review: https://reviewboard.asterisk.org/r/4323

	  AST-2015-001
	  ........

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	  ........

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2015-01-28 11:34 +0000 [3cccfac399]  Mark Michelson <mmichelson@digium.com>

	* Multiple revisions 431297-431298

	  ........
	    r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan 2015) | 17 lines
	    
	    Mitigate possible HTTP injection attacks using CURL() function in Asterisk.
	    
	    CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection
	    can be performed given properly-crafted URLs.
	    
	    Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may
	    get cURL URLs from user input or remote sources, we have made a patch to Asterisk
	    to prevent such HTTP injection attacks from originating from Asterisk.
	    
	    ASTERISK-24676 #close
	    Reported by Matt Jordan
	    
	    Review: https://reviewboard.asterisk.org/r/4364
	    
	    AST-2015-002
	  ........
	    r431298 | mmichelson | 2015-01-28 11:12:49 -0600 (Wed, 28 Jan 2015) | 3 lines
	    
	    Fix compilation error from previous patch.
	  ........

	  Merged revisions 431297-431298 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 431299 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2015-01-28 06:19 +0000 [f080ca6536]  Sean Bright <sean@malleable.com>

	* media formats: update res_format_attr_opus & silk

	  In r419044, we changed how formats were handled, but the return value
	  of the format_parse_sdp_fmtp functions in res_format_attr_opus and
	  res_format_attr_silk were not updated, causing calls to fail.  Ran
	  into this when getting codec_opus working with Asterisk 13.

	  Once the return value was corrected, we were crashing in opus_getjoint
	  because of NULL format attributes.  I've fixed this as well in this
	  patch.

	  Review: https://reviewboard.asterisk.org/r/4371/
	  ........

	  Merged revisions 431267 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-27 22:29 +0000 [69e107b24e]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration: Fix reload race condition.

	  Performing a CLI "module reload" command when there are new pjsip.conf
	  registration objects defined frequently failed to load them correctly.

	  What happens is a race condition between res_pjsip pushing its reload into
	  an asynchronous task processor task and the thread that does the rest of
	  the reloads when it gets to reloading the res_pjsip_outbound_registration
	  module.  A similar race condition happens between a reload and the CLI/AMI
	  show registrations commands.  The reload updates the current_states
	  container and the CLI/AMI commands call get_registrations() which builds a
	  new current_states container.

	  * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous()
	  instead of ast_sip_push_task() to eliminate two threads processing config
	  reloads at the same time.

	  * Made get_registrations() not replace the global current_states container
	  so the CLI/AMI show registrations command cannot interfere with reloading.
	  You could never add/remove objects in the container without the
	  possibility of the container being replaced out from under you by
	  get_registrations().

	  * Added a registration loaded sorcery instance observer to purge any dead
	  registration objects since get_registrations() cannot do this job anymore.
	  The struct ast_sorcery_instance_observer callbacks must be used because
	  the callback happens inline with the load process.  The struct
	  ast_sorcery_observer callbacks are pushed to a different thread.

	  * Added some global current_states NULL pointer checks in case the
	  container disappears because of unload_module().

	  * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded
	  callbacks guaranteed to be called before any struct
	  ast_sorcery_observer.loaded callbacks will be called.

	  * Moved the check for non-reloadable objects to before the sorcery
	  instance loading callbacks happen to short circuit unnecessary work.
	  Previously with non-reloadable objects, the sorcery instance
	  loading/loaded callbacks would always happen, the individual wizard
	  loading/loaded would be prevented, and the non-reloadable type logging
	  message would be logged for each associated wizard.

	  ASTERISK-24729 #close
	  Review: https://reviewboard.asterisk.org/r/4381/
	  ........

	  Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-27 16:58 +0000 [c7591ef6bc]  Kevin Harwell <kharwell@digium.com>

	* tcptls: Bad file descriptor error when reloading chan_sip

	  While running through some scenarios using chan_sip and tcp a problem would
	  occur that resulted in a flood of bad file descriptor messages on the cli:

	  tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor

	  The message is received because the underlying socket has been closed, so is
	  valid. This is probably happening because unloading of chan_sip is not atomic.
	  That however is outside the scope of this patch. This patch simply stops the
	  logging of multiple occurrences of that message.

	  ASTERISK-24728 #close
	  Reported by: Thomas Thompson
	  Review: https://reviewboard.asterisk.org/r/4380/
	  ........

	  Merged revisions 431218 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-01-27 13:31 +0000 [e826cb8a26]  Jonathan Rose <jrose@digium.com>

	* Manager: Fix Manager Action ModuleLoad to give correct response when reloading

	  Prior to this patch, ModuleLoad would respond with an error indicating that
	  the requested module wasn't found in spite of finding and reloading the
	  module.

	  Review: https://reviewboard.asterisk.org/r/4373/
	  ASTERISK-24721 #close
	  ........

	  Merged revisions 431153 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431201 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-27 13:22 +0000 [3b0f03ef7b]  Kevin Harwell <kharwell@digium.com>

	* chan_sip: stale nonce causes failure

	  When refreshing (with a small expiration) a registration that was sent to
	  chan_sip the nonce would be considered stale and reject the registration.
	  What was happening was that the initial registration's "dialog" still existed
	  in the dialogs container and upon refresh the dialog match algorithm would
	  choose that as the "dialog" instead of the newly created one. This occurred
	  because the algorithm did not check to see if the from tag matched if
	  authentication info was available after the 401. So, it ended up assuming
	  the original "dialog" was a match and stopped the search. The old "dialog"
	  of course had an old nonce, thus the stale nonce message.

	  This fix attempts to leave the original functionality alone except in the case
	  of a REGISTER. If a REGISTER is received if searches for an existing "dialog"
	  matching only on the callid. If the expires value is low enough it will reuse
	  dialog that is there, otherwise it will create a new one.

	  ASTERISK-24715 #close
	  Reported by: John Bigelow
	  Review: https://reviewboard.asterisk.org/r/4367/
	  ........

	  Merged revisions 431187 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-27 13:12 +0000 [e62bd46511]  Corey Farrell <git@cfware.com> (license 5909)

	* res_pjsip: make it unloadable (take 2)

	  Due to the original patch causing memory corruptions it was removed until the
	  problem could be resolved. This patch is the original patch plus some added
	  locking around stasis router subcription that was needed to avoid the memory
	  corruption.

	  Description of the original problem and patch (still applicable):

	  The res_pjsip module was previously unloadable. With this patch it can now
	  be unloaded.

	  This patch is based off the original patch on the issue (listed below) by Corey
	  Farrell with a few modifications. Namely, removed a few changes not required to
	  make the module unloadable and also fixed a bug that would cause asterisk to
	  crash on unloading.

	  This patch is the first step (should hopefully be followed by another/others at
	  some point) in allowing res_pjsip and the modules that depend on it to be
	  unloadable. At this time, res_pjsip and some of the modules that depend on
	  res_pjsip cannot be unloaded without causing problems of some sort.

	  The goal of this patch is to get res_pjsip and only res_pjsip to be able to
	  unload successfully and/or shutdown without incident (crashes, leaks, etc...).
	  Other dependent modules may still cause problems on unload.

	  Basically made sure, with the patch applied, that res_pjsip (with no other
	  dependent modules loaded) could be succesfully unloaded and Asterisk could
	  shutdown without any leaks or crashes that pertained directly to res_pjsip.

	  ASTERISK-24485 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4363/
	  patches:
	    pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909)
	  ........

	  Merged revisions 431179 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-27 11:48 +0000 [94eebd5ba5]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge: Repeatedly starting and stopping recording ref leaks the recording channel.

	  Starting and stopping conference recording more than once causes the
	  recording channels to be leaked.  For v13 the channels also show up in the
	  CLI "core show channels" output.

	  * Reworked and simplified the recording channel code to use
	  ast_bridge_impart() instead of managing the recording thread in the
	  ConfBridge code.  The recording channel's ref handling easily falls into
	  place and other off nominal code paths get handled better as a result.

	  ASTERISK-24719 #close
	  Reported by: John Bigelow

	  Review: https://reviewboard.asterisk.org/r/4368/
	  Review: https://reviewboard.asterisk.org/r/4369/
	  ........

	  Merged revisions 431135 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2015-01-27 11:34 +0000 [a43d24a9d3]  Joshua Colp <jcolp@digium.com>

	* bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

	  This change fixes two issues:

	  1. During a swap operation bridging added the new channel before having the swap channel
	  leave. This was not handled in bridge_native_rtp and could result in a channel not getting
	  reinvited back to Asterisk. After this change the swap channel will leave first and the
	  new channel will then join.

	  2. If a re-invite was received after a session had been established any upstream elements
	  (such as bridge_native_rtp) were not notified that they may want to re-evaluate things.
	  After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs
	  and upstream can react.

	  AST-1524 #close

	  Review: https://reviewboard.asterisk.org/r/4378/
	  ........

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2015-01-27 11:21 +0000 [fb8a2e0399]  Matt Jordan <mjordan@digium.com>

	* ARI: Improve wiki documentation

	  This patch improves the documentation of ARI on the wiki. Specifically, it
	  addresses the following:
	  * Allowed values and allowed ranges weren't documented. This was particularly
	    frustrating, as Asterisk would reject query parameters with disallowed values
	    - but we didn't tell anyone what the allowed values were.
	  * The /play/id operation on /channels and /bridges failed to document all of
	    the added media resource types.
	  * Documentation for creating a channel into a Stasis application failed to
	    note when it occurred, and that creating a channel into Stasis conflicts with
	    creating a channel into the dialplan.
	  * Some other minor tweaks in the mustache templates, including italicizing the
	    parameter type, putting the default value on its own sub-bullet, and some
	    other nicities.

	  Review: https://reviewboard.asterisk.org/r/4351
	  ........

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2015-01-27 11:16 +0000 [aa8fd7d1b9]  Matt Jordan <mjordan@digium.com>

	* app_confbridge: Restore user's menu name to CLI output of 'confbridge list'

	  When issuing a 'confbridge list XXXX' CLI command, the resulting output no
	  longer displays the menu associated with a ConfBridge participant.

	  The issue was caused by ASTERISK-22760. When that patch was done, it removed
	  the copying of the menu name associated with the user from the actual user
	  profile.

	  This patch fixes the issue by copying the menu name over to the user profile
	  when the menu hooks are applied to the user. Since that function now does a
	  little bit more than just apply the hooks, the name of the function has been
	  changed to cover the copying of the menu name over as well.

	  In addition, there is a disparity between the menu name length as it is stored
	  on the conf_menu structure and the confbridge_user structure; this patch makes
	  the lengths match so that a strcpy can be used.

	  Review: https://reviewboard.asterisk.org/r/4372/

	  ASTERISK-24723 #close
	  Reported by: Steve Pitts
	  ........

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2015-01-27 05:47 +0000 [2504f97b01]  Joshua Colp <jcolp@digium.com>

	* res_parking: Fix crash due to race condition when unloading.

	  There is currently a race condition when unloading the res_parking
	  module. Depending on the will of the universe the subscription
	  invocation may occur AFTER the module is unloaded. This is because
	  the module does NOT use stasis_unsubscribe_and_join when terminating
	  the subscription. It merely uses stasis_unsubscribe.

	  This change makes it use stasis_unsubscribe_and_join which is documented
	  for usage in this exact scenario.

	  AST-1520 #close

	  Review: https://reviewboard.asterisk.org/r/4375/
	  ........

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2015-01-26 08:50 +0000 [965777ccfc]  David M. Lee <dlee@digium.com>

	* Various fixes for OS X

	  This patch addresses compilation errors on OS X. It's been a while, so
	  there's quite a few things.

	   * Fixed __attribute__ decls in route.h to be portable.
	   * Fixed htonll and ntohll to work when they are defined as macros.
	   * Replaced sem_t usage with our ast_sem wrapper.
	   * Added ast_sem_timedwait to our ast_sem wrapper.
	   * Fixed some GCC 4.9 warnings using sig*set() functions.
	   * Fixed some format strings for portability.
	   * Fixed compilation issues with res_timing_kqueue (although tests still fail
	     on OS X).
	   * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue
	     on OS X).

	  ASTERISK-24539 #close
	  Reported by: George Joseph

	  ASTERISK-24544 #close
	  Reported by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4327/
	  ........

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2015-01-25 07:43 +0000 [a8ae5a7bcb]  Matt Jordan <mjordan@digium.com>

	* dynamic realtime: Updates fail to work due to update fields being passed over

	  When a crash was fixed due to usage of the REALTIME function in r423003, a
	  regression was introduced into ast_update2_realtime where the update fields
	  passed to the function would be skipped and the lookup field processed twice.

	  The use of this function is a bit interesting: A variable argument list is
	  used with two sentinel values - the first marks the end of the lookup
	  fields/values; the second marks the end of the update fields/values.
	  Unfortunately, ast_update2_realtime parses over the lookup fields twice, as
	  opposed to parsing over the update fields. This causes the lookups to succeed,
	  but the updates itself to have no effect.

	  Note that the most common instance of this problem occurred in app_voicemail
	  during the updating of a mailbox password.

	  Thanks to the issue reporter, Paddy Grice, for pointing out the problem.

	  Review: https://reviewboard.asterisk.org/r/4356/

	  ASTERISK-24231

	  ASTERISK-24626 #close
	  Reported by: Paddy Grice
	  ........

	  Merged revisions 431072 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431073 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-23 14:17 +0000 [b69b0d12ee]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge: Shorten CBRec channel names to CBRec/<conf_name>-<seq-num>

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431055 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-23 14:14 +0000 [c780223507]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge: Make CBRec channel names more unique.

	  Channel names should be different from other channels in the system while
	  the channel exists.

	  * Use a sequence number for CBRec channels instead of a random number
	  because the same random number could be picked again for the next CBRec
	  channel.
	  ........

	  Merged revisions 431052 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431053 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-23 13:51 +0000 [b38be992b1]  Richard Mudgett <rmudgett@digium.com>

	* app_confbridge: Whitespace

	  Because there is sometimes no sence to any whitespace.
	  ........

	  Merged revisions 431049 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 431050 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431051 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-23 12:46 +0000 [89610adda5]  David M. Lee <dlee@digium.com>

	* Add depend on pjproject to res_pjsip_config_wizard.c
	  ........

	  Merged revisions 431030 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-23 09:21 +0000 [ca02121ef7]  Kevin Harwell <kharwell@digium.com>

	* Investigate and fix memory leaks in Asterisk

	  Fixed memory leaks that were found in Asterisk.

	  ASTERISK-24693 #close
	  Reported by:  Kevin Harwell
	  Review: https://reviewboard.asterisk.org/r/4347/
	  ........

	  Merged revisions 430999 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431010 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-23 09:13 +0000 [49cbfa7de6]  Walter Doekes <walter+asterisk@wjd.nu>

	* Fix typo's (retrieve, specified, address).
	  ........

	  Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-23 08:39 +0000 [874cb5615d]  HZMI8gkCvPpom0tM (License 6658)

	* chan_sip: Case insensitive comparison of "defaultuser" parameter.

	  All the other configuration options are case insensitive, so this one
	  should be too.

	  ASTERISK-24355 #close
	  Reported by: HZMI8gkCvPpom0tM
	  patches:
	    ast.patch uploaded by HZMI8gkCvPpom0tM (License 6658)
	  ........

	  Merged revisions 430993 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430994 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430995 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-22 13:30 +0000 [9bff4eeca3]  Richard Mudgett <rmudgett@digium.com>

	* Bridge core: Pass a ref with the swap channel when joining a bridge.

	  When code imparts a channel into a bridge to swap with another channel, a
	  ref needs to be held on the swap channel to ensure that it cannot
	  dissapear before finding it in the bridge.

	  * The ast_bridge_join() swap channel parameter now always steals a ref for
	  the swap channel.  This is the only change to the bridge framework's
	  public API semantics.

	  * bridge_channel_internal_join() now requires the bridge_channel->swap
	  channel to pass in a ref.

	  ASTERISK-24649
	  Reported by: John Bigelow

	  Review: https://reviewboard.asterisk.org/r/4354/
	  ........

	  Merged revisions 430975 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-22 13:14 +0000 [e67ca431ee]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Minor code cleanup.

	  * Add an allocation failure check and assert in
	  sip_outbound_registration_response_cb().

	  * Made sip_outbound_registration_state_destroy() handle partially created
	  state objects from sip_outbound_registration_state_alloc().

	  Review: https://reviewboard.asterisk.org/r/4366/
	  ........

	  Merged revisions 430957 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-22 12:10 +0000 [49f405fe4c]  Scott Griepentrog <sgriepentrog@digium.com>

	* stasis transfer: fix a race condition on stasis bridge push

	  After a bridge transfer completes where a local replacement
	  channel is used, a stasis transfer message with the details
	  of the transfer is sent.  This is processed by stasis which
	  then sets the stasis app name and replaced channel snapshot
	  on the replacement channel.

	  However, since a separate thread was already started to run
	  stasis on the new replacement channel, a race was on to see
	  if the message processing would be completed before the app
	  name was needed, otherwise the channel would be hung up.

	  This change moves the calls used to set the stasis app name
	  and the replace snapshot to the bridge_stasis_push function
	  callback from the bridge transfer logic, allowing the steps
	  to be completed earlier and more deterministically, and the
	  race elimianted.

	  NOTE: the swap channel parameter to bridge_stasis_push (and
	  thus all bridge push callbacks) must always be present when
	  performing a swap with another channel.

	  ASTERISK-24649 #close
	  Reported by: John Bigelow
	  Review: https://reviewboard.asterisk.org/r/4341/
	  ........

	  Merged revisions 430939 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430940 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-22 08:23 +0000 [7fcc9ce8bc]  Gareth Palmer (License 5169)

	* apps/app_voicemail: Trigger MWI notification with MixMonitor m() option

	  The MixMonitor m() option allows a recording to be pushed to a specific
	  voicemail mailbox. If the message is delivered to the mailbox's INBOX, however,
	  no MWI notification is currently raised.

	  This patch corrects the issue by properly calling notify_new_state from the
	  msg_create_from_file function. This will cause MWI to be triggered if the
	  message was placed in the mailbox's INBOX.

	  ASTERISK-24709 #close
	  Reported by: Gareth Palmer
	  patches:
	    app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169)
	  ........

	  Merged revisions 430920 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430921 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430922 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-21 15:57 +0000 [38738a7316]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_outbound_registration.c: Move unref to a better place.

	  Move an unconditional unref of client_state so it doesn't look like it
	  could be used after the last ref has destroyed it.
	  ........

	  Merged revisions 430902 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-21 07:36 +0000 [5835bf7a7f]  Matt Jordan <mjordan@digium.com>

	* channels/chan_sip: Fix registration leak during reload

	  When the SIP registrations were migrated to using ao2 in what was then trunk,
	  the explicit destruction of the registrations on module reload was removed and
	  not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the
	  issue reporter, on ASTERISK-24673 confirmed that the reference in the
	  registry_list container was being leaked.

	  Since the purpose of cleanup_all_regs is to prep a registration for
	  destruction, this function now calls an ao2_callback function callback with the
	  OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the registrations.
	  This cleans up each registration, and also removes it from the registration
	  container registry_list.

	  Review: https://reviewboard.asterisk.org/r/4355/

	  ASTERISK-24640 #close
	  Reported by: Max Man

	  ASTERISK-24673 #close
	  Reported by: Stefan Engström
	  Tested by: Stefan Engström
	  ........

	  Merged revisions 430864 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-21 07:27 +0000 [958a41a884]  Matt Jordan <mjordan@digium.com>

	* AMI: Add documentation for the missing Cdr/CEL events.

	  This patch adds AMI event documentation for the Cdr and CEL AMI events.

	  Note that while these events do share fields with each other and with other
	  channel related events, they do not contain all of the fields in a standard
	  channel snapshot, nor is the description of the fields identical. As such,
	  the patch opts for documentation for each field, for each event.

	  Review: https://reviewboard.asterisk.org/r/4350/

	  ASTERISK-24671 #close
	  Reported by: Dan Jenkins
	  ........

	  Merged revisions 430862 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-21 07:12 +0000 [4740ef50f4]  Matt Jordan <mjordan@digium.com>

	* apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values

	  The Dial application has some interesting options with the mid-call Macro (M)
	  and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific
	  values, the Dial application will take some action upon the channels involved
	  in the dial operation (such as hanging up a particular party, etc.) The Dial
	  application ensures that a Stasis message is published in the event that
	  MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so
	  that there is a corresponding DialEnd event published in AMI/ARI for the
	  DialBegin event that preceeded it.

	  A bug exists where that same DialEnd event will be published on Stasis even if
	  the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial
	  application cares about. This causes two DialEnd events to be published - one
	  with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all
	  sorts of wrong.

	  This patch fixes the bug by ensuring that we only publish a DialEnd message to
	  Stasis if the Dial application's mid-call Macro/GoSub returns something that
	  Dial cares about.

	  Review: https://reviewboard.asterisk.org/r/4336

	  ASTERISK-24682 #close
	  Reported by: Matt Jordan
	  ........

	  Merged revisions 430842 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-21 07:06 +0000 [228fdb3f4e]  Matt Jordan <mjordan@digium.com>

	* main/rtp_engine: Format NTP timestamps as unsigned longs

	  When the RTCP reports are created, the NTP timestamps are stored as strings,
	  as JSON does not have an integer type long enough to store the value. However,
	  on 32-bit systems, a signed long may overflow for some portion of the
	  timestamp.

	  This patch corrects the overflow by formatting the timestamps as unsigned
	  longs.
	  ........

	  Merged revisions 430840 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430841 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-20 11:15 +0000 [804ab70f9d]  Ashley Sanders <asanders@digium.com>

	* ARI: Fixed crash that occurred when updating a bridge when the optional query parameter 'name' was not supplied.

	  Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk.

	  ASTERISK-24560 #close
	  Reported By: Kinsey Moore

	  Review: https://reviewboard.asterisk.org/r/4349/
	  ........

	  Merged revisions 430818 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430820 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-20 10:59 +0000 [e4738a59eb]  Richard Mudgett <rmudgett@digium.com>

	* CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge.

	  Calling ast_channel_bridge_peer() cannot be done while holding any channel
	  locks.  The reported issue hit the deadlock in chan_iax2, but an audit of
	  the ast_channel_bridge_peer() calls found three more locations where the
	  same deadlock can occur.

	  * Made CHANNEL(peer), res_fax, and the SNMP agent not call
	  ast_channel_bridge_peer() with any channel locked.  For CHANNEL(peer) I
	  had to rework the logic to not hold the channel lock.

	  * Made chan_iax2 no longer call ast_channel_bridge_peer().  It was done
	  for legacy reasons that no longer apply.

	  * Removed the iax.conf forcejitterbuffer option.  It is now always enabled
	  when the jitterbuffer option is enabled.  If you put a jitter buffer on a
	  channel it will be on the channel.

	  ASTERISK-24600 #close
	  Reported by: Jeff Collell

	  Review: https://reviewboard.asterisk.org/r/4342/
	  ........

	  Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-19 20:41 +0000 [14b8e03dad]  Ben Klang (License 5876)

	* contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hosts

	  On Debian based systems, the install_prereq tool uses a search command on
	  Debian that results in selecting both 64-bit and 32-bit packages. Besides the
	  waste of disk space, this can actually cause aptitude use 100% of memory on a
	  VM with 1GB of RAM as it tried to work out all of the 32-bit package
	  dependencies.

	  This patch filters out the 32-bit packages on a 64-bit machine, and leaves
	  32-bit machines alone.

	  ASTERISK-24048 #close
	  Reported by: Ben Klang
	  Tested by: Ben Klang, Matt Jordan
	  patches:
	    install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876)
	  ........

	  Merged revisions 430798 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430799 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430800 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-19 20:33 +0000 [112bf1597e]  LEI FU (License 6640)

	* app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend

	  When using ODBC or IMAP storage, temporary files created on the file system
	  must be disposed of using the DISPOSE macro. The DELETE macro will map to a
	  deletion function for the backend storage, but does not clean up any local
	  files created as a result of the operation.

	  When using voicemail with the operator and review options enabled, pressing
	  0 to enter the menu, followed by 1 to save the message, followed by any
	  other DTMF press to delete the message, will result in the temporary file
	  lingering on the file system.

	  This patch properly calls DISPOSE after the DELETE. This causes the local
	  file to be disposed of.

	  ASTERISK-24288 #close
	  Reported by: LEI FU
	  patches:
	    voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640)
	  ........

	  Merged revisions 430795 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430796 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430797 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-19 12:15 +0000 [7dc784ffa9]  Mark Michelson <mmichelson@digium.com>

	* Call extension state callbacks at hint creation.

	  When a hint gets created, any subsequent device or presence
	  state changes result in extension status events getting sent
	  out to interested parties. However, at the time of hint creation,
	  no such event gets sent out, so watchers of extension state are
	  potentially left in the dark until the first state change after
	  hint creation.

	  Patch contributed by John Hardin (License #6512)
	  ........

	  Merged revisions 430776 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-19 07:19 +0000 [e43912f3f3]  Joshua Colp <jcolp@digium.com>

	* res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions.

	  The first thing this patch fixes is UAS dialogs. Previously if a transport was
	  configured on an endpoint and an inbound session was created there was no guarantee
	  that requests sent on the dialog would use the correct transport and address
	  information. This has now been fixed so an explicitly configured transport
	  is taken into account.

	  The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed
	  module attempts to determine what transport a message should go out on and what
	  addressing information should go into the message itself. In a scenario where
	  multiple transports exist bound to the same IP address but a different port the
	  code would incorrectly alter the transport and change the message to the wrong
	  transport. This change makes the res_pjsip_multihomed module smarter so it will
	  only change the transport and address information in the message when it is
	  possible and makes sense.

	  ASTERISK-24615 #close
	  Reported by: David Justl

	  Review: https://reviewboard.asterisk.org/r/4331/
	  ........

	  Merged revisions 430755 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430756 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-16 18:35 +0000 [07e2a48ab1]  Kevin Harwell <kharwell@digium.com>

	* REVERTING res_pjsip: make it unloadable

	  Due to the original patch causing memory corruptions the patch is
	  being removed until the problem can be resolved.
	  ........

	  Merged revisions 430734 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-16 16:14 +0000 [1111944afb]  Mark Michelson <mmichelson@digium.com>

	* Change PJProject version requirement for ca_list_path transport option in CHANGES file.
	  ........

	  Merged revisions 430716 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430717 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-16 16:13 +0000 [831acba826]  Mark Michelson <mmichelson@digium.com>

	* Fix problem where a hung channel could occur on a failed blind transfer.

	  Different clients react differently to being told that a blind transfer
	  has failed. Some will simply send a BYE and be done with it. Others will
	  attempt to reinvite themselves back onto the call.

	  In the latter case, we were creating a new channel and then leaving it to
	  sit forever doing nothing. With this code change, that new channel will
	  not be created and the dialog with the transferring channel will be cleaned
	  up properly.

	  ASTERISK-24624 #close
	  Reported by Zane Conkle

	  Review: https://reviewboard.asterisk.org/r/4339
	  ........

	  Merged revisions 430714 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430715 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-16 15:46 +0000 [023fa0f9e8]  cloos <cloos@jhcloos.com> (License #5956)

	* Add support for the ca_list_path option for PJSIP transports.

	  This allows for a path to be specified that has a collection of CA
	  certificates in it.

	  ASTERISK-24575 #close
	  Reported by cloos
	  Patches:
	  	pj-ca-path-trunk.diff uploaded by cloos (License #5956)

	  Review: https://reviewboard.asterisk.org/r/4344
	  ........

	  Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-15 11:36 +0000 [a8ea2f9287]  Richard Mudgett <rmudgett@digium.com>

	* res_fax.c, res_fax_spandsp.c: Remove redundant locking.

	  When FAX was developed, apparently the faxregistry.container used to be a
	  linked list that was converted to an ao2 container.  Some of the
	  replacement ao2 container operations still had explicit lock/unlocks
	  around them.

	  Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the
	  channel even though the routine did not lock the channel and other code
	  paths in the routine do not unlock the channel.

	  Review: https://reviewboard.asterisk.org/r/4340/
	  ........

	  Merged revisions 430687 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430688 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-15 11:28 +0000 [9b1c36d3fa]  Richard Mudgett <rmudgett@digium.com>

	* res_fax.c, res_fax_spandsp.c: Fix some curlies on the end of function definitions.
	  ........

	  Merged revisions 430685 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-15 06:10 +0000 [1e605d950b]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_outbound_registration: Fix race condition when reloading and listing registrations.

	  Due to the split of outbound registration state from configuration it is possible during
	  a reload for a "pjsip show registrations" CLI command to be executed which gets an older
	  snapshot of the configuration. This configuration may include outbound registrations which
	  have been removed due to a reload operation occurring at the same time. The code for
	  printing the outbound registration did not take this into account but now it does.

	  AST-1506 #close

	  Review: https://reviewboard.asterisk.org/r/4338/
	  ........

	  Merged revisions 430664 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-14 20:19 +0000 [f11fb76205]  abelbeck <lonnie@abelbeck.com> (License 5903)

	* configure: If cross-compiling, assume we have working semaphores

	  The Asterisk 13 configure.ac checks for HAS_WORKING_SEMAPHORE but does not have
	  an option for cross-compiling so it fails with an exit. Since we're cross-
	  compiling, we can't exactly go looking for the header. The semaphore.h header
	  is relatively common:
	  * It's part of the POSIX standard
	  * It's part of GNU C Library
	  As such, we assume that it will be present when cross-compiling.

	  As such, this patch defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling
	  is detected.

	  If you're cross-compiling to a platform that doesn't support this, then make
	  sure you re-define this to 0.

	  ASTERISK-24663 #close
	  Reported by: abelbeck
	  patches:
	    asterisk-13-anonymous-semaphores.patch uploaded by abelbeck (License 5903)
	  ........

	  Merged revisions 430646 from http://svn.asterisk.org/svn/asterisk/branches/13


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2015-01-14 17:15 +0000 [49542a794b]  Corey Farrell <git@cfware.com> (license 5909)

	* res_pjsip: make it unloadable

	  The res_pjsip module was previously unloadable. With this patch it can now
	  be unloaded.

	  This patch is based off the original patch on the issue (listed below) by Corey
	  Farrell with a few modifications. Namely, removed a few changes not required to
	  make the module unloadable and also fixed a bug that would cause asterisk to
	  crash on unloading.

	  This patch is the first step (should hopefully be followed by another/others at
	  some point) in allowing res_pjsip and the modules that depend on it to be
	  unloadable. At this time, res_pjsip and some of the modules that depend on
	  res_pjsip cannot be unloaded without causing problems of some sort.

	  The goal of this patch is to get res_pjsip and only res_pjsip to be able to
	  unload successfully and/or shutdown without incident (crashes, leaks, etc...).
	  Other dependent modules may still cause problems on unload.

	  Basically made sure, with the patch applied, that res_pjsip (with no other
	  dependent modules loaded) could be succesfully unloaded and Asterisk could
	  shutdown without any leaks or crashes that pertained directly to res_pjsip.

	  ASTERISK-24485 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4311/
	  patches:
	    pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909)
	  ........

	  Merged revisions 430628 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-14 14:39 +0000 [67234b3ee2]  Mark Michelson <mmichelson@digium.com>

	* Prevent slow graceful shutdown when outbound publications never started.

	  The code was missing the case for explicitly destroying an outbound publication
	  when Asterisk had never actually published anything. The result was that Asterisk
	  would hang for a while on a graceful shutdown.

	  With this change, the case is taken into account, and on a graceful shutdown, these
	  publications are destroyed without the need to actually send a PUBLISH request.

	  ASTERISK-24655 #close
	  Reported by Kevin Harwell

	  Review: https://reviewboard.asterisk.org/r/4325
	  ........

	  Merged revisions 430608 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430609 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-14 09:40 +0000 [3eec8e4c44]  Diederik de Groot (License 6600)

	* build_tools/mkpkgconfig: Fix Cflags concatenation error in asterisk.pc

	  The mkpkgconfig script incorrectly concatenates Cflags options together. As an
	  example, the following:
	  Cflags: -I/usr/include/libxml2 -g3

	  Is instead generated as:
	  Cflags: -I/usr/include/libxml2-g3

	  This patch corrects the generation of Cflags in mkpkgconfig such that the
	  Cflags options are output correctly.

	  Review: https://reviewboard.asterisk.org/r/3707/

	  ASTERISK-23991 #close
	  Reported by: Diederik de Groot
	  patches:
	    fix_mkpkgconfig.diff uploaded by Diederik de Groot (License 6600)
	  ........

	  Merged revisions 430589 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430590 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430591 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-13 12:17 +0000 [1780de95e4]  Richard Mudgett <rmudgett@digium.com>

	* app_macro: Don't restore the calling location on a channel redirect.

	  v11: If a channel redirect to a macro exten of a macro that is active
	  happens, the redirect location doesn't get executed.  Instead the original
	  macro location is restored and gets reexecuted.

	  v13: An additional effect happens if a parked call times out to an
	  extension in the macro that parked the call then the macro is reexecuted
	  instead of the expected park return location.

	  * Made not restore the macro calling location on an
	  AST_SOFTHANGUP_ASYNCGOTO.

	  * Increased the locked channel range when setting up the macro execution
	  environment to cover things that should be done while the channel is
	  locked.

	  * Removed unnecessary NULL tests before calling ast_free() in
	  _macro_exec().

	  ASTERISK-23850 #close
	  Reported by: Andrew Nagy

	  Review: https://reviewboard.asterisk.org/r/4292/
	  ........

	  Merged revisions 430564 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430565 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-13 06:09 +0000 [0e631a541d]  Joshua Colp <jcolp@digium.com>

	* chan_pjsip: Add configure check for 'pjsip_get_dest_info' function.

	  The 'pjsip_get_dest_info' function is used to determine if the signaling transport
	  of the dialog is secure or not. This function was added in PJSIP 2.3 and does not
	  exist in earlier versions.

	  This configure check allows Asterisk to build and run with older versions at the
	  loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of
	  this argument will require upgrading to PJSIP 2.3.

	  ASTERISK-24665 #close
	  Reported by: Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/4329/
	  ........

	  Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-12 13:13 +0000 [4dd6b6ff59]  Richard Mudgett <rmudgett@digium.com>

	* AMI: Revert non-backwards compatible changes from earlier commit.

	  * Reverted the change to astman_send_listack() to not use the listflag
	  parameter and always set the value to "Start" so the start capitalization
	  is consistent.  Unfortunately changing the case of a returned value is not
	  a backward compatible change so for now FAXSessions is going to have to
	  remain inconsistent with all of the other AMI list actions.

	  * Reverted the minor protocol error fix in action_getconfig() when no
	  requested categories are found.  Each line needs to be formatted as
	  "Header: text".

	  Caught by the testsuite.

	  ASTERISK-24049
	  ........

	  Merged revisions 430528 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430529 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-12 12:28 +0000 [aa7e06f797]  Niklas Larsson (License 5068)

	* configs/samples/features.conf.sample: Document attended transfer DTMF options

	  The sample config was missing the configuration options for DTMF attended
	  transfer completion scenarios. The configuration options 'atxferabort',
	  'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the
	  appropriate configuration file.

	  ASTERISK-24678 #close
	  Reported by: Niklas Larsson
	  patches:
	    features.conf.sample.diff uploaded by Niklas Larsson (License 5068)
	  ........

	  Merged revisions 430526 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-12 12:09 +0000 [c7ea108e02]  Richard Mudgett <rmudgett@digium.com>

	* Revert -r430452 It needs to be redone for the next major AMI version change instead.

	  ASTERISK-24049


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-12 12:01 +0000 [9065488ddd]  Michael L. Young (license 5026)

	* main/syslog: Allow dynamic logs, such as security events, to log to the syslog

	  The security event log uses a dynamic log level (SECURITY) that is registered
	  with the Asterisk logging core. Unfortunately, the syslog would ignore log
	  statements that had a dynamic log level associated with them. Because the
	  syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic
	  log entries sent to the syslog as logs with a level of NOTICE.

	  ASTERISK-20744 #close
	  Reported by: Michael Keuter
	  Tested by: Michael L. Young, Jacek Konieczny
	  patches:
	    asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026)
	  ........

	  Merged revisions 430506 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430507 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430508 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-12 09:18 +0000 [b38acbce6e]  Kristian Hogh (License 6639)

	* funcs/func_curl: Fix memory leak when CURLOPT channel datastore is destroyed

	  When the channel datastore associated with the usage of CURLOPT on a specific
	  channel is freed, the underlying structure holding the list of options is not
	  disposed of. This patch properly frees the structure in the datastore .destroy
	  callback.

	  ASTERISK-24672 #close
	  Reported by: Kristian Hogh
	  patches:
	    func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639)
	  ........

	  Merged revisions 430487 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430488 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430489 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-09 16:09 +0000 [fba836cc02]  Scott Griepentrog <sgriepentrog@digium.com>

	* sip_to_pjsip: improve ability to parse input files

	  General improvements to SIP to PJSIP conversion utility:

	  1) track default section of input file to allow parsing
	     an include file that doesn't specify a [section]

	  2) informatively handle case of assignment without [section]

	  3) correctly handle getting sections from included files
	     - [section]'s are inherited by included file

	  4) provide null string as default transport bind ip

	  5) gracefully handle missing portions of registration string

	  6) denote steps of operation during conversion and confirm
	     top level files as a convenience

	  ASTERISK-24474 #close
	  Review: https://reviewboard.asterisk.org/r/4280/
	  Reported by: John Kiniston
	  ........

	  Merged revisions 430469 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-09 15:45 +0000 [5b30938394]  Scott Griepentrog <sgriepentrog@digium.com>

	* app_bridge: return to the next dialplan priority

	  When app_bridge grabs a channel and puts it into
	  a bridge, the channel should then continue where
	  it left off in the dialplan after the bridge has
	  ended.   Although it stores the current dialplan
	  location as an after bridge goto on the channel,
	  it was executing the same priority again instead
	  of going to the next priority.   By swapping the
	  "specific" version of bridge_set_after_goto with
	  bridge_set_after_go_on, the next priority in the
	  dialplan is executed instead.

	  ASTERISK-24637 #close
	  Review: https://reviewboard.asterisk.org/r/4322/
	  Reported by: John Bigelow
	  ........

	  Merged revisions 430467 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430468 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-09 12:53 +0000 [ef34a05f21]  Richard Mudgett <rmudgett@digium.com>

	* AMI: Remove no longer used parameter from astman_send_listack().

	  Follow-up issue to -r430435 from reviewboard review.

	  ASTERISK-24049
	  Review: https://reviewboard.asterisk.org/r/4315/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-09 12:16 +0000 [52a7cdb101]  Richard Mudgett <rmudgett@digium.com>

	* AMI: Make AMI actions that generate event lists consistent.

	  * Made the following AMI actions use list API calls for consistency:
	  Agents
	  BridgeInfo
	  BridgeList
	  BridgeTechnologyList
	  ConfbridgeLIst
	  ConfbridgeLIstRooms
	  CoreShowChannels
	  DAHDIShowChannels
	  DBGet
	  DeviceStateList
	  ExtensionStateList
	  FAXSessions
	  Hangup
	  IAXpeerlist
	  IAXpeers
	  IAXregistry
	  MeetmeList
	  MeetmeListRooms
	  MWIGet
	  ParkedCalls
	  Parkinglots
	  PJSIPShowEndpoint
	  PJSIPShowEndpoints
	  PJSIPShowRegistrationsInbound
	  PJSIPShowRegistrationsOutbound
	  PJSIPShowResourceLists
	  PJSIPShowSubscriptionsInbound
	  PJSIPShowSubscriptionsOutbound
	  PresenceStateList
	  PRIShowSpans
	  QueueStatus
	  QueueSummary
	  ShowDialPlan
	  SIPpeers
	  SIPpeerstatus
	  SIPshowregistry
	  SKINNYdevices
	  SKINNYlines
	  Status
	  VoicemailUsersList

	  * Incremented the AMI version to 2.7.0.

	  * Changed astman_send_listack() to not use the listflag parameter and
	  always set the value to "Start" so the start capitalization is consistent.
	  i.e., The FAXSessions used "Start" while the rest of the system used
	  "start".  The corresponding complete event always used "Complete".

	  * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
	  AMI ActionID for all of its list events.

	  * Fixed off-nominal AMI protocol error in manager_bridge_info(),
	  manager_parking_status_single_lot(), and
	  manager_parking_status_all_lots().  Use of astman_send_error() after
	  responding to the original AMI action request violates the action response
	  pattern by sending two responses.

	  * Fixed minor protocol error in action_getconfig() when no requested
	  categories are found.  Each line needs to be formatted as "Header: text".

	  * Fixed off-nominal memory leak in manager_build_parked_call_string().

	  * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().

	  ASTERISK-24049 #close
	  Reported by: Jonathan Rose

	  Review: https://reviewboard.asterisk.org/r/4315/
	  ........

	  Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-09 08:53 +0000 [77ee23210d]  Kinsey Moore <kmoore@digium.com>

	* res_fax: Add T.38 negotiation timeout option

	  This change makes the T.38 negotiation timeout configurable via
	  't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously
	  hard coded to be 5000 milliseconds.

	  This change also handles T.38 switch failures by aborting the fax since
	  in the case where this can happen, both sides have agreed to switch to
	  T.38 and Asterisk is unable to do so.

	  Review: https://reviewboard.asterisk.org/r/4320/
	  ........

	  Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-08 15:41 +0000 [8786fe13a4]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown

	  If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't 
	  survive.  If you do a 'core (shutdown|restart) now' or asterisk terminates for 
	  some reason, they do.  Here's why...

	  When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to 
	  subscribers for each subscription.  This not only tells the subscribers that the 
	  dialog/state machine is done, it also frees the last reference to the 
	  subscription tree which causes the persistent subscription to get deleted from 
	  astdb.  When asterisk restarts, nothing's left.  Just preventing the delete from 
	  astdb doesn't work because we already told the subscriber to terminate the 
	  dialog so we can't restart it even if it was still in astdb.  Everything works 
	  OK if asterisk terminates unexpectedly because we never send the 'terminated' 
	  message so on restart, the subscription is still in astdb and the subscriber is 
	  none the wiser.

	  This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for 
	  persistent connections.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4318/
	  ........

	  Merged revisions 430397 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430398 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-08 15:38 +0000 [c55f86c69d]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_outbound_registration: Fix reference leak.

	  Every time a registration started, sip_outbound_registration_response_cb bumps 
	  the ref count on client_state then pushes a handle_registration_response task.  
	  handle_registration_response never unreffed it though.  So every time a 
	  registration goes out, the ref count goes up by one.

	  This patch adds the unreffs to handle_registration_response.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4303/
	  ........

	  Merged revisions 430395 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-08 11:51 +0000 [030facce94]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_outbound_registration: Fix several reload issues

	  There are 2 issues with reloading registrations...

	  1.  The 'can_reuse_registration' test wasn't considering the intervals or 
	  expiration in its determination of whether a registration changed or not so if 
	  you changed any of the intervals or the expiration and reloaded, the object 
	  would get reloaded but the actual timers wouldn't change.  
	  can_reuse_registration now does a sorcery diff on the old and new objects 
	  instead of discretely testing certain fields.  Now if you change expiration for 
	  instance, and reload, the timer is updated and re-registration will occur on the 
	  new value.

	  2.  If you mung up your password on an outbound registration you get a permanent 
	  failure.  If you fix the password (on the outbound_auth object) and reload, 
	  nothing tells outbound_registration to try again because the registration itself 
	  didn't change.  This patch adds an observer on the "auth" object type and if any 
	  auth changes, existing registration states are searched and those in a 
	  REJECTED_PERMANENT state are retried.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4304/
	  ........

	  Merged revisions 430373 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430374 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-07 15:26 +0000 [f8c4909eb7]  Kinsey Moore <kmoore@digium.com>

	* ARI: Allow usage of ASYNCGOTO with Stasis()

	  When the AMI Redirect action is used with a channel bridged inside
	  Stasis() and not running a pbx, the channel is hung up instead of
	  proceeding to the desired location in dialplan. This change allows
	  such channels to be Redirected properly by detecting the operation
	  used by Redirect (ASYNCGOTO) and using the code already established
	  for functionality of the ARI channel continue operation.

	  ASTERISK-24591 #close
	  Review: https://reviewboard.asterisk.org/r/4271/
	  ........

	  Merged revisions 430355 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430356 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-07 12:54 +0000 [7f836c1c15]  Mark Michelson <mmichelson@digium.com>

	* Add the ability to continue and originate using priority labels.

	  With this patch, the following two ARI commands

	  POST /channels
	  POST /channels/{id}/continue

	  Accept a new parameter, label, that can be used to continue to or originate
	  to a priority label in the dialplan.

	  Because this is adding a new parameter to ARI commands, the API version of
	  ARI has been bumped from 1.6.0 to 1.7.0.

	  This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks!

	  ASTERISK-24412 #close
	  Reported by Nir Simionovich

	  Review: https://reviewboard.asterisk.org/r/4285
	  ........

	  Merged revisions 430337 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430338 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-07 12:17 +0000 [e83853eebc]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_exten_state: Change 'does not exist' warning to notice

	  The 'new_subscribe: Extension <> does not exist or has no associated hint'
	  is a config issue and doesn't need to clutter up logs with warnings.
	  Changed to notice.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4307/
	  ........

	  Merged revisions 430319 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430321 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-07 12:15 +0000 [8cde7443c2]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_mwi: Change "MWI Subscription failed" message from warning to notice

	  The "MWI Subscription failed" message means the client is trying to subscribe
	  to a mailbox that doesn't exist.  There's no need to clutter up logs with
	  warnings for a client misconfiguration so I changed it to a notice.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4306/
	  ........

	  Merged revisions 430317 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430318 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-07 11:54 +0000 [685f7ef924]  gtjoseph <george.joseph@fairview5.com>

	* func_config: Add ability to retrieve specific occurrence of a variable

	  I guess nobody uses templates with AST_CONFIG because today if you have a
	  context that inherits from a template and you call AST_CONFIG on the context,
	  you'll get the value from the template even if you've overridden it in the
	  context.  This is because AST_CONFIG only gets the first occurrence which is
	  always from the template.

	  This patch adds an optional 'index' parameter to AST_CONFIG which lets you
	  specify the exact occurrence to retrieve, or '-1' to retrieve the last.
	  The default behavior is the current behavior.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4313/
	  ........

	  Merged revisions 430315 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-07 11:45 +0000 [464647d8f8]  Mark Michelson <mmichelson@digium.com>

	* Fix ability to perform a remote attended transfer with PJSIP.

	  This fix has two parts:

	  * Corrected an error message to properly state that external_replaces is an extension. The
	    error message also prints what dialplan context the external_replaces extension was being
	    looked for in.
	  * Corrected the printing of the Replaces: header in an INVITE request. We were duplicating
	    "Replaces: " in the header.

	  ASTERISK-24376 #close
	  Reported by Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4296
	  ........

	  Merged revisions 430313 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-07 10:56 +0000 [56de48107f]  gtjoseph <george.joseph@fairview5.com>

	* config: Add option to NOT preserve effective context when changing a template

	  Let's say you have a template T with variable VAR1 = ON and you have a
	  context C(T) that doesn't specify VAR1.  If you read C, the effective value
	  of VAR1 is ON.  Now you change T VAR1 to OFF and call
	  ast_config_text_file_save.  The current behavior is that the file gets
	  re-written with T/VAR1=OFF but C/VAR1=ON is added.  Personally, I think this
	  is a bug. It's preserving the effective state of C even though I didn't
	  specify C/VAR1 in th first place.  I believe the behavior should be that if
	  I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should
	  continue to follow the inherited state.  Now, if I DID explicitly specify
	  C/VAR1, the it should be preserved even if the template changes.

	  Even though I think the existing behavior is a bug, it's been that way forever
	  so I'm not changing it.  Instead, I've created ast_config_text_file_save2()
	  that takes a bitmask of flags, one of which is to preserve the effective context
	  (the current behavior).  The original ast_config_text_file_save calls *2 with
	  the preserve flag.  If you want the new behavior, call *2 directly without a
	  flag.

	  I've also updated Manager UpdateConfig with a new parameter
	  'PreserveEffectiveContext' whose default is 'yes'.  If you want the new behavior
	  with UpdateConfig, set 'PreserveEffectiveContext: no'.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4297/
	  ........

	  Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-06 21:01 +0000 [0c5234f12a]  Kinsey Moore <kmoore@digium.com>

	* Fix dev-mode build on recent gcc
	  ........

	  Merged revisions 430274 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430275 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-06 16:46 +0000 [220df246d9]  Matt Jordan <mjordan@digium.com>

	* Blocked revisions 430252

	  ........
	  contrib/ast-db-manage: Correct down_revision path for user_eq_phone

	  When the user_eq_phone patch was backported to 13, it referenced the downward
	  revision that the PJSIP optimistic encryption option also references. This
	  creates a multi-path upgrade Exception when generating the SQL files.

	  This patch corrects this in the 13 branch. Note that trunk, which already
	  contained both of these features, is unaffected by this problem.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430254 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-06 11:53 +0000 [8b5bde3e5a]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_mwi: Change warning to notice

	  When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi,
	  if a contact hasn't registered yet, res_pjsip_mwi spits out a warning.
	  This is a perfectly normal situation though and doesn't require something
	  as serious as a warning.  It's also self correcting. The device will start
	  getting mwi as soon as it registers.

	  This patch changes the warning to a notice.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4314/
	  ........

	  Merged revisions 430227 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430228 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-06 11:49 +0000 [5f60ebc004]  gtjoseph <george.joseph@fairview5.com>

	* bridge_native_rtp: Change local/remote message from debug/2 to verb/4

	  Change the "Locally bridged"/"Remotely bridged" messages from dbg/2 to verb/4.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4300/
	  ........

	  Merged revisions 430225 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430226 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-06 11:43 +0000 [fb3c8e3424]  gtjoseph <george.joseph@fairview5.com>

	* outbound_registration: Add 'pjsip send register' and update 'send unregister'

	  The current behavior of 'pjsip send unregister' is to send the unregister
	  (REGISTER with 0 exp) but let the next scheduled register proceed normally.
	  I don't think that's a good idea.  If you unregister, it should stay
	  unregistered until you decide to start registrations again.  So this patch
	  just adds a cancel_registration call to the current unregister_task to
	  cancel the timer.

	  Of course, now you need  a way to start registration again so I've added
	  a 'pjsip send register' command that unregisters and cancels any existing
	  registration (the same as send unregister), then sends an immediate
	  registration and starts the timer back up again.

	  Both changes also ripple to AMI.  There's a new PJSIPRegister command.

	  There's no harm in calling either command repeatedly.  They don't care
	  about the actual state.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4301/
	  ........

	  Merged revisions 430223 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-06 11:29 +0000 [7dc0c88fc6]  gtjoseph <george.joseph@fairview5.com>

	* pjsip cli: Fix sorting of contacts for 'pjsip list contacts'

	  For some reason I was using a hash container instead of a list to gather the
	  contacts for 'pjsip list/show contacts' so even though I had a sort function,
	  the output wasn't sorted.  This patch just changes the hash container to a
	  list container and the contacts now appear sorted in the CLI.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4305/
	  ........

	  Merged revisions 430221 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430222 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-05 16:50 +0000 [0b8fbf9238]  Scott Griepentrog <sgriepentrog@digium.com>

	* bridge: avoid leaking channel during blond transfer pt2

	  A blond transfer to a failed destination, when followed
	  by a recall attempt, lead to a leak of the reference to
	  the destination channel.  In addition to correcting the
	  regression on the previous attempt (r429826) this fixes
	  the leak and two additional reference leaks on failures
	  of bridge_import.

	  ASTERISK-24513 #close
	  Review: https://reviewboard.asterisk.org/r/4302/
	  ........

	  Merged revisions 430199 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 430200 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430201 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-05 11:57 +0000 [e0bd2ca104]  Joshua Colp <jcolp@digium.com>

	* pjsip: Document addition of 'PJSIP_AOR' and 'PJSIP_CONTACT' in CHANGES file.
	  ........

	  Merged revisions 430181 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430182 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-05 11:53 +0000 [f7cf988a82]  Joshua Colp <jcolp@digium.com>

	* pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions.

	  The PJSIP_AOR dialplan function allows inspection of configured AORs including
	  what contacts are currently bound to them.

	  The PJSIP_CONTACT dialplan function allows inspection of contacts in existence.
	  These can include both externally added (by way of registration) or permanent
	  ones.

	  ASTERISK-24341
	  Reported by: xrobau

	  Review: https://reviewboard.asterisk.org/r/4308/
	  ........

	  Merged revisions 430179 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430180 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-31 12:54 +0000 [8d059c3808]  Scott Griepentrog <sgriepentrog@digium.com>

	* rtp_engine: keep payload types in correct range

	  In r428708 additional codecs were added including
	  a payload type of 128 which is outside of nominal
	  range of 0-127.  This change moves changes 128 to
	  96 to avoid causing a pjsip assertion when making
	  a call to an endpoint configured with allow=all.

	  ASTERISK-24367 #close
	  Review: https://reviewboard.asterisk.org/r/4286/



	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430164 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-29 07:14 +0000 [cb6a737359]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Update transport method documentation

	  This updates the documentation for the 'method' configuration option to
	  be more verbose about the behaviors of values 'unspecified' and
	  'default'. They do exactly the same thing which is to select the
	  default as defined by PJSIP which is currently TLSv1.

	  Review: https://reviewboard.asterisk.org/r/4264/
	  ........

	  Merged revisions 430145 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430146 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-24 15:28 +0000 [91becf952a]  Kevin Harwell <kharwell@digium.com>

	* app_queue: Update sample conf documenation

	  Updated the queues.conf.sample file to explicitly state which channel queue
	  variables are propagated to.

	  ASTERISK-24267
	  Reported by: Mitch Claborn
	  ........

	  Merged revisions 430126 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430127 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430128 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-24 10:59 +0000 [3a73c6c90e]  Matt Jordan <mjordan@digium.com>

	* main/pbx.c: Fix double lock of contexts lock introduced by r429967

	  We only need to hold the context_merge_lock once. Locking it twice will make
	  many other parts of Asterisk very sad.

	  ASTERISK-24641 #close


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430111 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-23 17:19 +0000 [7ea4156a5e]  gtjoseph <george.joseph@fairview5.com>

	* pjsip_options: Fix continued qualifies after endpoint/aor deletion

	  If you remove an endpoint/aor from pjsip.conf then do a core reload,
	  qualifies will continue even though the object are gone.  This happens
	  because nothing clears out the qualify tasks.

	  This patch unschedules all existing qualify tasks before scheduling
	  new ones on reload.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4290/
	  ........

	  Merged revisions 430064 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430067 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-23 17:16 +0000 [62d1dba271]  gtjoseph <george.joseph@fairview5.com>

	* test_astobj2: Fix warning for missing trailing slash in category

	  This patch adds a trailing slash to the category for this test.
	  No more warning.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4295/
	  ........

	  Merged revisions 430059 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-22 15:20 +0000 [1c0604e905]  Richard Mudgett <rmudgett@digium.com>

	* DTMF atxfer: Setup recall channels as if the transferee initiated the call.

	  After the initial DTMF atxfer call attempt to the transfer target fails to
	  answer during a blonde transfer, the recall callback channels do not get
	  setup with information from the initial transferrer channel.  As a result,
	  the recall callback to the transferrer does not have callid, channel
	  variables, datastores, accountcode, peeraccount, COLP, and CLID setup.  A
	  similar situation happens with the recall callback to the transfer target
	  but it is less visible.  The recall callback to the transfer target does
	  not have callid, channel variables, datastores, accountcode, peeraccount,
	  and COLP setup.

	  * Added missing information to the recall callback channels before
	  initiating the call.  callid, channel variables, datastores, accountcode,
	  peeraccount, COLP, and CLID

	  * Set callid of the transferrer channel on the DTMF atxfer controller
	  thread attended_transfer_monitor_thread().

	  * Added missing channel unlocks and props unref to off nominal paths in
	  attended_transfer_properties_alloc().

	  ASTERISK-23841 #close
	  Reported by: Richard Mudgett

	  Review: https://reviewboard.asterisk.org/r/4259/
	  ........

	  Merged revisions 430034 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430041 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-22 14:25 +0000 [7d954f4cb1]  Richard Mudgett <rmudgett@digium.com>

	* Fix compilation since the patch for ASTERISK-24363 went in.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430028 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-22 14:08 +0000 [bbd9ff122e]  Richard Mudgett <rmudgett@digium.com>

	* queue_log: Post QUEUESTART entry when Asterisk fully boots.

	  The QUEUESTART log entry has historically acted like a fully booted event
	  for the queue_log file.  When the QUEUESTART entry was posted to the log
	  was broken by the change made by ASTERISK-15863.

	  * Made post the QUEUESTART queue_log entry when Asterisk fully boots.
	  This restores the intent of that log entry and happens after realtime has
	  had a chance to load.

	  AST-1444 #close
	  Reported by: Denis Martinez

	  Review: https://reviewboard.asterisk.org/r/4282/
	  ........

	  Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 430010 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-22 09:40 +0000 [264a50c52a]  Karsten Wemheuer (License 5930)

	* chan_sip: Send CANCEL via original INVITE destination even after UPDATE request

	  Given the following scenario:
	  * Three SIP phones (A, B, C), all communicating via a proxy with Asterisk
	  * A call is established between A and B. B performs a SIP attended transfer of
	    A to C. B sets the call on hold (A is hearing MOH) and dials the extension of
	    C. While phone C is ringing, B transfers the call (that is, what we typically
	    call a 'blond transfer').
	  * When the transfer completes, A hears the ringing of phone C, while B is idle.

	  In the SIP messaging for the above scenario, a REFER request is sent to
	  transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an
	  UPDATE request to phone C to update party information. This update is sent
	  directly to phone C, not through the intervening proxy. This has the unfortunate
	  side effect of providing route information, which is then set on the sip_pvt
	  structure for C. If someone (e.g. B) is trying to get the call back (through a
	  directed pickup), Asterisk will send a CANCEL request to C. However, since we
	  have now updated the route set, the CANCEL request will be sent directly to C
	  and not through the proxy. The phone ignores this CANCEL according to RFC3261
	  (Section 9.1).

	  This patch updates reqprep such that the route is not updated if an UPDATE
	  request is being sent while the INVITE state is INV_PROCEEDING or
	  INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent
	  to the correct location.

	  Review: https://reviewboard.asterisk.org/r/4279

	  ASTERISK-24628 #close
	  Reported by: Karsten Wemheuer
	  patches:
	    issue.patch uploaded by Karsten Wemheuer (License 5930)
	  ........

	  Merged revisions 429982 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 429983 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429984 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-22 08:33 +0000 [0c38276d6e]  Gareth Palmer (License 5169)

	* presencestate: Allow channel drivers to provide presence state information

	  This patch adds the ability for channel drivers to supply presence information
	  in a similar manner to device state. The patch does not provide any channel
	  driver implementations, but it does provide the core infrastructure necessary
	  for channel drivers to provide such information.

	  The core handles multiple providers of presence state information. Ordering
	  of presence state is as follows:
	   INVALID < NOT_SET < AVAILABLE < UNAVAILABLE < CHAT < AWAY < XA < DND

	  Each provider can trump the previous if it provides a presence state that
	  supercedes a previous one.

	  Review: https://reviewboard.asterisk.org/r/4050

	  ASTERISK-24363 #close
	  Reported by: Gareth Palmer
	  patches:
	    chan_presencestate-428146.patch uploaded by Gareth Palmer (License 5169)


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429967 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-22 06:16 +0000 [2afeadcc84]  Matt Jordan <mjordan@digium.com>

	* app_confbridge: Fix build error caused by XML validation errors

	  Summaries can't contain XML nodes, as they are defined to contain only text
	  data.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-21 20:35 +0000 [b79a4a464f]  Gareth Palmer (License 5169)

	* app_confbridge: Add the ability to pass options/command to MixMonitor

	  This patch adds the ability to pass options and a command to MixMontor when
	  recording a conference using ConfBridge.

	  New options are -

	  * record_options: Options to MixMontor, eg: m(), W() etc.
	  * record_command: The command to execute when recording is over.
	  * record_file_timestamp: Append the start time to the file name.

	  These options can also be used with the CONFBRIDGE function, e.g.,
	  Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))

	  Review: https://reviewboard.asterisk.org/r/4023

	  ASTERISK-24351 #close
	  Reported by: Gareth Palmer
	  patches:
	    record_command-428838.patch uploaded by Gareth Palmer (License 5169)



	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-21 18:17 +0000 [b137a92aef]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_phoneprovi_provider: Fix reload

	  Reloading wasn't working correctly because on a reload, the sorcery apply
	  handler was never being called for unchanged users.  So, instead of using
	  an apply handler, I'm now iterating over all users.  Works much more reliably.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4288/
	  ........

	  Merged revisions 429914 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429915 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-20 14:57 +0000 [ba403e83bd]  Joshua Colp <jcolp@digium.com>

	* acl: Fix reloading of configuration if configuration file does not exist at startup.

	  The named ACL code incorrectly destroyed the config options information if loading
	  of the configuration file failed at startup. This would result in reloading
	  also failing even if a valid configuration file was put in place.

	  ASTERISK-23733 #close
	  Reported by: Richard Kenner
	  ........

	  Merged revisions 429893 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2014-12-19 14:56 +0000 [54bd1c9683]  Richard Mudgett <rmudgett@digium.com>

	* res_http_websocket.c: Fix incorrect use of sizeof in ast_websocket_write().

	  This won't fix the reported issue but it is an incorrect use of sizeof.

	  ASTERISK-24566
	  Reported by:  Badalian Vyacheslav
	  ........

	  Merged revisions 429867 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2014-12-19 11:34 +0000 [b508b3474e]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi: Don't ignore setvar when using configuration section scheme.

	  When the configuration section scheme of chan_dahdi.conf is used (keyword
	  dahdichan instead of channel) all setvar= options are completely ignored.
	  No variable defined this way appears in the created DAHDI channels.

	  * Move the clearing of setvar values to after the deferred processing of
	  dahdichan.

	  AST-1378 #close
	  Reported by: Guenther Kelleter
	  Patch by: Guenther Kelleter
	  ........

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	  ........

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2014-12-19 11:27 +0000 [07d1012383]  Scott Griepentrog <sgriepentrog@digium.com>

	* bridge: avoid leaking channel during blond transfer

	  After a blond transfer (start attended and hang up)
	  to a destination that also hangs up without answer,
	  the Local;1 channel was leaked and would show up on
	  core show channels.  This was happening because the
	  attended state blond_nonfinal_enter() resetting the
	  props->transfer_target to null while releasing it's
	  own reference, which would later prevent props from
	  releasing another reference during destruction. The
	  change made here is simply to not assign the target
	  to NULL.

	  ASTERISK-24513 #close
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4262/
	  ........

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2014-12-18 16:40 +0000 [2cbfafa8c1]  Richard Mudgett <rmudgett@digium.com>

	* chan_dahdi.c, res_rtp_asterisk.c: Change some spammy debug messages to level 5.

	  ASTERISK-24337 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 429804 from http://svn.asterisk.org/svn/asterisk/branches/11
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2014-12-18 14:09 +0000 [eacbb4ceb5]  Richard Mudgett

	* chan_dahdi: Populate CALLERID(ani2) for incoming calls in featdmf signaling mode.

	  For the featdmf signaling mode the incoming MF Caller-ID information is
	  formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}#

	  Rather than discarding the ani2 digits, populate the CALLERID(ani2) value
	  with what is received instead.

	  AST-1368 #close
	  Reported by: Denis Martinez
	  Patches:
	        extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett
	  ........

	  Merged revisions 429783 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2014-12-18 09:55 +0000 [546a54574f]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_sdp_rtp: wrong bridge chosen when the DTMF mode is not compatible

	  A native rtp bridge was being chosen (it shouldn't have been) when using two
	  pjsip channels with incompatible DTMF modes.  This patch sets the rtp instance
	  property, AST_RTP_PROPERTY_DTMF, for the appropriate DTMF mode(s) for pjsip.
	  It was not being set before, meaning all DTMF modes for pjsip were being treated
	  as compatible, thus native bridging would be chosen as the bridge type when it
	  shouldn't have been.

	  ASTERISK-24459 #close
	  Reported by: Yaniv Simhi
	  Review: https://reviewboard.asterisk.org/r/4265/
	  ........

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2014-12-18 09:40 +0000 [2f3e5b494a]  Mark Michelson <mmichelson@digium.com>

	* Prevent potential infinite outbound authentication loops in registration.

	  Prior to this patch, Asterisk would always respond to 401 responses to
	  registration attempts by trying to provide a registration with authentication
	  credentials. Even if subsequent attempts were rejected with 401 responses,
	  Asterisk would continue this behavior. If authentication credentials were
	  incorrect, this could continue forever.

	  With this patch, we keep track of whether we have attempted authentication
	  on an outbound registration attempt. If we already have, we don not try
	  again until the next attempt. This prevents the infinite loop scenario.

	  Review: https://reviewboard.asterisk.org/r/4273
	  ........

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2014-12-18 09:18 +0000 [2b1f2b5c1f]  Mark Michelson <mmichelson@digium.com>

	* Prevent possible race condition on dual redirect of channels in the same bridge.

	  The AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent bridges from
	  prematurely acting on orphaned channels in bridges. The problem with the AMI
	  redirect action was that it was setting this flag on channels based on the presence
	  of a PBX, not whether the channel was in a bridge. Whether a channel has a PBX
	  is irrelevant, so the condition has been altered to check if the channel is in a
	  bridge.

	  ASTERISK-24536 #close
	  Reported by Niklas Larsson

	  Review: https://reviewboard.asterisk.org/r/4268
	  ........

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2014-12-18 08:50 +0000 [cc1405bd38]  Mark Michelson <mmichelson@digium.com>

	* Ensure the correct value is returned for CHANNEL(pjsip, secure)

	  Prior to this patch, we were using the PJSIP dialog's secure flag
	  to determine if a secure transport was being used. Unfortunately,
	  the dialog's secure flag was only set if a SIPS URI were in use,
	  as required by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested
	  in is not dialog security, but transport security. This code change
	  switches to a model where we use the dialog's target URI to determine
	  what transport would be used to communicate, and then check if that
	  transport is secure.

	  AST-1450 #close
	  Reported by John Bigelow

	  Review: https://reviewboard.asterisk.org/r/4277
	  ........

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2014-12-17 18:11 +0000 [18b5a336ef]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard: fix unload SEGV

	  If certain pjsip modules aren't loaded, the wizard causes a SEGV
	  when it unloads.  Added a check for the presense of the object
	  type wizard before trying to clean it up.

	  Tested-by: George Joseph
	  ........

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2014-12-17 17:06 +0000 [c4360796f7]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard: Change FILEUNCHANGED config_load2 flag determination

	  The module now applies the FILEUNCHANGED flag when both reloaded is
	  specified AND there's no last_config for the object type.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4276/
	  ........

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2014-12-17 04:23 +0000 [8b6ecc449c]  Walter Doekes <walter+asterisk@wjd.nu>

	* Fix printf problems with high ascii characters after r413586 (1.8).

	  In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
	  Those fixes included things like:

	      -out += sprintf(out, "%%%02X", (unsigned char) *ptr);
	      +out += sprintf(out, "%%%02X", (unsigned) *ptr);

	  That works for low ascii characters, but for the high range that yields
	  e.g. FFFFFFC3 when C3 is expected.

	  This changeset:
	  - fixes those casts to use the 'hh' unsigned char modifier instead
	  - consistently uses %02x instead of %2.2x (or other non-standard usage)
	  - adds a few 'h' modifiers in various places
	  - fixes a 'replcaes' typo
	  - dev/urandon typo (in 13+ patch)

	  Review: https://reviewboard.asterisk.org/r/4263/

	  ASTERISK-24619 #close
	  Reported by: Stefan27 (on IRC)
	  ........

	  Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11
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	  ........

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2014-12-16 11:53 +0000 [c4cc668ba9]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard: fix test breakage

	  Fix test breakage caused by not checking for res_pjsip before
	  calling ast_sip_get_sorcery.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4269/
	  ........

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2014-12-16 10:39 +0000 [58095d2486]  Andreas Steinmetz (license 6523)

	* chan_sip: Allow T.38 switch-over when SRTP is in use.

	  Previously when SRTP was enabled on a channel it was not possible
	  to switch to T.38 as no crypto attributes would be present.

	  This change makes it so it is now possible. If a T.38 re-invite
	  comes in SRTP is terminated since in practice you can't encrypt
	  a UDPTL stream. Now... if we were doing T.38 over RTP (which
	  does exist) then we'd have a chance but almost nobody does that so
	  here we are.

	  ASTERISK-24449 #close
	  Reported by: Andreas Steinmetz
	  patches:
	   udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523)
	  ........

	  Merged revisions 429632 from http://svn.asterisk.org/svn/asterisk/branches/11
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2014-12-16 09:44 +0000 [b5182a6795]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_t38: Fix T.38 failure when peer reinvites immediately.

	  If a remote endpoint reinvites to T.38 immediately the state machine
	  will go into a peer reinvite state. If a T.38 capable application
	  (such as ReceiveFax) queries it will receive this state. Normally
	  the application will then indicate so that the channel driver will
	  queue up the T.38 offer previously received. Once it receives this
	  offer the application will act normally and negotiate.

	  The res_pjsip_t38 module incorrectly partially squashed this indication.
	  This would cause the application to think the request had failed when
	  in reality it had actually worked.

	  This change makes it so that no T.38 control frames (or indications)
	  are squashed.
	  ........

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2014-12-15 11:08 +0000 [39b54a21dc]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_config_wizard: Allow streamlined config of common pjsip scenarios

	  res_pjsip_config_wizard
	  ------------------
	   * This is a new module that adds streamlined configuration capability for
	     chan_pjsip.  It's targetted at users who have lots of basic configuration
	     scenarios like 'phone' or 'agent' or 'trunk'.  Additional information
	     can be found in the sample configuration file at
	     config/samples/pjsip_wizard.conf.sample.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4190/
	  ........

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2014-12-15 09:48 +0000 [53e5b377a0]  Mark Michelson <mmichelson@digium.com>

	* Activate persistent subscriptions when they are recreated.

	  Prior to this change, recreating persistent subscriptions would
	  create the subscription but would not activate it. This led to subscriptions
	  being listed in the "NULL" state by diagnostics and not sending NOTIFYs
	  when expected.

	  Review: https://reviewboard.asterisk.org/r/4261
	  ........

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2014-12-12 17:57 +0000 [6472568bc6]  gtjoseph <george.joseph@fairview5.com>

	* loader: Move definition of ast_module_reload from _private.h to module.h

	  No functionality change.  Just move the definition of ast_module_reload
	  from _private.h to module.h so it can be public.

	  Also removed the include of _private.h from manager.c since ast_module_load
	  was the only reason for including it.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4251/
	  ........

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2014-12-12 17:49 +0000 [308c1b41dd]  Richard Mudgett <rmudgett@digium.com>

	* DEBUG_THREADS: Fix regression and lock tracking initialization problems.

	  This patch started with David Lee's patch at
	  https://reviewboard.asterisk.org/r/2826/ and includes a regression fix
	  introduced by the ASTERISK-22455 patch.

	  The initialization of a mutex's lock tracking structure was not protected
	  in a critical section.  This is fine for any mutex that is explicitly
	  initialized, but a static mutex may have its lock tracking double
	  initialized if multiple threads attempt the first lock simultaneously.

	  * Added a global mutex to properly serialize initialization of the lock
	  tracking structure.  The painful global lock can be mitigated by adding a
	  double checked lock flag as discussed on the original review request.

	  * Defer lock tracking initialization until first use.

	  * Don't be "helpful" and initialize an uninitialized lock when
	  DEBUG_THREADS is enabled.  Debug code is not supposed to fix or change
	  normal code behavior.  We don't need a lock initialization race that would
	  force a re-setup of lock tracking.  Lock tracking already handles
	  initialization on first use.

	  * Properly handle allocation failures of the lock tracking structure.

	  * No need to initialize tracking data in __ast_pthread_mutex_destroy()
	  just to turn around and destroy it.


	  The regression introduced by ASTERISK-22455 is the result of manipulating
	  a pthread_mutex_t struct outside of the pthread library code.  The
	  pthread_mutex_t struct seems to have a global linked list pointer member
	  that can get changed by other threads.  Therefore, saving and restoring
	  the contents of a pthread_mutex_t struct is a bad thing.

	  Thanks to Thomas Airmont for finding this obscure regression.

	  * Don't overwrite the struct ast_lock_track.reentr_mutex member to restore
	  tracking data in __ast_cond_wait() and __ast_cond_timedwait().  The
	  pthread_mutex_t struct must be treated as a read-only opaque variable.


	  Miscellaneous other items fixed by this patch:

	  * Match ast_suspend_lock_info() with ast_restore_lock_info() in
	  __ast_cond_timedwait().

	  * Made some uninitialized lock sanity checks return EINVAL and try a
	  DO_THREAD_CRASH.

	  * Fix bad canlog initialization expressions.

	  ASTERISK-24614 #close
	  Reported by: Thomas Airmont

	  Review: https://reviewboard.asterisk.org/r/4247/
	  Review: https://reviewboard.asterisk.org/r/2826/
	  ........

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2014-12-12 16:54 +0000 [901221ffae]  Matt Jordan <mjordan@digium.com>

	* res/res_agi: Make Verbose message for 'stream file' match other playbacks

	  The Verbose message displayed when a file is played back via 'stream file'
	  was formatted differently than other playbacks:
	  * It didn't include the channel name
	  * It didn't include the channel language
	  It does, however, include the playback offset as well as any escape digits.
	  That information was kept; however, this patch updates the formatting to more
	  closely match the Verbose messages displayed when a file is played back by
	  'control stream file', Playback, ControlPlayback, or any other file playback
	  operation.
	  ........

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2014-12-12 11:01 +0000 [8d325be503]  Joshua Colp <jcolp@digium.com>

	* media: Fix crash when determining sample count of a frame during shutdown.

	  When shutting down Asterisk the codecs are cleaned up. As a result anything
	  attempting to get a codec based on ID or details will find that no codec
	  exists. This currently occurs when determining the sample count of a frame.
	  This code did not take this situation into account.

	  This change fixes this by getting the codec directly from the format and
	  eliminates the lookup. This is both faster and also provides a guarantee
	  that the codec will exist and will be valid.

	  ASTERISK-24604 #close
	  Reported by: Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4260/
	  ........

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2014-12-12 09:31 +0000 [72499dc697]  Kevin Harwell <kharwell@digium.com>

	* chan_pjsip: Race between channel answer and bridge setup when using direct media

	  When direct media is enabled and a pjsip channel is answered a race would occur
	  between the handling of the answer and bridge setup. Sometimes the media
	  negotiation would take place after the native bridge was setup. This resulted
	  in a NULL media address, which in turn resulted in Asterisk using its address
	  as the remote media address when sending a reinvite.  This patch makes the
	  chan_pjsip answer handler synchronous thus alleviating the race condition (the
	  bridge won't start setting things up until after it returns).

	  ASTERISK-24563 #close
	  Reported by: Steve Pitts
	  Review: https://reviewboard.asterisk.org/r/4257/
	  ........

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2014-12-12 09:03 +0000 [2e6d2b1484]  David M. Lee <dlee@digium.com>

	* Fix crash for sorcery misconfigs

	  res_pjsip_outbound_publish was missing the CHECK_PJSIP_MODULE_LOADED()
	  call in load_module, and would crash with a segfault if res_pjsip
	  declined to load.

	  Review: https://reviewboard.asterisk.org/r/4258/
	  ........

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2014-12-12 08:12 +0000 [a6cf13f2e9]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Allow use of 'inactive' streams for hold

	  This allows use of the 'inactive' stream direction identifier to be
	  used for hold where 'sendonly' is normally used. Some Seimens phones
	  use 'inactive' and this change allows music on hold to operate
	  properly.

	  Review: https://reviewboard.asterisk.org/r/4252/
	  Reported by: Steve Pitts
	  ........

	  Merged revisions 429432 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-12-12 08:04 +0000 [b99770d4fe]  Kinsey Moore <kmoore@digium.com>

	* Sorcery: Log when old config remains in use

	  This adds a log message notifying the user that a stale configuration
	  is in place upon reload when a config object fails to load. This
	  situation can end up causing confusion when the object failed to load
	  but exists from a previous config load especially when the old config
	  is significantly different from the new config.

	  Review: https://reviewboard.asterisk.org/r/4250/
	  Reported by: Thomas Thompson
	  ........

	  Merged revisions 429429 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-12-12 07:06 +0000 [74d43977cf]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.

	  Given the scenario where a PJSIP channel is in a native RTP bridge with direct
	  media and the channel is then hung up the code will currently re-INVITE the channel
	  back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
	  this greatly.

	  This change makes it so that if a re-INVITE transaction is in progress the BYE
	  is queued to occur after the completion of the transaction (be it through normal
	  means or a timeout).

	  Review: https://reviewboard.asterisk.org/r/4248/
	  ........

	  Merged revisions 429409 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429410 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-12 06:32 +0000 [8d384f3825]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_session: Fix issue where a declined media stream in a re-INVITE would fail SDP negotiation.

	  In the past the SDP negotiation within res_pjsip_session was made more tolerant of
	  certain situations. The only case where SDP negotiation will fail is when a major
	  error occurs during negotiation. Receiving an already declined media stream is
	  not considered a major error.

	  When producing the local SDP the logic took this into account so on the initial INVITE
	  the declined media stream did not cause an SDP negotiation failure. Unfortunately
	  the logic for handling media streams with a handler did not mirror this logic and
	  considered an already declined media stream an error and thus failed the SDP
	  negotiation.

	  This change makes the logic between both situations match so only under major
	  errors will the SDP negotiation fail.

	  ASTERISK-24607 #close
	  Reported by: Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4254/
	  ........

	  Merged revisions 429407 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429408 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-11 14:32 +0000 [63d3f0af95]  Kevin Harwell <kharwell@digium.com>

	* ARI/AMI: Include language in standard channel snapshot output

	  The CHANGES verbiage for the "language" addition had been put under the wrong
	  release. This moves it to be under 13.1 to 13.2 changes.

	  ASTERISK-24553
	  Reported by: Matt Jordan
	  ........

	  Merged revisions 429387 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429388 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-11 07:53 +0000 [d64b9904fd]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Update unittest for channel snapshots

	  This adjusts the unit test for channel snapshots to take the new
	  language key into account.
	  ........

	  Merged revisions 429352 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429353 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-10 09:43 +0000 [e890f9f653]  Kevin Harwell <kharwell@digium.com>

	* ARI/AMI: Include language in standard channel snapshot output

	  Adding information about including "language" in the standard channel snapshot
	  output to the CHANGES file. Note the actual source changes have already been
	  previously committed.

	  ASTERISK-24553
	  Reported by: Matt Jordan
	  ........

	  Merged revisions 429325 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429326 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-12-10 07:35 +0000 [03c94ef761]  Joshua Colp <jcolp@digium.com>

	* res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero.

	  Frames with a payload length of 0 were incorrectly handled in res_http_websocket.
	  Provided a frame with a payload had been received prior it was possible for a double
	  free to occur. The realloc operation would succeed (thus freeing the payload) but be
	  treated as an error. When the session was then torn down the payload would be
	  freed again causing a crash. The read function now takes this into account.

	  This change also fixes assumptions made by users of res_http_websocket. There is no
	  guarantee that a frame received from it will be NULL terminated.

	  ASTERISK-24472 #close
	  Reported by: Badalian Vyacheslav

	  Review: https://reviewboard.asterisk.org/r/4220/
	  Review: https://reviewboard.asterisk.org/r/4219/
	  ........

	  Merged revisions 429270 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 429272 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429273 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-12-10 07:16 +0000 [0cba439c4d]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Fix assert on initial mass qualify

	  This fixes the MWI test regressions caused by r429127 and ensures that
	  contacts have non-zero qualify_frequency before attempting scheduling.
	  ........

	  Merged revisions 429245 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429246 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-12-09 14:47 +0000 [8fe45f0f0a]  Scott Griepentrog <sgriepentrog@digium.com>

	* core: avoid possible asterisk -r crash from long id

	  When connecting to the remote console, an id string
	  is first provided that consts of the hostname, pid,
	  and version.  This is parsed by the remote instance
	  using a buffer that may be too short, and can allow
	  a buffer overrun because it is not terminated. This
	  patch adds termination and a larger buffer.

	  Review: https://reviewboard.asterisk.org/r/4182/
	  ........

	  Merged revisions 429223 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-09 14:20 +0000 [d673209abc]  Kevin Harwell <kharwell@digium.com>

	* ARI/AMI: Include language in standard channel snapshot output

	  The channel "language" was already part of a channel snapshot, however is was
	  not sent out over AMI or ARI. This patch makes it so the channel "language" is
	  included in the appropriate AMI or ARI events.

	  ASTERISK-24553 #close
	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/4245/
	  ........

	  Merged revisions 429204 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429206 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429209 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-09 14:03 +0000 [c17cef1c38]  Kevin Harwell <kharwell@digium.com>

	* Direct Media calls within private network sometimes get one way audio

	  When endpoints with direct_media enabled, behind a firewall (Asterisk on a
	  separate network) and were bridged sometimes Asterisk would send the ip
	  address of the firewall in the sdp to one of the phones in the reinvite
	  resulting in one way audio. When sending the reinvite Asterisk will retrieve
	  the media address from the associated rtp instance, but if frames were being
	  read this can be overwritten with another address (in this case the
	  firewall's).  This patch ensures that Asterisk uses the original device
	  address when using direct media.

	  ASTERISK-24563
	  Reported by: Steve Pitts
	  Review: https://reviewboard.asterisk.org/r/4216/
	  ........

	  Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429196 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-09 12:36 +0000 [7844266e21]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard

	  When using a non-default sorcery wizard (in this instance realtime) for outbound
	  publishes Asterisk will crash after a stack overflow occurs due to the code
	  infinitely recursing.  The fix entails removing the outbound publish state
	  dependency from the outbound publish sorcery object and instead keeping an in
	  memory container that can be used to lookup the state when needed.

	  ASTERISK-24514 #close
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4178/
	  ........

	  Merged revisions 429175 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429176 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-09 09:45 +0000 [60ab564ad2]  Joshua Colp <jcolp@digium.com>

	* ari: Add support for specifying an originator channel when originating.

	  If an originator channel is specified when originating a channel the linked ID
	  of it will be applied to the newly originated outgoing channel. This allows
	  an association to be made between the two so it is known that the originator
	  has dialed the originated channel.

	  ASTERISK-24552 #close
	  Reported by: Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4243/
	  ........

	  Merged revisions 429153 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429154 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-09 08:01 +0000 [b6e18cae5c]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Stagger outbound qualifies

	  This change staggers initiation of outbound qualify (OPTIONS) attempts
	  to reduce instantaneous server load and prevent network congestion.

	  Review: https://reviewboard.asterisk.org/r/4246/
	  ASTERISK-24342 #close
	  Reported by: Richard Mudgett
	  ........

	  Merged revisions 429127 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429128 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429129 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-08 10:54 +0000 [fe6cbf455a]  Matt Jordan <mjordan@digium.com>

	* AMI/ARI: Update version to 2.6.0/1.6.0 respectively for new features

	  AMI/ARI are getting a few enhancements in the next release of Asterisk 13. Per
	  semantic versioning, that warrants a bump in the minor version number, as it
	  reflects a backwards compatible change. Hence, this commit.
	  ........

	  Merged revisions 429091 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-08 10:43 +0000 [bba1763f47]  Mark Michelson <mmichelson@digium.com>

	* Fix a crash that would occur when receiving a 491 response to a reinvite.

	  The reviewboard description does a fine job of summarizing this, so here it is:

	  A reporter discovered that Asterisk would crash when attempting to retransmit
	  a reinvite that had previously received a 491 response. The crash occurred
	  because a pjsip_tx_data structure was being saved for reuse, but its reference
	  count was not being increased. The result was that the pjsip_tx_data was being
	  freed before we were actually done with it. When we attempted to re-use the
	  structure when re-sending the reinvite, Asterisk would crash.

	  The fix implemented here is not to try holding onto the pjsip_tx_data at all.
	  Instead, when we reschedule sending the reinvite, we create a brand new
	  pjsip_tx_data and send that instead. Because of this change, there is no need
	  for an ast_sip_session_delayed_request structure to have a pjsip_tx_data on
	  it any more. So any code referencing its use has been removed.

	  When this initial fix was introduced, I encountered a second crash when
	  processing a subsequent 200 OK on a rescheduled reinvite. The reason was
	  that when rescheduling the reinvite, we gave the wrong location for a
	  response callback. This has been fixed in this patch as well.

	  ASTERISK-24556 #close
	  Reported by Abhay Gupta

	  Review: https://reviewboard.asterisk.org/r/4233
	  ........

	  Merged revisions 429089 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-08 10:24 +0000 [fe7671fee6]  Mark Michelson <mmichelson@digium.com>

	* Add new AMI and ARI events for connected line changes on a channel.

	  The AMI event is called NewConnectedLine and the ARI event is called
	  ChannelConnectedLine.

	  ASTERISK-24554 #close
	  Reported by Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/4231
	  ........

	  Merged revisions 429064 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429084 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-08 09:45 +0000 [4bb556a847]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Fix StasisStart/End order and missing events

	  This corrects several bugs that currently exist in the stasis
	  application code.

	  * After a masquerade, the resulting channels have channel topics that
	    do not match their uniqueids
	  ** Masquerades now swap channel topics appropriately
	  * StasisStart and StasisEnd messages are leaked to observer
	    applications due to being published on channel topics
	  ** StasisStart and StasisEnd publishing is now properly restricted
	     to controlling apps via app topics
	  * Race conditions exist where StasisStart and StasisEnd messages due to
	    a masquerade may be received out of order due to being published on
	    different topics
	  ** These messages are now published directly on the app topic so this
	     is now a non-issue
	  * StasisEnds are sometimes missing when sent due to masquerades and
	    bridge swaps into and out of Stasis()
	  ** This was due to StasisEnd processing adjusting message-sent flags
	     after Stasis() had already exited and Stasis() had been re-entered
	  ** This was corrected by adjusting these flags prior to sending the
	     message while the initial Stasis() application was still shutting
	     down

	  Review: https://reviewboard.asterisk.org/r/4213/
	  ASTERISK-24537 #close
	  Reported by: Matt DiMeo
	  ........

	  Merged revisions 429061 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429062 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-06 12:16 +0000 [49aa87e17c]  Nuno Borges (License 6116)

	* res/res_monitor: Reset in/out sample counts on Monitor start

	  When repeatedly starting/stopping a Monitor on a channel, the accumulated
	  in/out sample counts are never reset to 0. This can cause inadvertent jumps
	  in the recordings, as the code in the channel core will determine incorrectly
	  that a jump in the recorded file position should occur. Setting the sample
	  counts to 0 simply reflects the initial state a Monitor should be in when it
	  is started, as this is the initial count that would be on the channels at that
	  time.

	  ASTERISK-24573 #close
	  Reported by: Nuno Borges
	  patches:
	    24573.patch uploaded by Nuno Borges (License 6116)
	  ........

	  Merged revisions 429031 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 429032 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429033 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-06 11:36 +0000 [0cdb71aae9]  Nuno Borges (License 6116)

	* apps/app_meetme: Apply default values on initial load with no config file

	  When the app_meetme module is loaded without its configuration file, the
	  module settings aren't initialized. In particular, this impacts the use
	  of logging realtime members. This patch guarantees that we always set the
	  default module settings on initial load.

	  Review: https://reviewboard.asterisk.org/r/4242/

	  ASTERISK-24572 #close
	  Reported by: Nuno Borges
	  patches:
	    24572.patch uploaded by Nuno Borges (License 6116)
	  ........

	  Merged revisions 429027 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 429028 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429029 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-05 11:08 +0000 [d04445c24a]  gtjoseph <george.joseph@fairview5.com>

	* sorcery: Add additional observer capabilities.

	  Add new global, instance and wizard observers.
	  instance_created
	  wizard_registered
	  wizard_unregistered
	  instance_destroying
	  instance_loading
	  instance_loaded
	  wizard_mapped
	  object_type_registered
	  object_type_loading
	  object_type_loaded
	  wizard_loading
	  wizard_loaded

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4215/
	  ........

	  Merged revisions 428999 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 429000 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429001 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-04 11:13 +0000 [19992844be]  Matt Jordan <mjordan@digium.com>

	* main/test: Fix compilation issue on 32-bit systems

	  On a 32-bit system, a type of intmax_t will result in a compilation warning
	  when formatted as a 'long int'. Use the format specifier of %jd (which was
	  what was used originally in manager.c) to format the JSON extracted integer
	  on both 32-/64-bit systems.
	  ........

	  Merged revisions 428972 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428973 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-04 09:48 +0000 [343a83d7d8]  Matt Jordan <mjordan@digium.com>

	* main/test: Fix race condition between AMI topic and Test Suite topic

	  This patch fixes a race condition between the raising of test AMI events (which
	  drive many tests in the Asterisk Test Suite) and other AMI events. Prior to
	  this patch, the Stasis messages published to the test topic were not forwarded
	  to the AMI topic. Instead, the code in manager had a dedicated handler for test
	  messages that was independent of the topics forwarded to the AMI topic. This
	  results in no synchronization between the test messages and the rest of the
	  Stasis messages published out over AMI. In some test with very tight timing
	  constraints, this can result in out of order messages and spurious test
	  failures. Properly forwarding the Test Suite topic to the AMI topic ensures
	  that the messages are synchronized properly.

	  This patch does that, and moves the message handling to the Stasis definition
	  of the Test Suite message in test.c as well.

	  Review: https://reviewboard.asterisk.org/r/4221/
	  ........

	  Merged revisions 428945 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428946 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-03 14:59 +0000 [7cb2c446b4]  Matt Jordan <mjordan@digium.com>

	* tests/test_cel: Add test_cel_attended_transfer_bridges_link to racey tests

	  Despite failing less often, the ordering of the ATTENDEDTRANSFER event and the
	  BRIDGE_EXIT event for the Alice and David channels is not defined. This makes
	  the test still fail.
	  ........

	  Merged revisions 428918 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428919 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-03 13:49 +0000 [7475e1c948]  Matt Jordan <mjordan@digium.com>

	* tests/test_cel: Fix CEL unit test failures caused by attended transfer changes

	  When the publication of attended transfer messages were pushed to another
	  thread, some subtle race conditions were introduced with the CEL unit tests.
	  This patch fixes one of them, and pushes the other to ASTERISK-22367, which
	  already exists to fix another bouncy CEL unit test.

	  In particular, this patch fixes the test_cel_attended_transfer_bridges_link
	  test, and defers the test_cel_attended_transfer_bridges_swap test to the
	  aforementioned JIRA issue.

	  ASTERISK-22367
	  ........

	  Merged revisions 428891 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-03 10:45 +0000 [6d4ef7ddf4]  David Duncan Ross Palmer (License 6660)

	* apps/app_voicemail: Fix crash with IMAP when streams are opened simultaneously

	  The UW IMAP library is instrinsically not thread-safe, and relies upon higher
	  level applications to guarantee thread safety. For the most part, this is
	  provided by the vms object, which provides locking for individual streams.
	  Unfortunately, this is not sufficient for calls to mail_open which create the
	  IMAP stream. mail_open can, on some systems, call into a UW IMAP specific
	  function for determining the address of a system based on a hostname,
	  ip_nametoaddr.

	  In the ip6_unix implementation of this function, static variables are used
	  to hold parsing buffers. This can cause a crash if multiple threads attempt
	  to convert a hostname to an address at the same time. Locking on a single
	  mail stream is not sufficient to prevent simultaneous access to these static
	  variables.

	  In the IMAP library, this function can be called from the mail_open and
	  imap_status functions. As the imap_status function is not used by
	  app_voicemail, locking on access to mail_open is sufficient to prevent
	  any mangling of the buffers.

	  Review: https://reviewboard.asterisk.org/r/4188/

	  ASTERISK-24516 #close
	  Reported by: David Duncan Ross Palmer
	  Tested by: David Duncan Ross Palmer
	  patches:
	    ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660)
	  ........

	  Merged revisions 428863 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428864 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428865 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-02 15:54 +0000 [63cbd28999]  gtjoseph <george.joseph@fairview5.com>

	* CHANGES: Add item for new 'pjsip show identif(y|ies) commands

	  Tested-by: George Joseph
	  ........

	  Merged revisions 428836 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428837 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-02 13:04 +0000 [dd00e80cbe]  Matt Jordan <mjordan@digium.com>

	* tests/test_stasis: Resolve compilation issues from Asterisk 12 merge

	  When merging the changes up stream in r428687, I missed the fact that the
	  signature for stasis_message_type_create was changed. This patch fixes
	  the compilation issues introduced by that merge.
	  ........

	  Merged revisions 428815 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-02 11:10 +0000 [08636aadec]  Birger Harzenetter (License 5870)

	* pbx/pbx_loopback: Speed up switches by avoiding unneeded lookups

	  This patch makes a small rearrangement to only do dialplan lookups during
	  loopback switches if the pattern matches. Prior to this patch, the dialplan
	  lookups were always performed, even when the result would be discarded.
	  Dialplan lookups can be very costly if remote switches - like DUNDi - are
	  present. In those cases extension matching is sped up considerably, making
	  the issue of lost digits more manageable.

	  As collateral damage, 6 trailing spaces were killed.

	  Review: https://reviewboard.asterisk.org/r/4211

	  ASTERISK-24577 #close
	  Reported by: Birger Harzenetter
	  patches:
	    ast-loopback.patch uploaded by Birger Harzenetter (License 5870)
	  ........

	  Merged revisions 428787 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428788 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428789 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-02 06:21 +0000 [0c1aaa7da5]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_refer: Fix issue where native bridge may not occur upon completion of a transfer.

	  There are two methods within res_pjsip_refer for keeping track of the state of a transfer.
	  The first is a framehook which looks at frames passing by to determine the state. The second
	  subscribes to know when the channel joins a bridge. In the case when the channel joins the
	  bridge the framehook is *NOT* removed and this prevents the native RTP bridging technology
	  from getting used.

	  This change gets the channel and if it still exists remove the framehook.

	  Review: https://reviewboard.asterisk.org/r/4218/
	  ........

	  Merged revisions 428760 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-01 18:38 +0000 [f128ff61ab]  gtjoseph <george.joseph@fairview5.com>

	* config: Create ast_variable_find_in_list()

	  Add
	  const char *ast_variable_find_in_list(const struct ast_variable *list,
	     const char *variable);

	  ast_variable_find() requires a config category to search whereas
	  ast_variable_find_in_list() just needs the root list element which is
	  useful if you don't have a category.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4217/
	  ........

	  Merged revisions 428733 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428734 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-01 18:31 +0000 [f418f25c44]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_endpoint_identifier_ip: Add 'show identify(ies)' cli commands

	  While troubleshooting other things I realized there were no pjsip cli
	  commands for identify.  This patch adds them.  It also also fixes a
	  reference leak when a 'show endpoint' displayed identifies and properly
	  sets the return code if load_module can't allocate a cli formatter structure.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4212/
	  ........

	  Merged revisions 428725 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428731 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428732 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-01 12:51 +0000 [4ff6bd831f]  Joshua Colp <jcolp@digium.com>

	* rtp_engine: Add support for transporting signed linear at 12kHz, 24kHz, 32kHz, 44kHz, 48kHz, 96kHz, and 192kHz over RTP.

	  This change adds mappings in the RTP engine layer for the remaining signed linear formats.

	  ASTERISK-24274 #close
	  Reported by: Frankie Chin

	  Review: https://reviewboard.asterisk.org/r/4093/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428708 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-01 11:59 +0000 [1106e8fd0f]  Matt Jordan <mjordan@digium.com>

	* main/stasis: Allow subscriptions to use a threadpool for message delivery

	  Prior to this patch, all Stasis subscriptions would receive a dedicated
	  thread for servicing published messages. In contrast, prior to r400178
	  (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
	  shared a thread pool. It was discovered during some initial work on Stasis
	  that, for a low subscription count with high message throughput, the
	  threadpool was not as performant as simply having a dedicated thread per
	  subscriber.

	  For situations where a subscriber receives a substantial number of messages
	  and is always present, the model of having a dedicated thread per subscriber
	  makes sense. While we still have plenty of subscriptions that would follow
	  this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
	  the following two categories:
	  * Large number of subscriptions, specifically those tied to endpoints/peers.
	  * Low number of messages. Some subscriptions exist specifically to coordinate
	    a single message - the subscription is created, a message is published, the
	    delivery is synchronized, and the subscription is destroyed.
	  In both of the latter two cases, creating a dedicated thread is wasteful (and
	  in the case of a large number of peers/endpoints, harmful). In those cases,
	  having shared delivery threads is far more performant.

	  This patch adds the ability of a subscriber to Stasis to choose whether or not
	  their messages are dispatched on a dedicated thread or on a threadpool. The
	  threadpool is configurable through stasis.conf.

	  Review: https://reviewboard.asterisk.org/r/4193

	  ASTERISK-24533 #close
	  Reported by: xrobau
	  Tested by: xrobau
	  ........

	  Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-12-01 07:41 +0000 [ef9ca8bc32]  Ben Smithurst (license 6529)

	* app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording.

	  The Record dialplan function trims 1/4 of a second from the end of recordings in case
	  they are terminated because of DTMF. When hanging up, however, you don't want this to happen.
	  This change makes it so on hangup this does not occur.

	  ASTERISK-24530 #close
	  Reported by: Ben Smithurst
	  patches:
	   app_record_v2.diff submitted by Ben Smithurst (license 6529)

	  Review: https://reviewboard.asterisk.org/r/4201/
	  ........

	  Merged revisions 428653 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-12-01 07:08 +0000 [7db3d1642b]  snuffy <snuffy22@gmail.com> (license 5024)

	* channel: Extend size of buffer for codecs in "core show channeltype" CLI command.

	  The static buffer for codecs when invoking the "core show channeltype" CLI command
	  did not have enough room for all codecs. This has been extended so it does.

	  ASTERISK-24542 #close
	  Reported by: snuffy
	  patches:
	   channeltype-tech.diff submitted by snuffy (license 5024)

	  Review: https://reviewboard.asterisk.org/r/4204/
	  ........

	  Merged revisions 428632 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-11-24 14:39 +0000 [3e08619faf]  Richard Mudgett <rmudgett@digium.com>

	* test_channel_feature_hooks.c: Fix unit test for DTMF hooks.

	  Fix the failing /channels/features/test_features_channel_dtmf unit test.

	  DTMF emulation does not work without a stream of packets to prod the
	  emulation code.

	  Review: https://reviewboard.asterisk.org/r/4199/
	  ........

	  Merged revisions 428604 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-24 14:32 +0000 [c38ffca9a1]  Richard Mudgett <rmudgett@digium.com>

	* DTMF hooks: Leaving channels need to push any collected digits into the bridge.

	  Any partially collected DTMF digits for a DTMF hook need to be pushed into
	  the bridge when a channel leaves the bridging system as if there were a
	  timeout.

	  Review: https://reviewboard.asterisk.org/r/4199/
	  ........

	  Merged revisions 428601 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-21 13:16 +0000 [3576ae47f4]  Richard Mudgett <rmudgett@digium.com>

	* manager: Fix could not extend string messages.

	  When shutting down Asterisk that has an active AMI connection, you get
	  several "failed to extend from %d to %d" messages because use of the
	  EVENT_FLAG_SHUTDOWN attempts to add all AMI permission strings to the
	  event.

	  * Created MAX_AUTH_PERM_STRING to use when creating stack based struct
	  ast_str variables used with the authority_to_str() and
	  user_authority_to_str() functions instead of a variety of magic numbers
	  that could be too small.

	  * Added a special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so
	  it will not attempt to add all permission level strings.

	  Review: https://reviewboard.asterisk.org/r/4200/
	  ........

	  Merged revisions 428570 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428571 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428573 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-21 11:49 +0000 [4394e0431c]  gtjoseph <george.joseph@fairview5.com>

	* sorcery: Make is_object_field_registered handle field names that are regexes.

	  As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime
	  was tossing database fields that didn't have an exact match to a sorcery
	  registered field.  This broke the ability to use regexes as field names which
	  manifested itself as a failure of res_pjsip_phoneprov_provider which uses
	  this capability.  It also broke handling of fields that start with '@' in
	  realtime but I don't think anyone noticed.

	  This patch does the following...
	  * Modifies ast_sorcery_fields_register to pre-compile the name regex.
	  * Modifies ast_sorcery_is_object_field_registered to test the regex if it
	    exists instead of doing an exact strcmp.
	  * Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work
	    with realtime.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4185/
	  ........

	  Merged revisions 428543 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428544 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428545 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-21 07:59 +0000 [d663e045f5]  Olle Johansson <oej@edvina.net>

	* sip.conf.sample - note that media_address does not change listen address, just the SDP


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-20 20:17 +0000 [2be984fb11]  Matt Jordan <mjordan@digium.com>

	* main/bridge_basic: Fix features regressions introduced by r428165

	  In r428165, two bugs were introduced:

	  * Prior to entering the features retry loop, the buffer that holds the
	    collected digits is wiped. However, this inadvertently wipes out the
	    first collected digit on the first pass through, which is obtained
	    in ast_stream_and_wait. This caused all of the features tests to fail.
	  * If ast_app_dtget returns a hangup (-1), the loop would retry incorrectly.
	    If we detect a hangup, we have to stop trying the feature.

	  This patch fixes both issues.

	  Review: https://reviewboard.asterisk.org/r/4196/
	  ........

	  Merged revisions 428505 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428506 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-20 10:37 +0000 [2f78fde10f]  Matt Jordan <mjordan@digium.com> (License #6283)

	* Fix error with mixed address family ACLs.

	  Prior to this commit, the address family of the first item in an ACL
	  was used to compare all incoming traffic. This could lead to traffic
	  of other IP address families bypassing ACLs.

	  ASTERISK-24469 #close

	  Reported by Matt Jordan
	  Patches:
	  	ASTERISK-24469-11.diff uploaded by Matt Jordan (License #6283)

	  AST-2014-012
	  ........

	  Merged revisions 428402 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 428417 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428422 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428426 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-20 10:35 +0000 [2486b48cec]  Gareth Palmer (license 5169)

	* AST-2014-018 - func_db: DB Dialplan function permission escalation via AMI.

	  The DB dialplan function when executed from an external protocol (for instance
	  AMI), could result in a privilege escalation.

	  Asterisk now inhibits the DB function from being executed from an external
	  interface if the live_dangerously option is set to no.

	  ASTERISK-24534
	  Reported by: Gareth Palmer
	  patches: submitted by Gareth Palmer (license 5169)
	  ........

	  Merged revisions 428331 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 428363 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428409 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428418 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-20 10:25 +0000 [2f97486d43]  Jonathan Rose <jrose@digium.com>

	* PJSIP ACLs: Fix ACLs not loading on startup and apply/acl issues on contact

	  The biggest problem this patch fixes is that ACLs weren't previously being
	  loaded when the res_pjsip_acl module was loaded. Yikes. In addition, the
	  ACL options contact_permit and contact_acl were effectively interpreted as
	  contact_deny and this patch fixes that as well.

	  AST-1418 #close
	  Reported by: Thomas Thompson
	  Review: https://reviewboard.asterisk.org/r/4120/

	  ASTERISK-24531 #close
	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/4171/
	  ........

	  Merged revisions 428333 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428376 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-20 09:57 +0000 [a389f2d7a0]  Kevin Harwell <kharwell@digium.com>

	* AST-2014-017 - app_confbridge: permission escalation/ class authorization.

	  Confbridge dialplan function permission escalation via AMI and inappropriate
	  class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan
	  function when executed from an external protocol (for instance AMI), could
	  result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord”
	  could also be used to execute arbitrary system commands without first checking
	  for system access. The AMI “ConfbridgeStopRecord” has also been updated to
	  only run under a system authorization.

	  Asterisk now inhibits the CONFBRIDGE function from being executed from an
	  external interface if the live_dangerously option is set to no.  Also, the
	  “ConfbridgeStartRecord” AMI action is now only allowed to execute under a
	  user with system level access.

	  ASTERISK-24490
	  Reported by: Gareth Palmer
	  ........

	  Merged revisions 428332 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428342 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-20 08:56 +0000 [1c88ca9d31]  Joshua Colp <jcolp@digium.com>

	* AST-2014-016: Fix crash when receiving an in-dialog INVITE with Replaces in res_pjsip_refer.

	  The implementation of INVITE with Replaces in res_pjsip_refer did not expect them to
	  occur in-dialog. As a result it would incorrectly attempt to hang up a channel it
	  thought was under its control. In reality the channel would be under the control of
	  another thread. When the other thread accessed the channel it would be accessing freed
	  memory and could crash.

	  This change makes res_pjsip_refer not act on an in-dialog INVITE with Replaces.

	  ASTERISK-24528 #close
	  Reported by: Joshua Colp
	  ........

	  Merged revisions 428304 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428306 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-20 08:49 +0000 [d25eda5fb2]  Joshua Colp <jcolp@digium.com>

	* AST-2014-015: Fix race condition in chan_pjsip when sending responses after a CANCEL has been received.

	  Due to the serialized architecture of chan_pjsip there exists a race condition where a CANCEL may
	  be received and processed before responses (such as 180 Ringing, 183 Session Progress, and 200 OK)
	  are sent. Since the session is in an unexpected state PJSIP will assert when this is attempted.

	  This change makes it so that these responses are not sent on disconnected sessions.

	  ASTERISK-24471 #close
	  Reported by: yaron nahum
	  ........

	  Merged revisions 428301 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428303 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-19 13:32 +0000 [57c6f89bf0]  Corey Farrell <git@cfware.com>

	* stringfields: Fix bug in ast_string_fields_copy.

	  ast_string_fields_copy relies on the fact that
	  __ast_string_field_release_active never previously
	  zeroed pool->used, so keeping the existing pointer
	  was "ok".  Now that existing pools can be reset to
	  'empty', it is important to set each field to
	  __ast_string_field_empty after releasing the memory.

	  ASTERISK-24535 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4186/
	  ........

	  Merged revisions 428272 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428274 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-19 11:22 +0000 [a7c9f4c668]  Richard Mudgett <rmudgett@digium.com>

	* ast_str: Fix improper member access to struct ast_str members.

	  Accessing members of struct ast_str outside of the string manipulation API
	  routines is invalid since struct ast_str is supposed to be treated as
	  opaque.

	  Review: https://reviewboard.asterisk.org/r/4194/
	  ........

	  Merged revisions 428244 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428245 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428255 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-19 06:50 +0000 [7f8b7ace72]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Add support for optimistic SRTP.

	  Optimistic SRTP is the ability to enable SRTP but not have it be
	  a fatal requirement. If SRTP can be used it will be, if not it won't be.
	  This gives you a better chance of using it without having your sessions
	  fail when it can't be.

	  Encrypt all the things!

	  Review: https://reviewboard.asterisk.org/r/3992/
	  ........

	  Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-19 06:45 +0000 [b2e766a6b7]  Joshua Colp <jcolp@digium.com>

	* alembic: Fix alembic migration for 'moh_passthrough' option in res_pjsip.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-19 05:51 +0000 [3119c3737f]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI.

	  There is no guarantee that when we get a Refer-To that it will be NULL terminated.
	  As the URI parsing function requires it to be we now NULL terminate it.

	  Additionally parsing the Refer-To as a 'To' header is needless and it can
	  simply be done as a URI. This also fixes a problem where certain Refer-To headers
	  would not be parsed as a 'To' header causing the REFER to fail.

	  ASTERISK-24508 #close
	  Reported by: Beppo Mazzucato

	  Review: https://reviewboard.asterisk.org/r/4187/
	  ........

	  Merged revisions 428195 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428196 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-11-18 13:12 +0000 [a94efa239c]  Richard Mudgett <rmudgett@digium.com>

	* parking_tests.c: Add missing newline on a unit test message.
	  ........

	  Merged revisions 428168 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428169 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-11-17 10:58 +0000 [2e750db120]  Mark Michelson <mmichelson@digium.com>

	* Allow for transferer to retry when dialing an invalid extension.

	  This allows for a configurable number of attempts for a transferer
	  to dial an extension to transfer the call to. For Asterisk 13, the
	  default values are such that upgrading between versions will not
	  cause a behaivour change. For trunk, though, the defaults will be
	  changed to be more user-friendly.

	  Review: https://reviewboard.asterisk.org/r/4167
	  ........

	  Merged revisions 428145 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-11-17 10:02 +0000 [4cea5fd4ba]  Corey Farrell <git@cfware.com>

	* chan_sip: Fix theoretical leak of p->refer.

	  If transmit_refer is called when p->refer is already allocated,
	  it leaks the previous allocation.  Updated code to always free
	  previous allocation during a new allocation.  Also instead of
	  checking if we have a previous allocation, always create a
	  clean record.

	  ASTERISK-15242 #close
	  Reported by: David Woolley
	  Review: https://reviewboard.asterisk.org/r/4160/
	  ........

	  Merged revisions 428117 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428118 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-17 09:27 +0000 [948af7fd79]  Matt Jordan <mjordan@digium.com>

	* apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves

	  When r428077 was made for ASTERISK-24522, it failed to take into account users
	  who are neither wait_marked nor end_marked. These users are *also* supposed to
	  hear the 'leader has left the conference' message. Granted, this behaviour is
	  a bit odd; however, that is how it used to work... and behaviour changes are
	  not good.

	  This patch ensures that if there are any 'normal' users present when the last
	  marked user leaves the conference, the message will still be played to them.

	  Note that this regression was caught by the Asterisk Test Suite's
	  confbridge_nominal test, which has a quirky combination of users.
	  ........

	  Merged revisions 428113 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428114 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-16 21:08 +0000 [fc2279afea]  Matt Jordan <mjordan@digium.com>

	* app_confbridge: Don't play leader leaving prompt if no one will hear it

	  Consider the following:
	  - A marked user in a conference
	  - One or more end_marked only users in the conference

	  When the marked users leaves, we will be in the conf_state_multi_marked state.
	  This currently will traverse the users, kicking out any who have the end_marked
	  flags. When they are kicked, a full ast_bridge_remove is immediately called on
	  the channels. At this time, we also unilaterally set the need_prompt flag.

	  When the need_prompt flag is set, we then playback a sound to the bridge
	  informing everyone that the leader has left; however, no one is left in the
	  bridge. This causes some odd behaviour for the end_marked users - they are
	  stuck waiting for the bridge to be unlocked. This results in them waiting for
	  5 or 6 seconds of dead air before hearing that they've been kicked.

	  Unfortunately, we do have to keep the bridge locked while we're playing back
	  the 'leader-has-left' prompt. If there are any wait_marked users in the
	  conference, this behaviour can't be easily changed - but we do make the case
	  of the end_marked users better with this patch.

	  Review: https://reviewboard.asterisk.org/r/4184/

	  ASTERISK-24522 #close
	  Reported by: Matt Jordan
	  ........

	  Merged revisions 428077 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 428078 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-16 15:13 +0000 [656601d8c4]  Joshua Colp <jcolp@digium.com>

	* chan_pjsip: Remove AOR check when dialing and one is specified.

	  The AOR value may contain the name of an AOR or a full SIP URI.
	  Checking if the AOR exists can't be done as a result of this.
	  ........

	  Merged revisions 428051 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 428052 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-11-16 06:12 +0000 [bc02cbabd9]  Joshua Colp <jcolp@digium.com>

	* chan_sip: Fix bug where DTLS configuration from general would copy dtlsenable.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-15 15:52 +0000 [6993743b1f]  Etienne Lessard (License 6394)

	* cel/cel_odbc: Provide microsecond precision in 'eventtime' column when possible

	  This patch adds microsecond precision when inserting a CEL record into a table
	  with an "eventtime" column of type timestamp, instead of second precision. The
	  documentation (configs/cel_odbc.conf.sample) was already saying that the
	  eventtime column included microseconds precision, but that was not the case.

	  Also, without this patch, if you had a table with an "eventtime" column of
	  type varchar, you had millisecond precision. With this patch, you also get
	  microsecond precision in this case.

	  Review: https://reviewboard.asterisk.org/r/3980

	  ASTERISK-24283 #close
	  Reported by: Etienne Lessard
	  patches:
	    cel_odbc_time_precision.patch uploaded by Etienne Lessard (License 6394)
	  ........

	  Merged revisions 427952 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  ........

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2014-11-15 15:36 +0000 [ece61f5ed1]  Joshua Colp <jcolp@digium.com>

	* chan_pjsip: Add additional log message when an AOR is specified when dialing and it does not exist.

	  ASTERISK-24499 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 428007 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-15 13:01 +0000 [49e63a191d]  Joshua Colp <jcolp@digium.com>

	* chan_motif / chan_pjsip: Fix incorrect "No such module" messages when reloading.

	  For chan_motif the direct return value of the underlying config options framework
	  was passed back. This can relay various states which the module loader would not
	  interpet as success. It has been changed so only on errors will it report back
	  an error.

	  For chan_pjsip the code implemented a dummy reload function which always
	  returned an error. This has been removed as all configuration is held within
	  res_pjsip instead.

	  ASTERISK-23651 #close
	  Reported by: Rusty Newton
	  ........

	  Merged revisions 427981 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-15 12:29 +0000 [9d2882d274]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Enforce requirements for session timer minimum expiration period and normal expiration period.

	  This change enforces the requirements in PJSIP for session timer configuration. The minimum
	  expiration period must be 90 seconds or higher and the normal expiration period can not
	  be lower than the minimum expiration period. If either of these were done the code would
	  assert at session setup time.

	  ASTERISK-24336 #close
	  Reported by: Leon Rowland
	  ........

	  Merged revisions 427978 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-15 10:31 +0000 [d0523b4b3c]  Michael K. (license 6621)

	* chan_sip: Add support for setting DTLS configuration in the general section.

	  Configuration of DTLS in the general section will be applied to any users
	  or peers. If configuration exists at their level it overrides the general
	  section values.

	  ASTERISK-24128 #close
	  Reported by: Michael K.
	  patches:
	    dtls_default_settings.patch submitted by Michael K. (license 6621)

	  Review: https://reviewboard.asterisk.org/r/3867/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-14 15:51 +0000 [3268544907]  Matt Jordan <mjordan@digium.com>

	* tests/test_cel: Unlock bridge on off nominal paths

	  If the test fails due to memory allocation errors, we may as well attempt to
	  unlock the bridge on the way out.
	  ........

	  Merged revisions 427927 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427932 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-14 12:12 +0000 [df2090b931]  Jonathan Rose <jrose@digium.com>

	* Documentation: Revise explanation of cdr.conf option 'Unanswered'

	  ASTERISK-24279 #close
	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/4109/
	  ........

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	  ........

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2014-11-14 09:52 +0000 [ba811ae1c3]  Scott Griepentrog <sgriepentrog@digium.com>

	* stun: correct attribute string padding to match rfc

	  When sending the USERNAME attribute in an RTP STUN
	  response, the implementation in append_attr_string
	  passed the actual length, instead of padding it up
	  to a multiple of four bytes as required by the RFC
	  3489.  This change adds separate variables for the
	  string and padded attributed lengths, and performs
	  padding correctly.

	  Reported by: Thomas Arimont
	  Review: https://reviewboard.asterisk.org/r/4139/
	  ........

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2014-11-14 09:28 +0000 [2d9471ab1f]  Mark Michelson <mmichelson@digium.com>

	* Fix race condition that could result in ARI transfer messages not being sent.

	  From reviewboard:

	  "During blind transfer testing, it was noticed that tests were failing
	  occasionally because the ARI blind transfer event was not being sent.
	  After investigating, I detected a race condition in the blind transfer
	  code. When blind transferring a single channel, the actual transfer
	  operation (i.e. removing the transferee from the bridge and directing
	  them to the proper dialplan location) is queued onto the transferee
	  bridge channel. After queuing the transfer operation, the blind transfer
	  Stasis message is published. At the time of publication, snapshots of
	  the channels and bridge involved are created. The ARI subscriber to the
	  blind transfer Stasis message then attempts to determine if the bridge
	  or any of the involved channels are subscribed to by ARI applications.
	  If so, then the blind transfer message is sent to the applications. The
	  way that the ARI blind transfer message handler works is to first see
	  if the transferer channel is subscribed to. If not, then iterate over
	  all the channel IDs in the bridge snapshot and determine if any of
	  those are subscribed to. In the test we were running, the lone
	  transferee channel was subscribed to, so an ARI event should have been
	  sent to our application. Occasionally, though, the bridge snapshot did
	  not have any channels IDs on it at all. Why?

	  The problem is that since the blind transfer operation is handled by a
	  separate thread, it is possible that the transfer will have completed and
	  the channels removed from the bridge before we publish the blind transfer
	  Stasis message. Since the blind transfer has completed, the bridge on
	  which the transfer occurred no longer has any channels on it, so the
	  resulting bridge snapshot has no channels on it. Through investigation of
	  the code, I found that attended transfers can have this issue too for the
	  case where a transferee is transferred to an application."

	  The fix employed here is to decouple the creation of snapshots for the transfer
	  messages from the publication of the transfer messages. This way, snapshots
	  can be created to reflect what they are at the time of the transfer operation.

	  Review: https://reviewboard.asterisk.org/r/4135
	  ........

	  Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-14 08:56 +0000 [737b811749]  Joshua Colp <jcolp@digium.com>

	* app_confbridge: Play "leader has left" sound even when musiconhold is enabled.

	  Currently if the leader of a conference bridge leaves any participant
	  that has musiconhold enabled will not hear the "leader has left" sound.
	  This is because musiconhold is started and THEN the sound is played.

	  This change makes it so that the sound is played and THEN musiconhold
	  is started. This provides a better experience for users as they may not
	  have known previously why they went back to musiconhold.

	  Review: https://reviewboard.asterisk.org/r/4177/
	  ........

	  Merged revisions 427844 from http://svn.asterisk.org/svn/asterisk/branches/11
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	  ........

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2014-11-14 08:40 +0000 [2454505d5a]  Mark Michelson <mmichelson@digium.com>

	* Fix race condition where duplicated requests may be handled by multiple threads.

	  This is the Asterisk 13 version of the patch. The main difference is in the pubsub
	  code since it was completely refactored between Asterisk 12 and 13.

	  Review: https://reviewboard.asterisk.org/r/4175
	  ........

	  Merged revisions 427841 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-11-13 16:26 +0000 [49b7a1cbaf]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash

	  When using a non-default sorcery wizard (in this instance realtime) for
	  outbound registrations and after adding in an appropriate call to
	  ast_sorcery_apply_config() (since it is missing) Asterisk will crash after
	  a stack overflow occurs due to the code infinitely recursing.  The fix entails
	  removing the outbound registration state dependency from the outbound
	  registration sorcery object and instead keeping an in memory container that
	  can be used to lookup the state when needed.

	  ASTERISK-24514
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4164/
	  ........

	  Merged revisions 427814 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-11-13 09:46 +0000 [74e706878b]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Fix StasisEnd message ordering

	  This change corrects message ordering in cases where a channel-related
	  message can be received after a Stasis/ARI application has received the
	  StasisEnd message. The StasisEnd message was being passed to
	  applications directly without waiting for the channel topic to empty.

	  As a result of this fix, other bugs were also identified and fixed:
	  * StasisStart messages were also being sent directly to apps and are
	    now routed through the stasis message bus properly
	  * Masquerade monitor datastores were being removed at the incorrect
	    time in some cases and were causing StasisEnd messages to not be sent
	  * General refactoring where necessary for the above
	  * Unsubscription on StasisEnd timing changes to prevent additional
	    messages from following the StasisEnd when they shouldn't

	  A channel sanitization function pointer was added to reduce processing
	  and AO2 lookups.

	  Review: https://reviewboard.asterisk.org/r/4163/
	  ASTERISK-24501 #close
	  Reported by: Matt Jordan
	  ........

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2014-11-12 18:23 +0000 [cc4c396647]  Matt Jordan <mjordan@digium.com>

	* main/rtp_engine: Fix crash when processing more than one RTCP report info block

	  Asterisk - in res_rtp_asterisk - only understands a single RTCP report info
	  block. When the RTCP information was refactored in the RTP Engine to be pushed
	  over the Stasis message bus, I put in the hooks into the engine to handle
	  multiple RTCP report info blocks, in the hope that a future RTP implementation
	  would be able to provide that data. Unfortunately, res_rtp_asterisk has a
	  tendency to "lie":
	  (1) It will send RTCP reports with a reception_report_count greater than 1
	      (which is pulled directly from the RTCP packet itself, so that part is
	      correct)
	  (2) It will only provide a single report block

	  When the rtp_engine goes to convert this to a JSON blob, hilarity ensues as it
	  looks for a report block that doesn't exist.

	  This patch updates the rtp_engine to be a bit more skeptical about what it is
	  presented with. While this could also be fixed in res_rtp_asterisk, this patch
	  prefers to fix it in the engine for two reasons:
	  (1) The engine is designed to work with multiple RTP implementation, and hence
	      having it be more robust is a good thing (tm)
	  (2) res_rtp_asterisk's handling of RTCP information is "fun". It should report
	      the correct reception_report_count; ideally it should also be giving us all
	      of the blocks - but it is *definitely* not designed to do that. Going down
	      that road is a non-trivial effort.

	  Review: https://reviewboard.asterisk.org/r/4158/

	  ASTERISK-24489 #close
	  Reported by: Gregory Malsack
	  Tested by: Gregory Malsack

	  ASTERISK-24498 #close
	  Reported by: Beppo Mazzucato
	  Tested by: Beppo Maazucato
	  ........

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	  ........

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2014-11-12 14:40 +0000 [ec1a7654f3]  Corey Farrell <git@cfware.com>

	* Fix leak in AMI Action Bridge

	  Add missing reference cleanup for newly created bridge.

	  ASTERISK-24281
	  Reported by: Stefan Engström
	  Review: https://reviewboard.asterisk.org/r/4154/
	  ........

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2014-11-12 10:13 +0000 [dbb8f0a935]  Joshua Colp <jcolp@digium.com>

	* pbx: Fix off-nominal case where a freed extension may still be used.

	  If during the operation of adding an extension a priority is added but
	  fails it is possible for the extension to be freed but still exist in
	  the PBX core. If this occurs subsequent lookups may try to access the
	  extension and end up in freed memory.

	  This change removes the extension from the PBX core when the priority
	  addition fails and then frees the extension.

	  ASTERISK-24444 #close
	  Reported by: Leandro Dardini

	  Review: https://reviewboard.asterisk.org/r/4162/
	  ........

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	  ........

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2014-11-12 07:47 +0000 [9f89b83269]  Corey Farrell <git@cfware.com>

	* Fix compiler error when using ./configure --enable-dev-mode --enable-coverage

	  When DONT_OPTIMIZE is enabled with dev-mode, it causes a shadow compilation
	  to be done with output to /dev/null.  This can cause errors with coverage
	  when GCC attempts to write to /dev/null.gcno.  This change disables
	  coverage for the shadow compilation.

	  ASTERISK-24502 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4151/
	  ........

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2014-11-09 02:01 +0000 [21c41e4542]  Corey Farrell <git@cfware.com>

	* manager: Fix HTTP connection reference leaks.

	  Fix reference leak that happens if (session && !blastaway).

	  ASTERISK-24505 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4153/
	  ........

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2014-11-08 18:38 +0000 [f4392c4b6d]  Xavier Hienne (License 6657)

	* channels/chan_mgcp: Fix regression which causes gateways to be skipped

	  In r227276, a while loop was turned into a for loop. Unfortunately, a portion
	  of the while loop was left in the code such that, when a static gateway is
	  encountered in the list of MGCP gateways, the next gateway would be skipped.
	  At best, we would simply flip past a gateway; at worst, this could lead to a
	  crash.

	  ASTERISK-24500 #close
	  Reported by: Xavier Hienne
	  patches:
	    chan_mgcp.patch uploaded by Xavier Hienne (License 6657)
	  ........

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2014-11-08 18:26 +0000 [d773f9d03e]  Dmitriy Bubnov (License 6651),Dmitry Bubnov (License 6651)

	* addons/chan_mobile: Increase buffer size of UCS2 encoded SMS messages

	  When UCS2 character encoding is used, one symbol in national language can be
	  expanded to 4 bytes. The current buffer used for receiving message in
	  do_monitor_phone is 256 bytes, which is not large enough for incoming messages.

	  For example:
	  * AT+CMGR phone response prefix
	    '+CMGR: "REC UNREAD","+7**********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes
	  * SMS body with UCS2 encoding (max) - 280 bytes
	  * AT+CMGR phone response suffix '\r\n\r\nOK\r\n' - 8 bytes
	  * Terminating null character - 1 byte

	  This results in a needed buffer size of 349 bytes. Hence, this patch opts for a
	  350 byte buffer.

	  ASTERISK-24468 #close
	  Reported by: Dmitriy Bubnov
	  patches:
	    chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651)
	    chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651)
	  ........

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2014-11-08 18:14 +0000 [08d773532b]  abelbeck <lonnie@abelbeck.com> (License 5903)

	* app_voicemail: Fix enhancement that allowed multiple recipients in To: header

	  An issue existed in r420577, which added multiple recipients to voicemail
	  emails. The patch, when looking at the intended recipients, looked ahead for
	  the '|' character inside a while loop which already had pulled out the
	  appropriate field parsing on the '|' character. This would cause it to skip
	  the recipients.

	  This patch fixes it such that it relies completely on the while loop to parse
	  through the e-mail fields.

	  Note that the original author of the patch looked at this fix and approved it.

	  ASTERISK-24250 #close
	  Reported by: abelbeck
	  patches:
	    voicemail-420577-to-comma-fix.diff uploaded by abelbeck (License 5903)
	  ........

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2014-11-08 18:04 +0000 [9a1ab5d548]  Matt Jordan <mjordan@digium.com>

	* bridge_native_rtp: Fix T.38 issues with remote bridges

	  After r425242 the fax/sip/directmedia_reinvite_t38 test started failing due to
	  the surviving channel not being re-INVITEd back from T.38 to audio. This patch
	  fixes that bug - a deeper explanation of what happened follows.

	  When two RTP channels are in a native bridge, the bridging layer will
	  investigate each via the get_rtp_info glue callback. This callback returns the
	  native bridge preference of the channel *at that moment in time* (that part is
	  key). At different points during the bridging, the native bridging layer will
	  inform the RTP capable channels of the status of the bridge via the update_peer
	  glue callback.

	  In a T.38 scenario with audio direct media, the sequence of events will often
	  look like the following:
	   * SIP/A and SIP/B both have audio and enter a native bridge.
	   * Asterisk re-INVITEs audio between SIP/A and SIP/B directly (via an
	     update_peer callback).
	   * SIP/A sends a re-INVITE to T.38, which causes Asterisk to send a re-INVITE
	     to T.38 to SIP/B. Assuming everyone 200 OKs the process, the UDPTL stack
	     receives UDPTL packets in Asterisk from both endpoints. From the perspective
	     of the channels, we are now in a local bridge for T.38, even though we are
	     technically still in a remote bridge in bridge_native_rtp. (YAY!)
	   * When one side hangs up, bridge_native_rtp is told to stop bridging. It then
	     re-evaluates the channels and asks them how they are bridged - and since
	     T.38 is enabled, they reply with a Local bridge (which is correct), but is
	     wrong because the audio portion is still technically in a remote bridge.
	   * Asterisk releases the surviving channel, whose audio is *not* re-INVITED
	     back to Asterisk as bridge_native_rtp incorrectly assumes that it was in a
	     local bridge.

	  Ironically, prior to r425242, this used to work mostly due to a fluke in the
	  bridging layer.

	  The purpose of the get_rtp_info callback shouldn't be modified: it should tell
	  the bridging layer what kind of bridge the channel prefers at that moment in
	  time. If you have T.38 enabled, that *must* be a local bridge, as the UDPTPL
	  stack must be in the media path. As such, this patch does not modify that
	  part of the code.

	  However, we have to tell the channels to re-evaluate themselves when they come
	  out of a native bridge, since we can no longer trust the get_rtp_info callbacks
	  when the native bridge is being stopped. Something else may have changed in the
	  channels, and they may now be lying to us. As such, this patch makes it so that
	  we unilaterally tell the channels that they are no longer bridged via the
	  update_peer callback. This is actually what the channels expect anyway: code in
	  both chan_sip and chan_pjsip's callbacks look at the T.38 state and - if they
	  were in T.38 - send a re-INVITE to get the audio back to Asterisk.

	  Review: https://reviewboard.asterisk.org/r/4157/
	  ........

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2014-11-08 12:20 +0000 [d4fd0774f4]  Corey Farrell <git@cfware.com>

	* chan_console: Fix reference leaks to pvt.

	  Fix a bunch of calls to get_active_pvt
	  where the reference is never released.

	  ASTERISK-24504 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4152/
	  ........

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2014-11-06 13:26 +0000 [7571bae5ab]  Richard Mudgett <rmudgett@digium.com>

	* app_agent_pool: Made agent alert interruptable by DTMF.

	  Made agent able to interrupt the alerting beep playback with DTMF.  Any
	  digit can interrupt if the call does not need to be acknowledged.  Only
	  the first digit of the acknowledgement can interrupt if the call needs to
	  be acknowledged.  The agent interrupting the alerting playback builds on
	  the ASTERISK-24447 patch because it knows what digit interrupted the
	  playback and needs to be able to pass that digit to the DTMF hook digit
	  collection code.

	  ASTERISK-24257 #close
	  Reported by: Steve Pitts

	  Review: https://reviewboard.asterisk.org/r/4123/
	  ........

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2014-11-06 13:12 +0000 [a68baad74f]  Richard Mudgett <rmudgett@digium.com>

	* Bridge DTMF hooks: Made audio pass from the bridge while waiting for more matching digits.

	  * Made collecting DTMF digits for the DTMF feature hooks pass frames from
	  the bridge.

	  * Made collecting DTMF digits possible by other bridge hooks if there is a
	  need.

	  ASTERISK-24447 #close
	  Reported by: Richard Mudgett

	  Review: https://reviewboard.asterisk.org/r/4123/
	  ........

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2014-11-06 12:21 +0000 [47074f4bfd]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Ensure in-dialog responses have an endpoint associated.

	  When handling incoming messages we determine if it is associated with
	  a dialog. If so we use that to determine what serializer and endpoint
	  to use for the message. Previously this would pass the endpoint to the
	  endpoint lookup module to actually place the endpoint completely on the
	  message. For in-dialog responses, however, this did not occur as
	  dialog processing took over and the endpoint lookup did not occur.

	  This change just places the endpoint in the expected spot immediately
	  instead of relying on the endpoint lookup module. In-dialog responses
	  thus have the expected endpoint.

	  AST-1459 #close

	  Review: https://reviewboard.asterisk.org/r/4146/
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2014-11-06 06:15 +0000 [4d80f223af]  Corey Farrell <git@cfware.com>

	* main/file.c: fix possible extra ast_module_unref to format modules.

	  fn_wrapper only adds a reference to the format's module if the file
	  was able to be opened.  If not this causes an unmatched
	  ast_module_unref in filestream_destructor.  Move ast_module_ref to
	  get_stream.

	  ASTERISK-24492 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4149/
	  ........

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2014-11-06 03:24 +0000 [c46664305a]  Corey Farrell <git@cfware.com>

	* res_hep: fix major leak that occurs when config is missing or enabled=no.

	  Add missing unreference in hepv3_send_packet.

	  ASTERISK-24491 #close
	  Reported by: Zane Conkle
	  Tested by: Zane Conkle
	  Review: https://reviewboard.asterisk.org/r/4150/
	  ........

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2014-11-06 03:18 +0000 [7e2369310c]  Corey Farrell <git@cfware.com>

	* Fix unintential memory retention in stringfields.

	  * Fix missing / unreachable calls to __ast_string_field_release_active.
	  * Reset pool->used to zero when the current pool->active reaches zero.

	  ASTERISK-24307 #close
	  Reported by: Etienne Lessard
	  Tested by: ibercom, Etienne Lessard
	  Review: https://reviewboard.asterisk.org/r/4114/
	  ........

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2014-11-05 20:41 +0000 [362dde2229]  gtjoseph <george.joseph@fairview5.com>

	* test_strings:  Remove string tests that exercise asserts.

	  Since unit tests are run with DO_CRASH, those tests were causing
	  the test to fail.

	  Tested-by: George Joseph
	  ........

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2014-11-05 13:53 +0000 [69f29e627f]  Mark Michelson <mmichelson@digium.com>

	* Make the disable_tcp_switch PJSIP system object enabled by default.

	  Testing has shown repeatedly that PJSIP's default behavior of switching
	  automatically to TCP for large messages can cause issues. The most common
	  issues are that devices that we are communicating with do not handle the
	  switch to TCP gracefully, thus causing situations such as broken calls or
	  broken subscriptions. Now, in order to have this behavior happen, you must
	  opt into it. The sample file has been updated to warn that enabling the
	  TCP switch behavior may cause issues for you, so use at your own risk.
	  ........

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2014-11-05 06:19 +0000 [b06078880b]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_multihomed: Add logging during startup to aid debugging if local DNS is misbehaving.

	  This change adds a bit of logging so if the local DNS is misbehaving it is easier
	  to track down what is going on and where Asterisk may be hanging.

	  ASTERISK-24438 #close
	  Reported by: Melissa Shepherd

	  Review: https://reviewboard.asterisk.org/r/4148/
	  ........

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2014-11-04 18:17 +0000 [d5de94201e]  gtjoseph <george.joseph@fairview5.com>

	* config: Make text_file_save and 'dialplan save' escape semicolons in values.

	  When a config file is read, an unescaped semicolon signals comments which are
	  stripped from the value before it's stored.  Escaped semicolons are then
	  unescaped and become part of the value.  Both of these behaviors are normal
	  and expected.  When the config is serialized either by 'dialplan save' or
	  AMI/UpdateConfig however, the now unescaped semicolons are written as-is.
	  If you actually reload the file just saved, the unescaped semicolons are
	  now treated as start of comments.

	  Since true comments are stripped on read, any semicolons in
	  ast_variable.value must have been escaped originally.  This patch
	  re-escapes semicolons in ast_variable.values before they're written to
	  file either by 'dialplan save' or config/ast_config_text_file_save which
	  is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting
	  issues nearby in pbx_config.c

	  Tested-by: George Joseph
	  ASTERISK-20127 #close

	  Review: https://reviewboard.asterisk.org/r/4132/
	  ........

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2014-11-04 16:51 +0000 [c77a71ad2f]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Apply the 'user_eq_phone' setting to the To header as well.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-04 16:31 +0000 [5e43d68717]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427257 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-04 14:49 +0000 [bd42a09d7f]  gtjoseph <george.joseph@fairview5.com>

	* config: BUG: Restore ability for non-templ to be used as base objs in config.

	  My recent refactor of config.c accidentally removed the capability for an
	  object to inherit from a non-template object.

	  This patch restores the capability to inherit from both template and
	  non-template objects.

	  Tested-by: George Joseph
	  Reported-by: Scott Griepentrog
	  ASTERISK-24487 #close

	  Review: https://reviewboard.asterisk.org/r/4147/
	  ........

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2014-11-04 13:46 +0000 [97e1c7f3a9]  Corey Farrell <git@cfware.com>

	* func_talkdetect: Fix stasis message leak in audiohook callback.

	  ASTERISK-24482 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4142/
	  ........

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2014-11-04 13:33 +0000 [9f2874639d]  Corey Farrell <git@cfware.com>

	* res_http_websockets: Fix extra unref of module

	  In websocket_add_protocol_internal is used to add the "echo"
	  protocol, but ast_websocket_remove_protocol is used to remove
	  it.  This causes an extra call to ast_module_unref.

	  ASTERISK-24480 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4140/
	  ........

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2014-11-04 08:11 +0000 [bdc35c77b9]  Corey Farrell <git@cfware.com>

	* Fix crash caused by merge error on review 4138

	  When merging from 12 to 13 there were conflicts,
	  I mistakenly had the loop run ast_closestream(others[0])
	  when it should be ast_closestream(others[x]).
	  ........

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2014-11-04 06:03 +0000 [d159885e50]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_outbound_registration: Add virtual line support.

	  Virtual line support establishes a relationship between messages
	  related to an outbound registration and a local endpoint. This is
	  accomplished by attaching a parameter to the Contact of the outbound
	  registration and looking for it on any received requests. If the
	  parameter exists and can be matched to an outbound registration
	  the configured endpoint is associated with the request.

	  Review: https://reviewboard.asterisk.org/r/2964/


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2014-11-03 12:22 +0000 [33f0251b6c]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Add disable_tcp_switch option.

	  When a packet exceeds the MTU, pjproject will switch from UDP to TCP.  In
	  some circumstances (on some networks), this can cause some issues with
	  messages not getting sent to the correct destination - and can also cause
	  connections to get dropped due to quirks in pjproject deciding to
	  terminate TCP connections with no messages.

	  While fixing the routing/messaging issues is important, having a
	  configuration option in Asterisk that tells pjproject to not switch over
	  to TCP would be useful.  That way, if some glitch is discovered on some
	  other network/site, we can at least disable the behavior until a fix is
	  put into place.

	  AFS-197 #close

	  Review: https://reviewboard.asterisk.org/r/4137/
	  ........

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	  ........

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2014-11-03 09:03 +0000 [b9aeff9580]  Joshua Colp <jcolp@digium.com>

	* chan_pjsip: Update CHANGES file to include 'moh_passthrough' setting


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427113 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-03 08:45 +0000 [ac091d4184]  Joshua Colp <jcolp@digium.com>

	* chan_pjsip: Add support for passing hold and unhold requests through.

	  This change adds an option, moh_passthrough, that when enabled will pass
	  hold and unhold requests through using a SIP re-invite. When placing on
	  hold a re-invite with sendonly will be sent and when taking off hold a
	  re-invite with sendrecv will be sent. This allows remote servers to handle
	  the musiconhold instead of the local Asterisk instance being responsible.

	  Review: https://reviewboard.asterisk.org/r/4103/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-11-02 20:36 +0000 [285be15aaf]  Corey Farrell <git@cfware.com>

	* Fix compile error caused by review 4138

	  There is no procedure called ast_closeframe, fix code to use
	  ast_closestream.

	  Reported By: Matt Jordan
	  ........

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2014-11-02 02:13 +0000 [509c04ef38]  Corey Farrell <git@cfware.com>

	* Fix ast_writestream leaks

	  Fix cleanup in __ast_play_and_record where others[x] may be leaked.
	  This was caught where prepend != NULL && outmsg != NULL, once
	  realfile[x] == NULL any further others[x] would be leaked. A cleanup
	  block was also added for prepend != NULL && outmsg == NULL.

	  11+: Fix leak of ast_writestream recording_fs in
	  app_voicemail:leave_voicemail.

	  ASTERISK-24476 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4138/
	  ........

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2014-11-02 01:40 +0000 [85c1822a9d]  Corey Farrell <git@cfware.com>

	* func_jitterbuffer: fix frame leaks.

	  Fix code paths where it is possible for frames to leak.
	  Fix uninitialized variable in jb_get_fixed and jb_get_adaptive.

	  ASTERISK-22409 #related
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4128/
	  ........

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2014-11-01 20:01 +0000 [5db1c978e3]  Matt Jordan <mjordan@digium.com>

	* res/res_stasis: Fix crash on module unload while performing operation

	  When the res_stasis module is unloaded, it will dispose of the apps_registry
	  container. This is a problem if an ARI operation is in flight that attempts
	  to use the registry, as the shutdown occurs in a separate thread. This patch
	  adds some sanity checks to the various routines that access the registry which
	  cause the operations to fail if the apps_registry does not exist.

	  Crash caught by the Asterisk Test Suite.
	  ........

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2014-10-31 11:52 +0000 [4219c40775]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* install init.d files on GNU/kFreeBSD

	  Review: https://reviewboard.asterisk.org/r/4118/
	  ........

	  Merged revisions 426926 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2014-10-31 11:41 +0000 [28173ddf05]  Scott Griepentrog <sgriepentrog@digium.com>

	* pjsip: clarify tls cert and key file usage

	  A question arose as to whether a .pem file
	  could be provided in place of the .crt and
	  .key files in a PJSIP TLS configuration. I
	  tested this and discovered that although a
	  cert will be read from the pem file, a key
	  will not, and thus the priv_key_file entry
	  is still required. This update to the fine
	  documentation clarifies the option usage.

	  AST-1448 #close
	  Review: https://reviewboard.asterisk.org/r/4129/
	  Reported by: John Bigelow
	  ........

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2014-10-31 11:24 +0000 [f59db388a7]  John Bigelow (License 5091)

	* pjsip: Handle outbound unregister correctly

	  This updates the status of the outbound registration
	  to reflect when it has been unregistered.  Since the
	  registration is unregistered but is not stopped, the
	  registration schedule remains active as before.  The
	  patch also updates the documentation of both the AMI
	  and CLI commands.

	  ASTERISK-24411 #close
	  Review: https://reviewboard.asterisk.org/r/4119/
	  Reported by: John Bigelow
	  patches:
	    unregister-patch1.txt uploaded by John Bigelow (License 5091)
	  ........

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2014-10-30 22:26 +0000 [d88282af40]  Matt Jordan <mjordan@digium.com>

	* channels/sip/reqresp_parser: Fix unit tests for r426594

	  When r426594 was made, it did not take into account a unit test that verified
	  that the function properly populated the unsupported buffer. The function
	  would previously memset the buffer if it detected it had any contents; since
	  this function can now be called iteratively on successive headers, the unit
	  tests would now fail. This patch updates the unit tests to reset the buffer
	  themselves between successive calls, and updates the documentation of the
	  function to note that this is now required.
	  ........

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2014-10-30 22:09 +0000 [bf684b63a3]  Corey Farrell <git@cfware.com>

	* REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scripts

	  This change ensures refcounter.py is installed to a place where it
	  can be found by the Asterisk testsuite if REF_DEBUG is enabled.

	  ASTERISK-24432 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4094/
	  ........

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2014-10-30 18:56 +0000 [e4374a3abe]  Corey Farrell <git@cfware.com>

	* app_queue: fix a couple leaks to struct call_queue in set_member_value

	  set_member_value has a couple leaks to references in the variable q
	  found through testsuite tests/queues/set_penalty.  Also remove the
	  REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible
	  with the updated REF_DEBUG code.

	  ASTERISK-24466 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4125/
	  ........

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2014-10-30 18:45 +0000 [ced81afff2]  Corey Farrell <git@cfware.com>

	* audiohooks: Clean references to formats

	  Cleanup references to in_translate[x].format and
	  out_translate[x].format in ast_audiohook_detach_list.

	  ASTERISK-24465 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4124/
	  ........

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2014-10-30 16:14 +0000 [a537e314d1]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash

	  Currently, it is possible for some subscriptions to get into a NULL state. When
	  this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a
	  device is subscribed for extension state then the associated subscription state
	  object can't be located.  The code then attempts to dereference a NULL object.
	  Added a NULL check to avoid the problem.

	  Reported by: John Bigelow
	  ........

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2014-10-30 12:18 +0000 [cd52456ea1]  Kevin Harwell <kharwell@digium.com>

	* res_pjsip: incorrect qualify statistics after disabling for contact

	  When removing the qualify_frequency from an AoR or a contact the statistics
	  shown when issuing "pjsip show aors" from the CLI are incorrect. This patch
	  deletes the contact's status object from sorcery, disassociating it from the
	  contact, if the qualify_freqency is removed from configuration.

	  ASTERISK-24462 #close
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4116/
	  ........

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2014-10-30 04:21 +0000 [5d8d90c402]  Walter Doekes <walter+asterisk@wjd.nu>

	* app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE.

	  In update_messages_by_imapuser(), messages were appended to a finite
	  array which resulted in a crash when an IMAP mailbox contained more
	  than 256 entries. This memory is now dynamically increased as needed.

	  Observe that this patch adds a bunch of XXX's to questionable code. See
	  the review (url below) for more information.

	  ASTERISK-24190 #close
	  Reported by: Nick Adams
	  Tested by: Nick Adams

	  Review: https://reviewboard.asterisk.org/r/4126/
	  ........

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2014-10-30 01:15 +0000 [c866ced76b]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* 
	  Add additional checks for NULL pointers to fix several crashes reported.

	  ASTERISK-24304 #close
	  Reported by: dhanapathy sathya
	  ........

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2014-10-29 20:59 +0000 [0ddc3bde24]  Olle Johansson <oej@edvina.net> (License 5267)

	* channels/chan_sip: Add improved support for 4xx error codes

	  This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER
	  response handling. This helps interoperability in a number of scenarios.

	  Review: https://reviewboard.asterisk.org/r/3437

	  patches:
	    rb3437.patch uploaded by oej (License 5267)
	  ........

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2014-10-29 20:48 +0000 [ff83ff564c]  Olle Johansson <oej@edvina.net> (License 5267)

	* channels/chan_sip: Support mutltiple Supported and Required headers

	  A SIP request may contain multiple Supported: and Required: headers. Currently,
	  chan_sip only parses the first Supported/Required header it finds. This patch
	  adds support for multiple Supported/Required headers for INVITE requests.

	  Review: https://reviewboard.asterisk.org/r/2478

	  ASTERISK-21721 #close
	  Reported by: Olle Johansson
	  patches:
	    rb2478.patch uploaded by oej (License 5267)
	  ........

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2014-10-29 08:02 +0000 [8a69aedd17]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* Fix building chan_phone on big endian systems

	  A left over from the formats conversion (Corey Farrell).

	  ASTERISK-24458 #close
	  Review: https://reviewboard.asterisk.org/r/4117/

	  ........

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2014-10-28 16:35 +0000 [0ed8aebda9]  Richard Mudgett <rmudgett@digium.com>

	* bridge_builtin_features: Add missing channel locks around ast_get_chan_features_general_config().

	  The feature_automonitor() and feature_automixmonitor() functions were not
	  locking the channel around ast_get_chan_features_general_config().
	  Accessing the channel datastore list without the channel locked is a good
	  way to corrupt the list or follow the pointer chain into oblivion.
	  ........

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2014-10-28 16:10 +0000 [7205d76d7d]  Corey Farrell <git@cfware.com>

	* res_fax: Resolve T38 gateway frame leak.

	  When frames are translated by a fax gateway they need to be freed.  The
	  existing call to ast_frfree was unreachable.  This change reorganizes
	  fax_gateway_framehook to ensure that ast_frfree is called when needed.

	  ASTERISK-24457 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4115/
	  ........

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2014-10-28 15:44 +0000 [67e496c275]  Corey Farrell <git@cfware.com>

	* manager: Unsubscribe from acl_change_sub at shutdown.

	  ASTERISK-24453 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4110/
	  ........

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2014-10-28 13:09 +0000 [1fe22c411d]  Malcolm Davenport <malcolmd@digium.com>

	* ASTERISK-23512, correct inaccurate comment in manager.conf.sample

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-10-28 11:41 +0000 [8e9f593e3a]  Matt Jordan <mjordan@digium.com>

	* main/bridge: Destroy features struct on off nominal path during bridge impart

	  When a channel is imparted to a bridge, the invocation of the function may
	  provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart,
	  the caller must assume that ownership has passed to the function, as in all
	  paths the function destroys the struct prior to returning (as its purpose is
	  to configure the behavior of the channel while in the bridge). On one off
	  nominal path - where the channel already has a PBX thread - the struct was not
	  being destroyed.

	  This patch fixes that glitch.

	  ASTERISK-24437 #close
	  Reported by: Scott Griepentrog
	  ........

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2014-10-28 09:59 +0000 [f4b4d42630]  Matt Jordan <mjordan@digium.com>

	* main/manager: Fix typo in AMI event documentation of "OriginateResponse"

	  The parameter name is "Response", not "Resonse".

	  ASTERISK-24430 #close
	  Reported by: Dafi Ni
	  ........

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2014-10-28 09:57 +0000 [68d9872f58]  Malcolm Davenport <malcolmd@digium.com>

	* ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-10-28 08:13 +0000 [684b8762a9]  Malcolm Davenport <malcolmd@digium.com>

	* ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-10-28 06:22 +0000 [2290393273]  Corey Farrell <git@cfware.com>

	* app_queue: Cleanup ao2_iterator

	  Clean ao2_iterator, resolving reference leak to queue members.

	  ASTERISK-24454 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4111/
	  ........

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2014-10-28 06:12 +0000 [ab16f46139]  Corey Farrell <git@cfware.com>

	* func_cdr: Fix CDR_PROP payload leak

	  Remove duplicate allocation of payload, preventing leak.

	  ASTERISK-24455 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4113/
	  ........

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2014-10-27 12:55 +0000 [ef8cdd40e5]  Sean Bright <sean@malleable.com>

	* configure: Add autoconf check for libopus.

	  Because opus transcoding support cannot be included in the standard Asterisk
	  distribution, a few codec_opus implementations have popped up.  To make it
	  easier for people to drop in opus support in their own installations, this
	  patch adds configure checks for libopus.

	  Review: https://reviewboard.asterisk.org/r/4106/
	  ........

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2014-10-26 21:47 +0000 [5a17878085]  Matt Jordan <mjordan@digium.com>

	* res/res_http_websocket: Fix minor nits found by wdoekes on r409681

	  When Moises committed the fixes for WSS (which was a great patch), wdoekes had
	  a few style nits that were on the review that got missed. This patch resolves
	  what I *think* were all of the ones that were still on the review.

	  Thanks to both moy for the patch, and wdoekes for the reviews.

	  Review: https://reviewboard.asterisk.org/r/3248/
	  ........

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2014-10-26 21:27 +0000 [62bee9b327]  Matt Jordan <mjordan@digium.com>

	* res/res_phoneprov: Fix crash on shutdown caused by container cleanup

	  In res_phoneprov, unloading the module first destroys the http_routes
	  container, followed by the users. However, users may have a route in
	  the http_routes container; the validity of this container is not checked
	  in the users destructor. Hence, we hit an assert as the container has already
	  been set to NULL.

	  This patch does two things:
	  (1) It adds a sanity check in the user destructor (because why not)
	  (2) It switches the order of destruction, so that users are disposed of prior
	      to the HTTP routes they may hold a reference to.

	  Note that this crash was caught by the Test Suite (go go testing!)
	  ........

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2014-10-26 20:47 +0000 [130a3fcd7f]  Matt Jordan <mjordan@digium.com>

	* res/res_srtp: Fix include issue for libsrtp 1.5.0

	  In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including
	  srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this
	  header file has been provided by the library since 2006, so this is a
	  relatively benign change.

	  ASTERISK-24436 #close
	  Reported by: Patrick Laimbock
	  ........

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2014-10-24 10:32 +0000 [c084728690]  Jonathan Rose <jrose@digium.com>

	* Documentation: Improve documentation for ExtensionStatus AMI events

	  Review: https://reviewboard.asterisk.org/r/4085/
	  ........

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2014-10-22 16:41 +0000 [c4d7e7e270]  Shaun Ruffell <sruffell@digium.com>

	* codec_dahdi: Cannot use struct ast_translator.core_{src,src}_codec.

	  This fixes a Segmentation fault introduced in r419044 "media formats: re-architect
	  handling of media for performance improvements".

	  The problem is that codec_dahdi was using core_src_codec and core_dst_codec in the
	  ast_translator structure when these fields were never set. Now instead of trying to map
	  the new core codec descriptions to the way DAHDI defines different codecs, we will store
	  the DAHDI specific formats in 'struct translator' directly so we can refer to them without
	  mapping.

	  This also allows us to remove the "global_format_map" structure, since we can now query
	  the list of translators directly to make sure we do not ever register a DAHDI based
	  translator for a specific path more than once and eliminate the need to keep the list and
	  the map in sync.

	  ASTERISK-24435 #close
	  Reported by: Marian Koniuszko

	  Review: https://reviewboard.asterisk.org/r/4105/
	  ........

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2014-10-21 13:04 +0000 [2165868be7]  Richard Mudgett <rmudgett@digium.com>

	* translage.c: Fix regression when generating translation path strings.

	  Fix the AMI Status action read and write translation path strings from
	  growing for each channel in the status event list by reseting the ast
	  string given to ast_translate_path_to_str() to fill in the given
	  translation path.
	  ........

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2014-10-20 09:20 +0000 [dad0334cf1]  abelbeck <lonnie@abelbeck.com> (License 5903),Matt Jordan <mjordan@digium.com> (License 6283)

	* AST-2014-011: Fix POODLE security issues

	  There are two aspects to the vulnerability:
	  (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use
	      TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the
	      TCP/TLS core, which should be done as an improvement at a latter date.
	  (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified,
	      will default to the OpenSSL SSLv23_method. This method allows for all
	      ecnryption methods, including SSLv2/SSLv3. A MITM can exploit this by
	      forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE.
	      This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration,
	      and explicitly disables SSLv2/SSLv3 if using SSLv23_method.

	  For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is
	  explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or
	  SSLv3.

	  Much thanks to abelbeck for reporting the vulnerability and providing a patch
	  for the res_jabber/res_xmpp modules.

	  Review: https://reviewboard.asterisk.org/r/4096/

	  ASTERISK-24425 #close
	  Reported by: abelbeck
	  Tested by: abelbeck, opsmonitor, gtjoseph
	  patches:
	    asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903)
	    asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903)
	    AST-2014-011-1.8.diff uploaded by mjordan (License 6283)
	    AST-2014-011-11.diff uploaded by mjordan (License 6283)
	  ........

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2014-10-19 12:09 +0000 [5e10e369b1]  gtjoseph <george.joseph@fairview5.com>

	* build: Force -fsigned-char on platforms where the default for char is unsigned

	  gcc on the ARM platform defaults 'char' to 'unsigned char' whereas Intel and
	  SPARC default to 'signed char'.  This is only an issue in the rare cases where
	  negative values are assigned to a 'char' but this this patch insures
	  compatibility by detecting platforms that default to 'unsigned' and adding an
	  '-fsigned-char' flag to _ASTCFLAGS.

	  If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh
	  and ./configure to regenerate the build files.  You shouldn't have to do this
	  for Intel or SPARC.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4091/
	  ........

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2014-10-18 23:03 +0000 [404b6ab3ab]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_sdp_rtp: Revert 425924

	  This patch for r425924 introduced a bug, wherein sending an INVITE request
	  with no SDP would cause Asterisk to not send an SDP Offer in the 200
	  OK. The current structure of res_pjsip_sdp_rtp is a bit hard to deal with
	  to fix this, as create_outgoing_sdp has no knowledge of whether or not it is
	  creating an SDP as a new Offer or an Answer. This is something of an oversight
	  in the callback definition, as the caller of it does have this information.



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2014-10-18 19:56 +0000 [b263c8bdae]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_sdp_rtp: Remove left over reference to override_prefs

	  The usage of the local override_prefs variable in create_outgoing_sdp_stream
	  was previously to track an override format preference set by PJSIP_MEDIA_OFFER.
	  Now, however, that function simply sets the joint capabilities structure,
	  session->req_caps. During the media format rework, the override_prefs was
	  instead used to check if there were any formats in session->req_caps.

	  However, this usage isn't useful in create_outgoing_sdp_stream.
	  session->req_caps contains the negotiated formats for *all* streams, not just
	  the current one being created. Thus, so long as any stream of any type has
	  provided a format, override_prefs will be non-zero. Hence, its usage in
	  checking whether or not we should look at the formats on the endpoint or
	  the joint capabilities is generally useless.

	  There's only two things useful to check:
	  (1) Does the endpoint have a format for the media type?
	  (2) Did we negotiate a format for the media type?

	  If either of those is a 'no', then we must kill the media stream.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-10-17 17:45 +0000 [b8f687f27c]  Jonathan Rose <jrose@digium.com>

	* Sample Configurations: make 'pjsip reload' reload all reloadable pjsip modules

	  AST-1432 #close
	  Reported by: John Bigelow
	  ........

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2014-10-17 08:35 +0000 [8f58592252]  Matt Jordan <mjordan@digium.com>

	* res_pjsip_session/res_pjsip_sdp_rtp: Be more tolerant of offers

	  When an inbound SDP offer is received, Asterisk currently makes a few
	  incorrection assumptions:

	  (1) If the offer contains more than a single audio/video stream, Asterisk will
	      reject the entire stream with a 488. This is an overly strict response;
	      generally, Asterisk should accept the media streams that it can accept and
	      decline the others.
	  (2) If the offer contains a declined media stream, Asterisk will attempt to
	      process it anyway. This can result in attempting to match format
	      capabilities on a declined media stream, leading to a 488. Asterisk should
	      simply ignore declined media streams.
	  (3) Asterisk will currently attempt to handle offers with AVPF with
	      use_avpf=No/AVP with use_avpf=Yes. This mismatch results in invalid SDP
	      answers being sent in response. If there is a mismatch between the media
	      type being offered and the configuration, Asterisk must reject the offer
	      with a 488.

	  This patch does the following:
	  * Asterisk will accept SDP offers with at least one media stream that it can
	    use. Some WARNING messages have been dropped to NOTICEs as a result.
	  * Asterisk will not accept an offer with a media type that doesn't match its
	    configuration.
	  * Asterisk will ignore declined media streams properly.

	  #SIPit31

	  Review: https://reviewboard.asterisk.org/r/4063/

	  ASTERISK-24122 #close
	  Reported by: James Van Vleet

	  ASTERISK-24381 #close
	  Reported by: Matt Jordan
	  ........

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2014-10-17 08:17 +0000 [0d0e38a0e1]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports.

	  This change adds a module which is configurable using the keep_alive_interval setting in the
	  global section that will send a CRLF keep alive to all active connection-oriented transports at
	  the provided interval. This is useful because it can help keep connections open through NATs.
	  This functionality also exists within PJSIP but can not be controlled at runtime and requires
	  recompiling it.

	  Review: https://reviewboard.asterisk.org/r/4084/


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2014-10-17 08:11 +0000 [86eea19c8f]  Damian Ivereigh (License 6632)

	* channels/chan_sip: Respect outboundproxy setting when sending qualify requests

	  The outboundproxy setting is currently ignored when sending OPTIONS requests
	  as a result of the qualify setting. This means that if an Asterisk server is
	  unable to send the packet directly to a peer, it is unable to qualify any
	  non-inbound registered peer (e.g. a peer SIP Trunk).

	  This patch grabs the outboundproxy information for a peer when a qualify
	  attempt is being constructed and, if it finds the information, uses it
	  when sending the OPTIONS request.

	  Review: https://reviewboard.asterisk.org/r/3948

	  ASTERISK-24063 #close
	  Reported by: Damian Ivereigh
	  patches:
	    outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632)
	  ........

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2014-10-17 06:30 +0000 [7144c739e9]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable.

	  This change adds a configuration option which adds a 'user=phone' parameter if the user
	  portion of the request URI or the From URI is determined to be a number.

	  Review: https://reviewboard.asterisk.org/r/4073/


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2014-10-16 21:49 +0000 [f91cb1207c]  Richard Mudgett

	* AMI: Add missing VarSet events when a channel inherits variables.

	  There should be AMI VarSet events when channel variables are inherited by
	  an outgoing channel.  Also local;2 should generate VarSet events when it
	  gets all of its channel variables from channel local;1.

	  ASTERISK-24415 #close
	  Reported by: Richard Mudgett
	  Patches:
	        jira_asterisk_24415_v12.patch (license #5621) patch uploaded by Richard Mudgett

	  Review: https://reviewboard.asterisk.org/r/4074/
	  ........

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2014-10-16 21:01 +0000 [df59a71b83]  Matt Jordan <mjordan@digium.com>

	* bridge_native_rtp: Fix audio issues when moving from remote bridge to softmix

	  When a native RTP bridge that is remotely bridging its participants switches
	  to a softmix bridge, it may not properly re-INVITE the media for one or both
	  participants back to Asterisk. This is due to the current bridge_native_rtp
	  code only re-INVITEs if it believes the channel will survive the bridge
	  operation. Currently, that code is failing, as it expects the channels to
	  have a soft hangup flag set on it indicating that a redirect has occurred
	  or that the channel is going to leave the bridge. (The code did not take into
	  account a smart bridge operation).

	  This patch also renames a few things to be more reflective of the underlying
	  types.

	  Review: https://reviewboard.asterisk.org/r/3997/

	  ASTERISK-24327 #close
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2014-10-16 20:46 +0000 [2ccbdd2624]  Matt Jordan <mjordan@digium.com>

	* test_cel: Update pickup test to expect CANCEL instead of ANSWSER

	  The CEL pickup test previously looked for a disposition of ANSWER between the
	  original caller/peer when the call is picked up. This is actually incorrect:
	  the disposition should, at the very least, not be ANSWER as the call was
	  never ANSWERed. The disposition is now CANCEL; this patch updates the test
	  accordingly.
	  ........

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2014-10-16 16:21 +0000 [873d956144]  Matt Jordan <mjordan@digium.com>

	* main/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size'

	  When refactoring CDRs to use the configuration framework, a 'whoops' was
	  introduced where the CDR batch size was used when rescheduling a batch,
	  as opposed to the time duration. This patch corrects that obvious mistake.

	  ASTERISK-24426 #close
	  Reported by: Shane Blaser
	  ........

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2014-10-16 12:32 +0000 [c2ec5f0f6f]  gtjoseph <george.joseph@fairview5.com>

	* config: Fix inf loop using ast_category_browse and ast_variable_retrieve

	  Fix infinite loop when calling ast_variable_retrieve inside an
	  ast_category_browse loop when there is more than 1 category with
	  the same name.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4089/
	  ........

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2014-10-16 11:32 +0000 [86a4ce4957]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Enforce module load dependencies

	  This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub
	  have loaded properly before attempting to load any modules that depend
	  on them since the module loader system is not currently capable of
	  resolving module dependencies on its own.

	  ASTERISK-24312 #close
	  Reported by: Dafi Ni
	  Review: https://reviewboard.asterisk.org/r/4062/
	  ........

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2014-10-16 01:22 +0000 [a770ca168d]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* 
	  Fix loss of voice after second call drops (on a second line) in case using multiple lines on unistim phones. There is regression was introduced in r391379.

	  Reported by: Rustam Khankishyiev
	  (closes issue ASTERISK-23846)
	  ........

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2014-10-15 20:26 +0000 [bfee1b4bc5]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't.

	  In the case where the ICE negotiation had not yet started current state would
	  get wiped when it shouldn't.

	  This also removes channel binding as in practice this does not work well with
	  other implementations.
	  ........

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2014-10-15 14:39 +0000 [28c11fff78]  Richard Mudgett <rmudgett@digium.com>

	* chan_motif: Cleanup jingle_tech.capabilities only once.
	  ........

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2014-10-15 14:17 +0000 [3d58066de9]  Jonathan Rose <jrose@digium.com>

	* parking_tests: Fix assertions and possibly crashes in res_parking unit tests

	  Assertions were caused by attempting to play music on hold to a channel with
	  no formats. Parking unit test channels were given formats and a technology so
	  that they would be able to pretend to read/write frames.

	  ASTERISK-24413 #close
	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/4075/
	  ........

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2014-10-15 05:03 +0000 [90c98d384b]  Alexandr Anikin <may@telecom-service.ru>

	* chan_ooh323: fix rtptimeout general value checking

	  correct condition to check rtptimeout in [general] config section

	  ASTERISK-24393 #close
	  Reported by:  Dmitry Melekhov
	  Tested by:  Dmitry Melekhov
	  Patches:
	    ASTERISK-24393.patch
	  ........

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2014-10-14 15:48 +0000 [104fca5001]  gtjoseph <george.joseph@fairview5.com>

	* config: Fix SEGV in unit test with MALLOC_DEBUG

	  With MALLOC_DEBUG the /main/config config_basic_ops test was causing a
	  SEGV while doing an ast_category_delete in an ast_category_browse loop.
	  Apparently this never worked but was also never tested.  I removed the
	  test, added 2 notes to config.h indicating that it's not supported and
	  added a few lines of code to ast_category_delete to prevent the SEGV
	  should someone attempt it in the future.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4078/
	  ........

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2014-10-14 14:12 +0000 [87b5006ff0]  Jonathan Rose <jrose@digium.com>

	* Scheduler: Fix a nasty scheduler caching bug which makes new tasks not execute

	  Tasks that were marked for pending deletion in the scheduler would be moved to
	  the cache for later reuse, but after being recycled the deleted mark wouldn't
	  be removed resulting in fresh tasks being deleted without reason... and
	  immediately moved back into the cache where they could be reused again. This
	  could cause horrendous things to happen in just about anything that used a
	  scheduler.

	  ASTERISK-24321 #close
	  Reported by: Steve Pitts
	  Review: https://reviewboard.asterisk.org/r/4071/
	  ........

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2014-10-14 13:13 +0000 [527b58aeb7]  gtjoseph <george.joseph@fairview5.com>

	* res_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup

	  Based on feedback from Richard, I created an accessor for
	  res_phoneprov/ast_phoneprov_std_variable_lookup and added
	  load priority to AST_MODULE_INFO.

	  Tested-by: George Joseph
	  Tested-by: Richard Mudgett

	  Review: https://reviewboard.asterisk.org/r/4076/
	  ........

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2014-10-14 11:47 +0000 [fbb19db0c8]  Corey Farrell <git@cfware.com>

	* res_fax: Fix reference leak caused by gateway sessions

	  Fax gateway session objects can be re-used, causing the
	  same gateway session to be added to faxregistry.container
	  more than once.  This change causes fax_session_new to
	  remove the reserved session from the container before
	  it's id is changed, ensuring it's possible for the
	  session to be freed.

	  ASTERISK-24392 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4049/
	  ........

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2014-10-14 11:43 +0000 [c61b66e107]  Richard Mudgett <rmudgett@digium.com>

	* stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2)

	  Masquerades into and out of channels that are involved in a dial operation
	  don't create the expected dial end event.  The missing dial end event goes
	  against the model for things like CDRs and generating Dial end manager
	  actions and such.

	  There are four cases:

	  1) A channel masquerades into the caller channel.  The case happens when
	  performing a blonde transfer using the channel driver's protocol.

	  2) A channel masquerades into a callee channel.  The case happens when
	  performing a directed call pickup.

	  3) The caller channel masquerades out of dial.  The case happens when
	  using the Bridge application on the caller channel.

	  4) A callee channel masquerades out of dial.  The case happens when using
	  the Bridge application on a peer channel.

	  As it turned out, all four cases need to be handled instead of just the
	  first one.

	  ASTERISK-24237
	  Reported by: Richard Mudgett

	  ASTERISK-24394 #close
	  Reported by: Richard Mudgett

	  Review: https://reviewboard.asterisk.org/r/4066/
	  ........

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2014-10-14 11:20 +0000 [01bdc80475]  Corey Farrell <git@cfware.com>

	* res_fax: Resolve module reference leak caused by reserved sessions

	  Remove reference to module providing reserved session after
	  adding a reference to the final module.  This re-reference
	  is done to ensure that module references are correct even
	  if the final session selects a different module than the
	  reserved session.

	  ASTERISK-18923 #close
	  Reported by: Grigoriy Puzankin
	  Review: https://reviewboard.asterisk.org/r/4048/
	  ........

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2014-10-13 11:12 +0000 [c7e6b6ba3d]  gtjoseph <george.joseph@fairview5.com>

	* manager/config: Support templates and non-unique category names via AMI

	  This patch provides the capability to manipulate templates and categories
	  with non-unique names via AMI.

	  Summary of changes:

	  GetConfig and GetConfigJSON: Added "Filter" parameter:  A comma separated list
	  of name_regex=value_regex expressions which will cause only categories whose
	  variables match all expressions to be considered.  The special variable name
	  TEMPLATES can be used to control whether templates are included.  Passing
	  'include' as the value will include templates along with normal categories.
	  Passing 'restrict' as the value will restrict the operation to ONLY templates.
	  Not specifying a TEMPLATES expression results in the current default behavior
	  which is to not include templates.

	  UpdateConfig: NewCat now includes options for allowing duplicate category
	  names, indicating if the category should be created as a template, and
	  specifying templates the category should inherit from.  The rest of the
	  actions now accept a filter string as defined above.  If there are non-unique
	  category names, you can now update specific ones based on variable values.

	  To facilitate the new capabilities in manager, corresponding changes had to be
	  made to config, most notably the addition of filter criteria to many of the
	  APIs.  In some cases it was easy to change the references to use the new
	  prototype but others would have required touching too many files for this
	  patch so a wrapper with the original prototype was created.  Macros couldn't
	  be used in this case because it would break binary compatibility with modules
	  such as res_digium_phone that are linked to real symbols.

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4033/
	  ........

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2014-10-12 16:09 +0000 [8d6f1d763c]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Make the ICE transport check case insensitive as some implementations use 'udp'.
	  ........

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2014-10-12 03:17 +0000 [9e72c74db5]  Walter Doekes <walter+asterisk@wjd.nu>

	* chan_sip: Fix so asterisk won't send reINVITE after a BYE.

	  After a reINVITE glare situation, Asterisk would re-send the reINVITE
	  even though the call had been hung up in the mean time.  This patch
	  unschedules the reinvite when handling the BYE.

	  ASTERISK-22791 #close
	  Reported by: Paolo Compagnini
	  Tested by: Paolo Compagnini

	  Review: https://reviewboard.asterisk.org/r/4056/
	  (testcase is in review r4055)
	  ........

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2014-10-12 02:57 +0000 [c0ac874106]  Walter Doekes <walter+asterisk@wjd.nu>

	* build: Relax badshell tilde test to allow for ~ in middle of DESTDIR.

	  The main Makefile has a target test called 'badshell' that tests if
	  DESTDIR does not happen to have an an-expanded tilde (~).  This might
	  be the case if you run: make install DESTDIR=~/somewhere/

	  That test also disallowed valid tildes in directory names. The test is
	  now changed to only trigger on a tilde at the start of the path.

	  ASTERISK-13797 #close
	  Reported by: Tzafrir Cohen

	  Review: https://reviewboard.asterisk.org/r/4064/
	  ........

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2014-10-12 02:47 +0000 [2a03efdbae]  Walter Doekes <walter+asterisk@wjd.nu>

	* res_calendar_ews: Relax neon version check to work with 0.30 too.

	  Allow res_calendar_ews to work not only with libneon-0.29 but also
	  with 0.30.

	  ASTERISK-24325 #close
	  Reported by: Tzafrir Cohen

	  Review: https://reviewboard.asterisk.org/r/4068/
	  ........

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2014-10-11 16:09 +0000 [6a3c11c75b]  gtjoseph <george.joseph@fairview5.com>

	* res_phoneprov: Cleanup module load error handling

	  Tested module load/reload interaction between res_phoneprov and
	  res_pjsip_phoneprov_provider in cases where res_phoneprov didn't
	  load correctly (usually misconfiguration or missing phoneprov.conf)

	  Tested-by: George Joseph

	  Review: https://reviewboard.asterisk.org/r/4069/
	  ........

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2014-10-10 15:48 +0000 [98d5b7090d]  Joshua Colp <jcolp@digium.com>

	* bridge: During a smart bridge operation provide a more complete bridge to the old technology.

	  When a smart bridge operation occurs and a bridge transitions from one
	  technology to another the old technology is provided the channels formerly
	  in it and told that they are leaving. Unfortunately the bridge provided
	  along with them is incomplete. The bridge, despite there being channels in it,
	  contains none. This forces technology implementations to have additional
	  logic when channels are leaving or to store their own duplicated
	  state.

	  This change makes the bridge more complete so it contains the expected
	  channels. Now that the bridge is complete special logic within
	  bridge_native_rtp is no longer needed and has been removed.

	  Review: https://reviewboard.asterisk.org/r/4057/
	  ........

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2014-10-10 09:31 +0000 [c3ff212cae]  Matt Jordan <mjordan@digium.com>

	* res/res_phoneprov: Bail on registration if res_phoneprov didn't load

	  If res_phoneprov failed to fully load (due to not being configured), the
	  providers container will be NULL. If a module attempts to register a phone
	  provisioning provider, it should check for the presence of the container.
	  If there is no providers container, it should return an error.

	  This patch makes the ast_phoneprov_provider_register function do that...
	  otherwise this would be a silly commit message.
	  ........

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2014-10-10 09:24 +0000 [c46100ad5f]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_phoneprov_provider: Add missing dependency on pjproject.
	  ........

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2014-10-10 08:03 +0000 [37b5f52da7]  Kinsey Moore

	* CallerID: Fix parsing regression

	  This fixes a regression in callerid parsing introduced when another bug
	  was fixed. This bug occurred when the name was composed entirely of
	  DTMF keys and quoted without a number section (<>). 

	  ASTERISK-24406 #close
	  Reported by: Etienne Lessard
	  Tested by: Etienne Lessard
	  Patches:
	      callerid_fix.diff uploaded by Kinsey Moore
	  Review: https://reviewboard.asterisk.org/r/4067/
	  ........

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2014-10-10 07:10 +0000 [0ef680cff0]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_nat: Place source port into rport of responses if 'force_rport' is on.

	  When the 'force_rport' option is enabled the behavior should be the same
	  as if the remote side placed rport into the message themselves. Therefore
	  any responses we send should include the source port of the request in the
	  rport of the Via header.

	  #SIPit31

	  ASTERISK-24387 #close
	  Reported by: Matt Jordan
	  ........

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2014-10-10 02:34 +0000 [d3f525fd8f]  Torrey Searle (License #5334),Nitesh Bansal (License #6418)

	* chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE.

	  If a device re-INVITEs at the same time as the dialog is hung up, and
	  if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would
	  fail to destroy the dialog after a while.  This resulted in (most
	  prominently) file handle leaks.

	  (Patch reindented by me.)

	  ASTERISK-20784 #close
	  ASTERISK-15879 #close
	  Reported by: Torrey Searle, Nitesh Bansal
	  Patches:
	    reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334)
	    patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418)

	  Reviewboard: https://reviewboard.asterisk.org/r/4052/
	  (testcase can be found at r4051)
	  ........

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2014-10-09 18:37 +0000 [aef63118da]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_phoneprov_provider: fix compile breakage on AST_VECTOR

	  endpoint->inbound_auths was changed to a vector in 13 and I
	  committed the 12 patch instead of the 13 patch.

	  Tested-by: George Joseph
	  ........

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2014-10-09 16:39 +0000 [6fc4df7279]  Kevin Harwell <kharwell@digium.com>

	* res_rtp_asterisk: Crash if no candidates received for component

	  When starting ice if there is not at least one remote ice candidate with an RTP
	  component asterisk will crash. This is due to an assertion in pjnath as it
	  expects at least one candidate with an RTP component. Added a check to make
	  sure at least one candidate contains an RTP component and at least one candidate
	  has an RTCP component.

	  ASTERISK-24383 #close
	  Review: https://reviewboard.asterisk.org/r/4039/
	  ........

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2014-10-09 15:55 +0000 [c6837c236f]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov

	  This module allows res_pjsip to integrate with res_phoneprov.  It handles
	  the pjsip 'phoneprov' object type.

	  Tested-by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3976/
	  ........

	  Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-10-09 13:44 +0000 [3a187aa14a]  Matt Jordan <mjordan@digium.com>

	* res/res_phoneprov: Don't cancel Asterisk load on module load failure
	  ........

	  Merged revisions 424985 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-10-09 12:46 +0000 [cc595f7353]  gtjoseph <george.joseph@fairview5.com>

	* res_phoneprov: Refactor phoneprov to allow pluggable config providers

	  This patch makes res_phoneprov more modular so other modules (like pjsip)
	  can provide configuration information instead of res_phoneprov relying solely
	  on users.conf and sip.conf.  To accomplish this a new ast_phoneprov public API
	  is now exposed which allows config providers to register themselves, set
	  defaults (server profile, etc) and add user extensions.

	  * ast_phoneprov_provider_register registers the provider and provides callbacks
	    for loading default settings and loading users.
	  * ast_phoneprov_provider_unregister clears the defaults and users.
	  * ast_phoneprov_add_extension should be called once for each user/extension
	    by the provider's load_users callback to add them.
	  * ast_phoneprov_delete_extension deletes one extension.
	  * ast_phoneprov_delete_extensions deletes all extensions for the provider.

	  Tested-by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3970/
	  ........

	  Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-10-09 11:38 +0000 [0f50e8856b]  Richard Mudgett <rmudgett@digium.com>

	* cdr.c: Make turning on CDR debug a one step process instead of two.

	  Now "cdr set debug on" doesn't also require "core set verbose 1" to see
	  CDR debug output.
	  ........

	  Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-10-09 03:10 +0000 [d0255c4a46]  Michael Myles (License #6626)

	* safe_asterisk: Don't automatically exceed MAXFILES value of 2^20.

	  On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided
	  by two can exceed the per-process file limit of 2^20. This patch
	  ensures the value is capped.

	  (Patch cleaned up by me.)

	  ASTERISK-24011 #close
	  Reported by: Michael Myles
	  Patches:
	    safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626)
	  ........

	  Merged revisions 424875 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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	  ........

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2014-10-08 13:47 +0000 [8b0089ea1d]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Allow only UDP ICE candidates.

	  The underlying library, pjnath, that res_rtp_asterisk uses for ICE
	  support does not have support for ICE-TCP. As candidates are
	  passed through directly to it this can cause error messages to occur
	  when it receives something unexpected (such as a TCP candidate).
	  This change merely ignores all non-UDP candidates so they never
	  reach pjnath.

	  ASTERISK-24326 #close
	  Reported by: Joshua Colp
	  ........

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2014-10-08 13:24 +0000 [5e50638539]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Relegate log message to dev-mode

	  This error message primarily applies to development tasks and will now
	  only show up when dev-mode is enabled via configure.
	  ........

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2014-10-08 09:54 +0000 [3dfc485e35]  Kinsey Moore <kmoore@digium.com>

	* Indexer: Format message types may not exist

	  In Asterisk 13+, any given message type is not guaranteed to exist even
	  if Asterisk comes up correctly since creation of the message type could
	  be declined. The indexer should not prevent Asterisk from starting
	  under these conditions.
	  ........

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2014-10-07 15:33 +0000 [d8bbf1ec1d]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Only log errors for non-declined types

	  When message type creation is declined via stasis.conf, certain
	  operations log errors assuming that the declined type is being used
	  before initialization or after destruction. These error messages get
	  quite spammy for oft used message types and should not be logged in the
	  first place since the message type is validly NULL.

	  Reported by: Matt DiMeo
	  ........

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2014-10-07 13:34 +0000 [f7225da08a]  Joshua Colp <jcolp@digium.com>

	* data: Properly access formats in capabilities structure when adding codecs.

	  Formats within a capabilities structure are addressed starting at 0, not 1.
	  Assuming 1 causes it to exceed an array.

	  ASTERISK-24389 #close
	  Reported by: Kevin Harwell
	  ........

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2014-10-07 12:44 +0000 [a9011106b6]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter

	  Prior to this patch, the auth_reject_permanent parameter was not initialized on
	  the registration client state, leading to the parameter being disabled
	  regardless of the value specified in pjsip.conf.

	  This patch initialized the setting on the registration client state to the
	  provided configuration value.

	  ASTERISK-24398 #close
	  ........

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	  ........

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2014-10-07 09:09 +0000 [523da7d1b3]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_pubsub: Fix typo in WARNING message
	  ........

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2014-10-06 13:39 +0000 [39bd5b7a70]  Peter Katzmann (License 5968)

	* message: Don't close an AMI connection on SendMessage action error

	  If SendMessage encounters an error (such as incorrect input provided to the
	  action), it will currently return -1. Actions should only return -1 if the
	  connection to the AMI client should be closed. In this case, SendMessage
	  causing the client to disconnect is inappropriate.

	  This patch causes the action to return 0, which simply causes the action to
	  fail.

	  Review: https://reviewboard.asterisk.org/r/4024

	  ASTERISK-24354 #close
	  Reported by: Peter Katzmann
	  patches:
	    sendMessage.patch uploaded by Peter Katzmann (License 5968)
	  ........

	  Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  ........

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2014-10-06 10:41 +0000 [c384532aa4]  Richard Mudgett <rmudgett@digium.com>

	* features.c: Fix lingering channel ref while Bridge() application is active.

	  Using the Bridge application to bridge a channel that is executing an
	  applicaiton such as Wait results in a lingering Surrogate channel in the
	  CLI "core show channels" output even though it has already hungup.

	  * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has
	  been put into the bridge.

	  * Eliminated bridge_exec()'s use of RAII_VAR().

	  ASTERISK-24224 #close
	  Reported by: Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/4041/
	  ........

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	  ........

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2014-10-06 07:39 +0000 [3a87f32dc0]  Matt Jordan <mjordan@digium.com>

	* sdp_srtp: Add new lines to some WARNING messages
	  ........

	  Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-10-05 20:01 +0000 [cce3d99ec8]  Matt Jordan <mjordan@digium.com>

	* res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint

	  An OPTIONS request that is sent to Asterisk but not to a specific endpoint is
	  currently sent a 404 in response. This is because, not surprisingly, an empty
	  extension is never going to be found in the dialplan.

	  This patch makes it so that we only attempt to look up the endpoint in the
	  dialplan if it is specified in the OPTIONS request URI.

	  #SIPit31

	  ASTERISK-24370 #close
	  Reported by: Matt Jordan
	  ........

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	  ........

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2014-10-05 19:53 +0000 [c013916869]  Matt Jordan <mjordan@digium.com>

	* pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels

	  Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
	  It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
	  FRACKing on assertion error, and quite likely crashing.

	  This patch adds checks to the read/write callbacks that ensure that the channel
	  technology is of type 'PJSIP' before attempting to operate on the channel.

	  #SIPit31

	  ASTERISK-24382 #close
	  Reported by: Matt Jordan
	  ........

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2014-10-05 19:31 +0000 [45b7b474ac]  Matt Jordan <mjordan@digium.com>

	* res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message

	  When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable
	  transport, it is possible (although it shouldn't occur) for pjproject to pass
	  up an rdata object with a NULL msg in the msg_info. Needless to say, things
	  that attempt to dereference this are in for a rough ride.

	  In particular, this caused crashes in three different locations, all of which
	  are 'low level' enough to intercept an rdata object early in processing:

	  (1) res_pjsip_logger
	  (2) res_hep_pjsip
	  (3) res_pjsip/distributor

	  Anything that can intercept an rdata object before res_pjsip/distributor should
	  be defensive when looking at the received packet.

	  #SIPit31

	  ASTERISK-24369 #close
	  Reported by: Matt Jordan
	  ........

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2014-10-05 19:13 +0000 [f27f41a288]  Matt Jordan <mjordan@digium.com>

	* res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions

	  A subscription that has been persisted can - for various reasons - fail to be
	  re-created on startup. This patch resolves a number of crashes that occurred
	  when a subscription cannot be re-created on several off-nominal paths.

	  #SIPit31

	  ASTERISK-24368 #close
	  Reported by: Matt Jordan
	  ........

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2014-10-04 19:49 +0000 [9611ef4f1e]  Corey Farrell <git@cfware.com>

	* Release AMI connections on shutdown.

	  ASTERISK-24378 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4037/
	  ........

	  Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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2014-10-04 19:15 +0000 [1b0902caa4]  Corey Farrell <git@cfware.com>

	* chan_motif: Correct last commit to use ao2_cleanup to free format cap

	  This fix applies to 13 and trunk.

	  ASTERISK-24384 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4043/
	  ........

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2014-10-04 19:02 +0000 [0cea12b9e8]  Corey Farrell <git@cfware.com>

	* chan_motif: Release format capabilities and config on module load error

	  ASTERISK-24384 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4043/
	  ........

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	  ........

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2014-10-03 16:58 +0000 [24ded9d9eb]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Fix XML typo and update CHANGES.

	  ASTERISK-24199
	  ........

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2014-10-03 14:42 +0000 [70301b0438]  Richard Mudgett <rmudgett@digium.com>

	* audiohooks: Reevaluate the bridge technology when an audiohook is added or removed.

	  Adding a mixmonitor to a channel causes the bridge to change technologies
	  from native to simple_bridge so the call can be recorded.  However, when
	  the mixmonitor is stopped the bridge does not switch back to the native
	  technology.

	  * Added unbridge requests to reevaluate the bridge when a channel
	  audiohook is removed.

	  * Moved the unbridge request into ast_audiohook_attach() ensure that the
	  bridge reevaluates whenever an audiohook is attached.  This simplified the
	  mixmonitor and chan_spy start code as well.

	  * Added defensive code to stop_mixmonitor_full() in case additional
	  arguments are ever added to the StopMixMonitor application.

	  * Made ast_framehook_detach() not do an unbridge request if the framehook
	  does not exist.

	  * Made ast_framehook_list_fixup() do an unbridge request if there are any
	  framehooks.  Also simplified the loop.

	  ASTERISK-24195 #close
	  Reported by: Jonathan Rose

	  Review: https://reviewboard.asterisk.org/r/4046/
	  ........

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2014-10-03 13:54 +0000 [cc11a78869]  Kristian Hogh

	* app_queue: Add dialplan function to get the channel name at the specified position in a queue.

	  The QUEUE_GET_CHANNEL function returns the caller's channel name at the
	  specified position in a queue.

	  QUEUE_GET_CHANNEL(<queuename>[,<position>])

	  The queue position parameter defaults to 1 if not specified.

	  Noop(${QUEUE_GET_CHANNEL(queuename, 2)})
	  "SIP/peer-00000002", if queue exist and have at least 2 callers

	  Noop(${QUEUE_GET_CHANNEL(queuename, 1)})
	  Noop(${QUEUE_GET_CHANNEL(queuename)})
	  "SIP/peer-00000000", if queue exist and have at least 1 caller

	  ASTERISK-24365 #close
	  Reported by: Kristian Hogh
	  Patches:
	        queue_get_firstchannel.patch (license #6639) patch uploaded by Kristian Hogh
	        rb4035.patch (license #6639) patch uploaded by Kristian Hogh
	        Patch morphed from QUEUE_GET_FIRSTCHANEL to the more general QUEUE_GET_CHANNEL
	        on reviewbord.

	  Review: https://reviewboard.asterisk.org/r/4035/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424493 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-10-03 12:47 +0000 [0165c5f95a]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip: Fix deadlock when masquerading PJSIP channels.

	  Performing a directed call pickup resulted in a deadlock when PJSIP
	  channels were involved.

	  A masquerade needs to hold onto the channel locks while it swaps channel
	  information between the two channels involved in the masquerade.  With
	  PJSIP channels, the fixup routine needed to push a fixup task onto the
	  PJSIP channel's serializer.  Unfortunately, if the serializer was also
	  processing a task that needed to lock the channel, you get deadlock.

	  * Added a new control frame that is used to notify the channels that a
	  masquerade is about to start and when it has completed.

	  * Added the ability to query taskprocessors if the current thread is the
	  taskprocessor thread.

	  * Added the ability to suspend/unsuspend the PJSIP serializer thread so a
	  masquerade could fixup the PJSIP channel without using the serializer.

	  ASTERISK-24356 #close
	  Reported by: rmudgett

	  Review: https://reviewboard.asterisk.org/r/4034/
	  ........

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2014-10-03 10:55 +0000 [4967478d18]  gtjoseph <george.joseph@fairview5.com>

	* sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function

	  When you call ast_sorcery_create() you don't necessarily know which wizard is
	  going to be invoked.  If it happens to be a wizard like 'config' that doesn't
	  have a 'create' virtual function you get a segfault in the
	  sorcery_wizard_create callback.  This patch catches the null function pointer,
	  does an ast_assert, and logs an error.

	  Review: https://reviewboard.asterisk.org/r/4044/
	  ........

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2014-10-03 08:59 +0000 [b1f8eba178]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Restore functional default for callerid_privacy

	  The pjsip config option default fixups from r424263 altered the
	  functional default from "allowed_not_screened" to "allowed". This
	  change restores the functional default value when none is provided.
	  ........

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2014-10-03 08:33 +0000 [4246652603]  Kinsey Moore <kmoore@digium.com>

	* Manager: Add missing fields and documentation for CoreShowChannels

	  This corrects some issues introduced in the responses to the
	  CoreShowChannels AMI command as well as adding documentation for the
	  responses. The command in Asterisk 12 was missing the following fields:
	  Duration, Application, ApplicationData, and BridgedChannel and
	  BridgedUniqueID (replaced with BridgeId).

	  ASTERISK-24262 #close
	  Reported by: Mitch Claborn
	  Review: https://reviewboard.asterisk.org/r/4040/
	  ........

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2014-10-02 16:55 +0000 [2b0777c017]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip: Make transport cipher option accept a comma separated list of cipher names.

	  Improvements to the res_pjsip transport cipher option.

	  * Made the cipher option accept a comma separated list of OpenSSL cipher
	  names.  Users of realtime will be glad if they have more than one name to
	  list.

	  * Added the CLI command 'pjsip list ciphers' so a user can know what
	  OpenSSL names are available for the cipher option.

	  * Updated the cipher option online XML documentation to specify what is
	  expected for the value.

	  * Updated pjsip.conf.sample to not indicate that ALL is acceptable since
	  ALL does not imply a preference order for the ciphers and PJSIP does not
	  simply pass the string to OpenSSL for interpretation.

	  ASTERISK-24199 #close
	  Reported by: Joshua Colp

	  Review: https://reviewboard.asterisk.org/r/4018/
	  ........

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2014-10-02 15:23 +0000 [b15cd42b5b]  Jonathan Rose <jrose@digium.com>

	* Alembic: Add enumerator value to sippeers -> directmedia - 'outgoing'

	  The 'outgoing' value was left off of the enumerator when first creating the
	  column. This patch adds it, and should gracefully upgrade keeping the existing
	  data in tact.

	  ASTERISK-23781 #close
	  Reported by: Stephen More
	  Review: https://reviewboard.asterisk.org/r/4013/
	  ........

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2014-10-02 10:33 +0000 [2f570094b7]  Jonathan Rose <jrose@digium.com>

	* chan_pjsip: Fix an assertion for channels that lack formats on creation

	  ASTERISK-24222 #close
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4017/
	  ........

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2014-10-02 08:36 +0000 [aa5458d6ab]  Scott Griepentrog <sgriepentrog@digium.com>

	* res_pjsip: document use of rewrite_contact in sample conf

	  Without setting rewrite_contact, an invite to an endpoint
	  behind NAT will not reach it - unless the endpoint itself
	  uses STUN or TURN to discover it's public URI.  Thus, the
	  use of this should be in the sample documentation.

	  Review: https://reviewboard.asterisk.org/r/4036/
	  ........

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2014-10-01 15:37 +0000 [a752ca00bd]  Corey Farrell <git@cfware.com>

	* res_hep: Release allocation reference to configuration.

	  ASTERISK-24362 #close
	  Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4026/
	  ........

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2014-10-01 11:39 +0000 [adba2a8d7f]  Joshua Colp <jcolp@digium.com>

	* res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.

	  During the latest update to DTLS-SRTP support the ability to configure
	  the hash used for fingerprints was added. This gave us two supported ones:
	  SHA-1 and SHA-256. The default was accordingly updated to SHA-256.
	  Unfortunately this configuration ability was not exposed within res_pjsip.
	  This change adds a dtls_fingerprint option that controls it.

	  #SIPit31
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2014-10-01 11:20 +0000 [9233b1cf44]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session.

	  #SIPit31
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2014-10-01 07:28 +0000 [4d2c7c23f8]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Handle defaults properly

	  This updates the code behind PJSIP configuration options with custom
	  handlers to deal with the assigned default values properly where it
	  makes sense and adjusting the default value where it doesn't. Before
	  applying this patch, there were several cases where the default value
	  for an option would prevent that config section from loading properly.

	  Reported by: Thomas Thompson
	  Review: https://reviewboard.asterisk.org/r/4019/
	  ........

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2014-10-01 07:15 +0000 [122cc050d0]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Force transport on contact rewrite

	  If contact rewriting is enabled but the contact differs in transport
	  from what is actually being used, messages after the initial INVITE
	  transaction can be sent to an incorrect transport/port combination. In
	  the case where this bug occurred the remote party never received a BYE
	  since it was sent to the remote party's TCP port over UDP.

	  Review: https://reviewboard.asterisk.org/r/4032/
	  ........

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2014-10-01 05:10 +0000 [c3a7524457]  ibercom <ibercom123@gmail.com> (License #6599)

	* chan_sip: Simplify some unref code by removing unlink_peer_from_tables.

	  ASTERISK-22945 #related
	  Reported by: ibercom
	  Patches:
	    asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599)
	  ........

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2014-10-01 04:55 +0000 [841d978a30]  ibercom <ibercom123@gmail.com> (License #6599)

	* chan_sip: Remove excess ref of realtime peer before sip_poke_peer.

	  The peer is referenced at the end of sip_poke_peer, it should not get
	  an extra ref before the call to sip_poke_peer. This fixes a memory
	  leak.

	  ASTERISK-22945 #close
	  Reported by: ibercom
	  Tested by: Yuriy Gorlichenko
	  Patches:
	    asterisk11.patch uploaded by ibercom (License #6599)

	  Review: https://reviewboard.asterisk.org/r/4031/
	  ........

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2014-09-30 06:42 +0000 [d7c29885ad]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Don't place an extra whitespace before 'rport' and don't put IPv6 addresses in brackets.

	  #SIPit31
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2014-09-30 06:36 +0000 [3641ebcf96]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP.

	  This change fixes an issue where ICE candidates put into the SDP did not contain
	  the 'raddr' and 'rport' information for server reflexive and relay candidates.

	  #SIPit31
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2014-09-29 17:00 +0000 [27396a6b59]  gtjoseph <george.joseph@fairview5.com>

	* pjsip_cli: Suppress header print on error or no objects

	  If there's an error on the pjsip command line or there are no objects, don't
	  print the column headers.

	  ASTERISK-24350 #close
	  Reported-by: Brad Latus
	  Tested-by: George Joseph
	  Tested-by: Brad Latus

	  Review: https://reviewboard.asterisk.org/r/4025/
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2014-09-29 16:32 +0000 [b56dfb78c5]  Walter Doekes <walter+asterisk@wjd.nu>

	* autosupport: Fix bashism.

	  '==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
	  'case' works better there.

	  Originally committed in r375059 and r375060 on 2012-10-16 21:13:08.

	  ASTERISK-20567 #close
	  Reported by: Tzafrir Cohen
	  ........

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2014-09-29 16:18 +0000 [270932635d]  Richard Mudgett <rmudgett@digium.com>

	* Simplify UUID generation in several places.

	  Replace code using ast_uuid_generate() with simpler and faster code using
	  ast_uuid_generate_str().  The new code avoids a malloc(), free(), and
	  copy.
	  ........

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2014-09-29 15:28 +0000 [9d2bc0675a]  Richard Mudgett <rmudgett@digium.com>

	* threadpool.c: Minor cleanup fixes.

	  * Fix threadpool_alloc() prototype.

	  * Add missing off-nominal NULL check of pool in threadpool_alloc().

	  * searializer_create() does not need to create the object with a lock as
	  the lock is not used.
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2014-09-27 12:29 +0000 [2eef53c465]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_session: Reduce SDP size by removing duplicate connection lines.

	  Due to the architecture of how media streams are handled each individual
	  handler adds connection details (IP address) for it. The first media stream
	  is then used as the top level SDP connection line. In practice each
	  line ends up being the same so to reduce the SDP size stream-level connection
	  information is also added to the SDP if it differs from the top level SDP
	  connection line.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424077 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-27 07:44 +0000 [76744543b4]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_session: Add additional checks for delaying session refreshes.

	  There are certain situations which no checks existed for which need to prevent
	  session refreshes. This includes sending a session refresh with SDP before SDP
	  negotiation has completed and sending a session refresh before the dialog itself
	  has been established. Checks for these have been added.

	  Additionally COLP related UPDATEs were including SDP when it is not needed.

	  Review: https://reviewboard.asterisk.org/r/4008/
	  ........

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2014-09-26 10:51 +0000 [3c1804eb0d]  Richard Mudgett <rmudgett@digium.com>

	* format_mp3: Made the get script conditionally apply patch if not already there.

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424039 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-26 10:43 +0000 [e0abb82ab8]  Walter Doekes <walter+asterisk@wjd.nu>

	* core: Ouch, forgot to undo a test free() in r423978.


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2014-09-26 10:28 +0000 [d07b9af24b]  Jeremy Laine

	* res_fax: Fix out of bounds error in update_modem_bits().

	  ASTERISK-24357 #close
	  Reported by: Jeremy Laine
	  Patches:
	        res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine
	  	  Modified patch to not use magic numbers.
	  ........

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2014-09-26 09:41 +0000 [37179a2b1f]  Walter Doekes <walter+asterisk@wjd.nu>

	* core: Don't allow free to mean ast_free (and malloc, etc..).

	  This gets rid of most old libc free/malloc/realloc and replaces them
	  with ast_free and friends. When compiling with MALLOC_DEBUG you'll
	  notice it when you're mistakenly using one of the libc variants. For
	  the legacy cases you can define WRAP_LIBC_MALLOC before including
	  asterisk.h.

	  Even better would be if the errors were also enabled when compiling
	  without MALLOC_DEBUG, but that's a slightly more invasive header
	  file change.

	  Those compiling addons/format_mp3 will need to rerun
	  ./contrib/scripts/get_mp3_source.sh.

	  ASTERISK-24348 #related
	  Review: https://reviewboard.asterisk.org/r/4015/


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2014-09-26 03:26 +0000 [b8c1130ed1]  Jeremy Lainé (License #6561)

	* docs: Escape unescaped minus sign in asterisk.8 manpage.

	  ASTERISK-23768 #close
	  Reported by: Jeremy Lainé
	  Patches:
	    escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561)
	  ........

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2014-09-25 16:03 +0000 [fa0c33ebc1]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip.c: Add missing off nominal cleanup in ast_sip_push_task_synchronous().

	  * Made memset the std struct in ast_sip_push_task_synchronous() because if
	  DEBUG_THREADS is enabled then uninitialized lock tracking data is used.
	  ........

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2014-09-25 15:49 +0000 [d172d84fe1]  Kristian Høgh (License #6639)

	* musiconhold: Add preferchannelclass=no option to prefer app class.

	  The new option 'preferchannelclass' is added to musiconhold.conf. If yes
	  (the default) the CHANNEL(musicclass) is preferred when choosing the
	  hold music. If it is no, the class suggested by the application that
	  calls the MoH (e.g. the Queue() app) gets preferred (new behaviour).

	  This way you set a different hold-music from the Queue-music by setting
	  both the CHANNEL(musicclass) and the queue-context musicclass.

	  ASTERISK-24276 #close
	  Reported by: Kristian Høgh
	  Patches:
	    app_override_channel_moh.patch uploaded by Kristian Høgh (License #6639)

	  Review: https://reviewboard.asterisk.org/r/4010/


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2014-09-24 13:35 +0000 [68077634fe]  Richard Mudgett <rmudgett@digium.com>

	* pjsip_options.c: Fix race condition stopping periodic out of dialog OPTIONS request.

	  The crash on the issues is a result of an invalid transport configuration
	  change when asterisk is restarted.  The attempt to send the qualify
	  request fails and we cleaned up.  However, the callback is also called
	  which results in a double unref of the objects involved.

	  * Put a wrapper around pjsip_endpt_send_request() to detect when the
	  passed in callback is called because of an error so callers can know to
	  not cleanup.

	  * Made send_request_cb() able to handle repeated challenges (Up to 10).

	  * Fix periodic endpoint qualify OPTIONS sched deletion race by avoiding
	  it.  The sched entry will no longer self stop and must be externally
	  stopped.

	  * Added REF_DEBUG description tags to struct sched_data in
	  pjsip_options.c.

	  * Fix some off-nominal ref leaks in schedule_qualify(),
	  qualify_and_schedule().

	  * Reordered pjsip_options.c module start/stop code to cleanup better on
	  error.

	  ASTERISK-24295 #close
	  Reported by: Rogger Padilla

	  Review: https://reviewboard.asterisk.org/r/3954/
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2014-09-24 03:55 +0000 [39fada4dc9]  Walter Doekes <walter+asterisk@wjd.nu>

	* chan_sip: Unref outbound proxy structure on dialog/pvt destruction.

	  Make sure outbound proxy refs are always unreffed on dialog destruction.

	  Review: https://reviewboard.asterisk.org/r/4016/
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2014-09-23 09:36 +0000 [a89964a510]  Mark Michelson <mmichelson@digium.com>

	* Make CDR and CEL unit tests less FRACKy.

	  Prior to this commit, CDR and CEL tests were expected to trigger
	  FRACKs (i.e. assertions) due to the fact that the channels they
	  create have no formats on them. Some code was independently added
	  recently that attempts to prevent FRACKs from occurring by failing
	  early when attempting to set up translation paths if one or both
	  channels support no formats. Unfortunately, this attempt to be helpful
	  made the CDR and CEL tests go from simply FRACKing to outright
	  failing and in some cases, failing so badly as to crash Asterisk.

	  This commit seeks to correct past mistakes by adding the ulaw format
	  to channels created by the CDR and CEL unit tests. This makes setting
	  up translation paths succeed, eliminates previously-seen FRACKs, and
	  ultimately causes the unit tests to succeed again.

	  Review: https://reviewboard.asterisk.org/r/4014
	  ........

	  Merged revisions 423783 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423784 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-22 14:49 +0000 [593455621b]  Torrey Searle (License #5334)

	* chan_sip: On INVITE retransmission, don't add an extra 503 response.

	  INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is
	  retransmitted, asterisk would generate a 503 in addition to the 486.

	  Thanks Torrey Searle for providing a working regression test.

	  ASTERISK-24335 #close

	  Review: https://reviewboard.asterisk.org/r/4003/
	  Patches:
	    retrans_486_invite.patch uploaded by Torrey Searle (License #5334)
	  ........

	  Merged revisions 423720 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 423721 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 423722 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423723 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-22 12:42 +0000 [63a4da4a0d]  Walter Doekes <walter+asterisk@wjd.nu>

	* cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled.

	  r421600 conflicted with r155763.

	  ASTERISK-24348 #close
	  ........

	  Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 423658 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 423659 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423660 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-20 20:16 +0000 [64a9e5f001]  Matt Jordan <mjordan@digium.com>

	* main/channel: Unlock channel in off-nominal path

	  In r423414 (13) / r423415 (trunk), an API call that determines if a format
	  capability structure is empty was added. This returns true if the format
	  capability structure is completely empty or "none". A check for this was added
	  in channel.c's set_format call. Unfortunately, when this check was true, it
	  returned from the function while still holding the channel lock. This caused
	  the CDR unit tests - which have a tendency to create channels with no formats -
	  to deadlock. Whoops.

	  This patch unlocks the channel on the off-nominal path.
	  ........

	  Merged revisions 423641 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423642 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-20 18:55 +0000 [9bf039346a]  Matt Jordan <mjordan@digium.com>

	* rest-api/api-docs/events.json: Remove non-compliant 'extends' attribute

	  Prior to the release of Swagger 1.2, the attribute 'extends' was being
	  promoted as a possible way to show that a particular object extends an existing
	  object. Instead, the Swagger specification went with the 'subTypes' attribute
	  in the base object. This patch removes the unsupported attribute; the object
	  that the offending objects proposed to extend already lists them in its
	  'subTypes' attribute.

	  ASTERISK-24300 #close
	  Reported by: Bradley Watkins
	  ........

	  Merged revisions 423620 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423621 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-20 18:41 +0000 [de6e467db7]  Matt Jordan <mjordan@digium.com>

	* rest-api/api-docs: Correct basePath in resources to match top resources file

	  The resources.json file that defines the resource JSON files used with ARI
	  references a basePath of 'http://localhost:8088/ari'. This does not match what
	  is defined in the resource files themselves, 'http://localhost:8088/stasis'.
	  The correct base path is the one that includes 'ari' in the URL; this patch
	  updates the various resource JSON files to have the correct basePath.

	  ASTERISK-24339 #close
	  Reported by: Bradley Watkins
	  ........

	  Merged revisions 423617 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423618 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-19 14:51 +0000 [354fff327d]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_notify: Fix crash on unload/load and don't say the module doesn't exist on reload.

	  When unloading the module did not unregister the CLI commands causing a crash upon
	  load when they were registered again.

	  When reloading the module the return value from the config options framework was not
	  checked to determine if an error occurred or not. This caused a message to be output
	  saying the module did not exist when reloading if no changes were present.

	  AST-1433 #close
	  AST-1434 #close
	  ........

	  Merged revisions 423579 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423580 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-19 12:16 +0000 [ec0313c411]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.

	  Outgoing PJSIP calls can result in non-negotiated formats listed in the
	  channel's native formats if video formats are listed in the endpoint's
	  configuration.  The resulting call could then use a non-negotiated format
	  resulting in one way audio.

	  * Simplified the update of session->req_caps in set_caps().  Why do
	  something in five steps when only one is needed?

	  AFS-162 #close

	  Review: https://reviewboard.asterisk.org/r/4000/
	  ........

	  Merged revisions 423561 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-19 10:54 +0000 [6dae345674]  Jonathan Rose <jrose@digium.com>

	* Stasis_channels: Resolve unfinished Dials when doing masquerades

	  Masquerades into channels that are in the dialing state don't end their dial
	  and this goes against the model for things like CDRs and generating Dial end
	  manager actions and such.

	  ASTERISK-24237 #close
	  Reported by: Richard Mudgett
	  Review: https://reviewboard.asterisk.org/r/3990/
	  ........

	  Merged revisions 423525 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423530 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-19 10:11 +0000 [7e602175ff]  Jonathan Rose <jrose@digium.com>

	* chan_iax2: Fix a crash when using chan_iax2 jitterbuffer settings

	  Caused by format changes in Asterisk 13

	  ASTERISK-24265 #close
	  Reported by: Dafi Ni
	  Review: https://reviewboard.asterisk.org/r/3999/
	  ........

	  Merged revisions 423524 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-19 07:50 +0000 [7f2623a26f]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Prevent T38 framehook being put on wrong channel

	  This change gives framehooks a reverse-direction masquerade callback in
	  addition to chan_fixup_cb similar to the callback added to datastores
	  to handle the same situation. The new callback provides the same
	  parameters as the fixup callback, but is called on the new channel's
	  framehooks before moving framehooks from the old channel to the new
	  channel. This gives the framehooks an oppurtunity to decide whether
	  they should remain on the new channel or be removed.

	  This new callback is used to prevent the PJSIP T.38 framehook from
	  remaining on a masqueraded channel if the new channel is not also a
	  PJSIP channel. This was causing a crash when a local channel was
	  masqueraded into a PJSIP channel and the framehook was executed on the
	  local channel since the channel's tech private data was not structured
	  as expected.

	  Review: https://reviewboard.asterisk.org/r/4001/
	  ........

	  Merged revisions 423503 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423504 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423505 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 14:31 +0000 [40e033a6b6]  Sean Bright <sean@malleable.com>

	* res_pjsip: Don't require a password when doing userpass authentication.

	  An empty password is valid for username/password authentication so we should
	  allow password to be empty/not supplied.

	  Review: https://reviewboard.asterisk.org/r/3988
	  ........

	  Merged revisions 423481 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423482 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 14:23 +0000 [ad8ef9175a]  gtjoseph <george.joseph@fairview5.com>

	* utils: Create ast_strsep function that ignores separators inside quotes

	  This function acts like strsep with three exceptions...
	  * The separator is a single character instead of a string.
	  * Separators inside quotes are treated literally instead of like separators.
	  * You can elect to have leading and trailing whitespace and quotes
	  stripped from the result and have '\' sequences unescaped.

	  Like strsep, ast_strsep maintains no internal state and you can call it
	  recursively using different separators on the same storage.

	  Also like strsep, for consistent results, consecutive separators are not
	  collapsed so you may get an empty string as a valid result.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3989/
	  ........

	  Merged revisions 423476 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423478 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423480 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 13:56 +0000 [de72f3edbc]  Mark Michelson <mmichelson@digium.com>

	* Add subscription state test events.

	  These are needed for a set of batched notification RLS tests that are
	  about to be committed to the testsuite.

	  Review: https://reviewboard.asterisk.org/r/3967
	  ........

	  Merged revisions 423462 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423463 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 12:22 +0000 [ac46240b62]  Jonathan Rose <jrose@digium.com>

	* res_pjsip_endpoint_identifier_ip: Fix parsing of match value with CIDR

	  Also fixes comma separates match lists

	  ASTERISK-24290 #close
	  Reported by: Ray Crumrine
	  Review: https://reviewboard.asterisk.org/r/3995/
	  ........

	  Merged revisions 423417 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423425 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423442 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 12:10 +0000 [02cf1835e3]  Richard Mudgett <rmudgett@digium.com>

	* bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent.

	  * Clarified some read/write format comments.

	  * Fixed a doxygen tag typo.
	  ........

	  Merged revisions 423423 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423424 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 11:56 +0000 [a7add3a257]  Richard Mudgett <rmudgett@digium.com>

	* astobj2.c/refcounter.py: Fix to deal with invalid object refs.

	  * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2
	  object ref/unref is attempted.  This is similar to the
	  constructor/destructor lines.

	  * Fixed refcounter.py to handle skewed objects that have
	  constructor/destructor states.

	  * Made refcounter.py highlight the invalid ao2 object refs by putting them
	  in their own section of the processed output file.

	  * Made refcounter.py highlight unreffing an object by more than one that
	  results in a negative ref count and the object being destroyed.  The
	  abnormally destroyed object is reported in the invalid and finalized
	  object sections of the output.

	  Review: https://reviewboard.asterisk.org/r/3971/
	  ........

	  Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 423400 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423418 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423422 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 11:38 +0000 [fa6313ad29]  Mark Michelson <mmichelson@digium.com>

	* Add API call to determine if format capability structure is "empty".

	  Empty here means that there are no formats in the format_cap structure
	  or the only format in it is the "none" format.

	  I've added calls to check the emptiness of a format_cap in a few places
	  in order to short-circuit operations that would otherwise be pointless
	  as well as to prevent some assertions from being triggered in cases
	  where channels with no formats are used.
	  ........

	  Merged revisions 423414 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423415 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 11:24 +0000 [389db2b720]  Mark Michelson (License #5049)

	* res_fax_spandsp: Properly handle cleanup before starting FAXes.

	  If faxing fails at a very early stage, then it is possible for
	  us to pass a NULL t30 state pointer to spandsp, which spandsp
	  is none too pleased with.

	  This patch ensures that we pass the correct pointer to spandsp
	  in the situation where we have not yet set our local t30 state
	  pointer.

	  ASTERISK-24301 #close
	  Reported by Matt Jordan
	  Patches:
	  	ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049)
	  ........

	  Merged revisions 423360 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 423365 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423372 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423380 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 11:09 +0000 [79eac1ffca]  Mark Michelson <mmichelson@digium.com>

	* res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies.

	  res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE
	  arrives.
	  * It checks that there is a subscription handler for the Event
	  * It checks that there are body generators for the types in the Accept header

	  The problem is, there's nothing that ensures that these two things will
	  actually mesh with each other. For instance, Asterisk will accept a subscription
	  to MWI that accepts pidf+xml bodies. That doesn't make sense.

	  With this commit, we add some type information to the mix. Subscription
	  handlers state they generate data of type X, and body generators state
	  that they consume data of type X. This way, Asterisk doesn't end up in
	  some hilariously mismatched situation like the one in the previous paragraph.

	  ASTERISK-24136 #close
	  Reported by Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/3877
	  Review: https://reviewboard.asterisk.org/r/3878
	  ........

	  Merged revisions 423344 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423348 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 10:14 +0000 [126334a7aa]  gtjoseph <george.joseph@fairview5.com>

	* res_pjsip: ami: Fix error in AMI output when an endpoint has no transport

	  When no transport is associated to an endpoint, the AMI output for
	  PJSIPShowEndpoint indicates an error instead of silently ignoring the
	  missing transport.

	  This patch causes the error to appear only if a transport was specified
	  on the endpoint and the transport doesn't exist.  It also fixes an issue
	  with counting the objects that were actually found.

	  ASTERISK-24161 #close
	  ASTERISK-24331 #close
	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3998/
	  ........

	  Merged revisions 423282 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423284 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423285 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 10:07 +0000 [b89491e39c]  David M. Lee <dlee@digium.com>

	* Only install dahdi_span_config_hook if DAHDI is enabled

	  This patch changes the install to only install the hook script if
	  DAHDI is enabled. It also adds the script to the uninstall task, and
	  moves the DAHDI_UDEV_HOOK_DIR variable so that it's not between the
	  _MAKEOPTS variables and their comment.

	  This allows installs which specify a --prefix to work normally, as
	  long as they don't enable DAHDI.

	  Review: https://reviewboard.asterisk.org/r/3972/
	  ........

	  Merged revisions 423281 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423283 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-18 09:46 +0000 [d120e40309]  gtjoseph <george.joseph@fairview5.com>

	* config: bug: Fix SEGV in ast_category_insert when matching category isn't found

	  If you call ast_category_insert with a match category that doesn't exist, the
	  list traverse runs out of 'next' categories and you get a SEGV.  This patch
	  adds check for the end-of-list condition and changes the signature to return
	  an int for success/failure indication instead of a void.

	  The only consumer of this function is manager and it was also changed to use
	  the return value.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3993/
	  ........

	  Merged revisions 423276 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 423277 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 423278 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423279 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423280 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-17 13:06 +0000 [8839ba3727]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered.
	  ........

	  Merged revisions 423253 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 423254 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423255 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423256 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-16 17:46 +0000 [fcc09fd0de]  Matt Jordan <mjordan@digium.com>

	* pbx/Makefile: Revert r423237

	  This patch was supposed to go into a team branch, but I was a bit fast on the
	  gun before 'svn switch' had apparently moved the target branch over.


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-16 17:42 +0000 [712b4195ef]  Matt Jordan <mjordan@digium.com>

	* Add some pbx python stuff


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423237 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-16 16:06 +0000 [618b46d8f0]  Joshua Colp <jcolp@digium.com>

	* Multiple revisions 423209,423212

	  ........
	    r423209 | file | 2014-09-16 17:35:34 -0300 (Tue, 16 Sep 2014) | 8 lines
	    
	    res_rtp_asterisk: Fix building when pjproject is not used.
	    ........
	    
	    Merged revisions 423207 from http://svn.asterisk.org/svn/asterisk/branches/11
	    ........
	    
	    Merged revisions 423208 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........
	    r423212 | file | 2014-09-16 18:03:59 -0300 (Tue, 16 Sep 2014) | 10 lines
	    
	    res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning.
	    
	    Side note: I need a vacation.
	    ........
	    
	    Merged revisions 423210 from http://svn.asterisk.org/svn/asterisk/branches/11
	    ........
	    
	    Merged revisions 423211 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 423209,423212 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-16 11:33 +0000 [662b687dbe]  Scott Griepentrog <sgriepentrog@digium.com>

	* Voicemail: get correct duration when copying file to vm

	  Changes made during format improvements resulted in the
	  recording to voicemail option 'm' of the MixMonitor app
	  writing a zero length duration in the msgXXXX.txt file.

	  This change introduces a new function ast_ratestream(),
	  which provides the sample rate of the format associated
	  with the stream, and updates the app_voicemail function
	  for ast_app_copy_recording_to_vm to calculate the right
	  duration.

	  Review: https://reviewboard.asterisk.org/r/3996/
	  ASTERISK-24328 #close
	  ........

	  Merged revisions 423192 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-16 07:12 +0000 [ceedf44edd]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_session: Fix usage of wrong memory pool when creating local SDP.
	  ........

	  Merged revisions 423172 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-09-16 06:12 +0000 [e977425bc8]  Joshua Colp <jcolp@digium.com>

	* res_rtp_asterisk: Fix a myriad of TURN client issues.

	  1. The number of file descriptors an ioqueue instance can handle is fixed, so we
	  now spawn the required number to handle the load.
	  2. Our transport identifiers were exceeding the range supported by pjnath.
	  3. The TURN client did not set up client binding causing needless bandwidth usage.
	  4. The code no longer updates address information on each packet.
	  5. STUN traffic was getting looped back to Asterisk instead of going through the
	  TURN server.
	  6. Synchronization now ensures things are completely setup or destroyed.
	  7. Logging now reflects the target the TURN server is sending to/receiving from
	  on our behalf.

	  ASTERISK-23577 #close
	  Reported by: Jay Jideliov

	  ASTERISK-23634 #close
	  Reported by: Roman Skvirsky

	  Review: https://reviewboard.asterisk.org/r/3982/
	  ........

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	  ........

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2014-09-15 05:50 +0000 [77834b72d3]  Zogot, cleaned up by me.

	* contrib: Fix verifyi typo in alembic DB script ps_transport table.

	  Reported by: Zogot (on IRC)
	  Patches:
	    tmp.diff uploaded by Zogot, cleaned up by me.
	  ........

	  Merged revisions 423128 from http://svn.asterisk.org/svn/asterisk/branches/12
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2014-09-14 10:54 +0000 [a62fedf0cb]  Walter Doekes <walter+asterisk@wjd.nu>

	* chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI.

	  Document it in sip.conf.

	  ASTERISK-24249 #close
	  Reported by: Avinash Mohod

	  Review: https://reviewboard.asterisk.org/r/3926/
	  ........

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	  ........

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	  ........

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	  ........

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2014-09-14 10:41 +0000 [9c1f34c7e9]  Walter Doekes <walter+asterisk@wjd.nu>

	* musiconhold: Add sort=randstart, and deprecate old stuff.

	  - adds sort=randstart (next to sort=, sort=random, sort=alpha)
	  - combines duplicate moh option parsing code into a single function
	  - adds deprecationwarnings for application=r to sort randomly
	  - adds deprecationwarnings for random=yes to sort randomly
	  - removes invisible code that was supposed to stay until 1.8 

	  The sort=randstart works like sort=alpha, except we start at a random
	  position.

	  Review: https://reviewboard.asterisk.org/r/3991/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-12 12:42 +0000 [02295456ef]  Joshua Colp <jcolp@digium.com>

	* chan_rtp: Add unicast RTP support.

	  This module supports sending both unicast and multicast RTP
	  to a specified target. Multicast functionality is the same as
	  chan_multicast_rtp was. In the case of unicast a specific
	  IP address and port can be specified, along with optional RTP
	  engine and format in the form of:

	  UnicastRTP/<ip address>:<port>/<engine>/<format>

	  This can be useful for sending a copy of a media stream to
	  another application for processing.

	  Review: https://reviewboard.asterisk.org/r/3981/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-12 11:19 +0000 [dd6bdede7d]  Jonathan Rose <jrose@digium.com>

	* Realtime: Fix a bug that caused realtime destroy command to crash

	  Also has could affect with anything that goes through ast_destroy_realtime.
	  If a CLI user used the command 'realtime destroy <family>' with only a single
	  column/value pair, Asterisk would crash when trying to create a variable list
	  from a NULL value.

	  ASTERISK-24231 #close
	  Reported by: Niklas Larsson
	  Review: https://reviewboard.asterisk.org/r/3985/
	  ........

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	  ........

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2014-09-11 17:17 +0000 [c212a71f0b]  Mark Michelson <mmichelson@digium.com>

	* Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.

	  ast_play_and_record_full() has a parameter called "acceptdtmf" that is a
	  string of acceptable DTMF digits that may be pressed by a caller to end
	  and accept the recording.

	  ARI uses this function in order to perform recording, and it provides
	  options for what is passed as acceptdtmf to ast_play_and_record_full().
	  By default, ARI passes an empty string, with the intention that no DTMF
	  can be used to end the recording.

	  The problem is that ast_play_and_record_full() attempts to be "helpful"
	  by setting "#" as the acceptdtmf if an empty string or NULL pointer
	  has been passed in. With ARI, this results in unexpected behavior
	  occurring if you have attempted to intercept "#" yourself in order
	  to perform some other manipulation of the live recording.

	  This change removes the "helpful" behavior by no longer accepting
	  "#" as a default acceptdtmf if none is specified by the caller of
	  ast_play_and_record_full(). This makes the ARI scenario work as
	  expected.

	  The other callers of ast_play_and_record_full() are app_voicemail
	  and app_minivm, and in both cases, they pass an explicit "#" to
	  ast_play_and_record_full() as acceptdtmf, so they are unaffected
	  by this change.
	  ........

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2014-09-10 11:07 +0000 [93894d53c4]  gtjoseph <george.joseph@fairview5.com>

	* config: bug: fix truncation of included config files on permissions error

	  ast_config_text_file_save() currently truncates include files as they
	  are processed.  If a subsequent include file or the main config file has
	  a permissions error that prevents writing, earlier include files are left
	  truncated resulting in a frantic search for backups.

	  This patch causes ast_config_text_file_save to check for write access
	  on all files before it truncates any of them.

	  Will be applied 1.8 > trunk.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3986/
	  ........

	  Merged revisions 422900 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2014-09-10 11:00 +0000 [7bd3287a11]  Sean Bright <sean@malleable.com>

	* pjsip/config_auth.c: Add missing whitespace to log messages.

	  The errors generated when validating 'auth' settings are missing a space which
	  makes the messages a little confusing.
	  ........

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	  ........

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2014-09-09 15:15 +0000 [a47873168a]  Richard Mudgett <rmudgett@digium.com>

	* Update CHANGES for CHANNEL(onhold).

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422885 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-09-09 15:11 +0000 [51f082af34]  Rusty Newton <rnewton@digium.com>

	* Sounds/BuildSystem: Modifications to include new releases and Japanese language.

	  Modifying Makefile and sounds.xml to include new core 1.4.26 and extra 1.4.15
	  sound prompt releases, plus the new Japanese core sound prompts contributed
	  by QLOOG.

	  ASTERISK-23324
	  Reported by: Kevin McCoy
	  Tested by: Rusty Newton
	  ........

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	  ........

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	  ........

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2014-09-09 11:14 +0000 [9183416fe2]  Richard Mudgett <rmudgett@digium.com>

	* func_channel: Add CHANNEL(onhold) item to get the current hold status of the channel.

	  It would be useful to get the current hold status of a channel.

	  Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
	  the hold status of a channel.

	  ASTERISK-24038
	  Reported by: Matt Jordan

	  AFS-113 #close
	  Reported by: Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/3983/


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2014-09-08 13:04 +0000 [baf99dffac]  Mark Michelson <mmichelson@digium.com>

	* Add note about configuring list_items on a single line.
	  ........

	  Merged revisions 422855 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-09-08 12:53 +0000 [5ad0edacb6]  Mark Michelson <mmichelson@digium.com>

	* Add sample configuration for resource lists.

	  On review /r/3977, it was recommended to note in the
	  sample configuration about the size limitation for
	  resource lists. However, since there was no section in
	  the sample configuration at all for resource list
	  subscriptions, I decided to make a separate commit
	  where I have added the necessary sample configuration
	  as well as the size limitation warning.
	  ........

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2014-09-08 12:35 +0000 [c6bc44f700]  Mark Michelson <mmichelson@digium.com>

	* Pre-allocate transmission data buffer for RLS NOTIFY requests.

	  PJSIP, unless a constant is modified at compilation time, limits
	  SIP requests to 4000 bytes. Full-state RLS notifications can easily
	  exceed this limit with moderately small lists.

	  This changeset allows for Asterisk to work around this size limit by
	  performing its own allocation of the transmission data buffer. This
	  way, Asterisk can allocate a buffer that exceeds the built-in maximum.

	  We still impose our own limit of 64000 bytes, mainly because making
	  allocations larger than that is a bit absurd.

	  ASTERISK-24181 #close
	  Reported by Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/3977
	  ........

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2014-09-08 10:58 +0000 [ef5f7a0e32]  Jonathan Rose <jrose@digium.com>

	* res_pjsip_pubsub: Check supported headers for eventlist when subscribing to
	  resource list

	  https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan
	  According to the off-nominal plan, if evenlist support is not specified in a
	  SUBSCRIBE's supported header(s), that subscription should be rejected with an
	  error.

	  ASTERISK-23871
	  Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/3960/diff/#index_header
	  ........

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2014-09-06 17:50 +0000 [71acca4de2]  Matt Jordan <mjordan@digium.com>

	* main/cdr: Copy over location information during a fork

	  When a CDR is forked, a new CDR is created and appended to the CDR chain for
	  the Party A. The forked CDR starts life off as a clone of the last
	  non-finalized for the particular Party A. In the past, merely copying over
	  the snapshots for Party A/Party B would be sufficient. However, as the CDRs
	  now contain cached information from Party A - specifically application/data,
	  context, and extension - we need to copy that over during a fork as well.

	  Huzzah for unit tests catching this when the context/extension were derived
	  from a cached value on the CDR instead of on Party A.
	  ........

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2014-09-06 17:22 +0000 [e4591f98b1]  Matt Jordan <mjordan@digium.com>

	* main/rtp_engine: Format NTP timestamps as unsigned ints

	  On some systems, a timeval's tv_sec/tv_usec will be unsigned lont ints, as
	  opposed to long ints. When the RTP engine formats these as strings, it was
	  previously formatting them as signed integers, which can result in some
	  odd negative timestamp values (particularly on 32-bit systems). This patch
	  formats the values as unsigned long integers.
	  ........

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2014-09-06 14:13 +0000 [fd8010de2b]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media stream.
	  ........

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2014-09-05 17:04 +0000 [d42b116925]  Matt Jordan <mjordan@digium.com>

	* main/cdrs: Preserve context/extension when executing a Macro or GoSub

	  The context/extension in a CDR is generally considered the destination of a
	  call. When looking at a 2-party call CDR, users will typically be presented
	  with the following:

	  context    exten      channel     dest_channel app  data
	  default    1000       SIP/8675309 SIP/1000     Dial SIP/1000,,20

	  However, if the Dial actually takes place in a Macro, the current behaviour
	  in 12 will result in the following CDR:

	  context    exten      channel     dest_channel app  data
	  macro-dial s          SIP/8675309 SIP/1000     Dial SIP/1000,,20

	  The same is true of a GoSub:

	  context    exten      channel     dest_channel app  data
	  subs       dial_stuff SIP/8675309 SIP/1000     Dial SIP/1000,,20

	  This generally makes the context/exten fields less than useful.

	  It isn't hard to preserve these values in the CDR state machine; however, we
	  need to have something that informs us when a channel is executing a
	  subroutine. Prior to this patch, there isn't anything that does this.

	  This patch solves this problem by adding a new channel flag,
	  AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a
	  Macro or a GoSub. The CDR engine looks for this value when updating a Party A
	  snapshot; if the flag is present, we don't override the context/exten on the
	  main CDR object. In a funny quirk, executing a hangup handler must *not* abide
	  by this logic, as the endbeforehexten logic assumes that the user wants to see
	  data that occurs in hangup logic, which includes those subroutines. Since
	  those execute outside of a typical Dial operation (and will typically have
	  their own dedicated CDR anyway), this is unlikely to cause any heartburn.

	  Review: https://reviewboard.asterisk.org/r/3962/

	  ASTERISK-24254 #close
	  Reported by: tm1000, Tony Lewis
	  Tested by: Tony Lewis
	  ........

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2014-09-05 16:56 +0000 [4499eb05d8]  Matt Jordan <mjordan@digium.com>

	* main/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenarios

	  This patch fixes an issue where CDRs would get stuck generating an infinite
	  number of CDRs, eventually crashing Asterisk (and consuming a lot of memory
	  along the way).

	  When a channel enters into a multi-party bridge, the CDR engine creates
	  mappings of each participant to each other participant, picking the 'A' party
	  as it goes. So, if we have four channels in a multi-party bridge (Alice, Bob,
	  Charlie, Denise), we would have something like:

	  Alice => Bob
	  Alice => Charlie
	  Alice => Denise
	  Bob => Charlie
	  Bob => Denise
	  Charlie => Denise

	  This works fine when participants enter the bridge a single time.

	  When a participant leaves a bridge, the CDRs for that channel are transitioned
	  to a finalized state.

	  The bug occurs if Bob rejoins. When the CDR engine creates mappings between the
	  channels, it walks through all the participants currently in the bridge, and
	  realizes that no one in the bridge can create a CDR with the channel (Bob).
	  As such it creates a new CDR for the candidate and appends it to that
	  candidate's chain. Unfortunately, on this particular code path, it doesn't
	  stop traversing the candidate's chain. Since we just added ourselves to the
	  chain, this causes the loop to keep going, constantly adding new CDRs.

	  This patch makes it so the engine bails when it creates a CDR match in this
	  case.

	  Review: https://reviewboard.asterisk.org/r/3964/

	  ASTERISK-24241 #close
	  Reported by: Deepak Singh Rawat
	  Tested by: Deepak Singh Rawat

	  ASTERISK-24208
	  Reported by: Frankie Chin
	  ........

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2014-09-05 15:38 +0000 [025bd1bf3f]  Richard Mudgett <rmudgett@digium.com>

	* func_channel.c: Add missing locking to some CHANNEL() requests.

	  * The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and
	  audiowriteformat now need locking since the media format rework when
	  accessing the channel's format pointers.

	  * Increased the buffer size for CHANNEL() audionativeformat and
	  videonativeformat output strings since the allow=all can be a lengthy
	  list.

	  * Tweaked the CHANNEL() XML documentation for secure_bridge_signaling,
	  secure_bridge_media, and state.

	  * Ensured the output buffer is initialized for secure_bridge_signaling and
	  secure_bridge_media.

	  * Made use the locked_copy_string() macro instead of inlining it for trace
	  and checkhangup.
	  ........

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2014-09-05 15:22 +0000 [85878c4dd8]  Jonathan Rose <jrose@digium.com>

	* Dial API: Add a dial option to indicate the dialed channel will replace dialer

	  Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes.

	  Review: https://reviewboard.asterisk.org/r/3968/
	  ........

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2014-09-05 14:39 +0000 [e19017fc00]  Jonathan Rose <jrose@digium.com>

	* Call IDs: Fix appearance of call ID in core show channels when NULL

	  NULL call IDs were meant to appear as '(none)' but instead were showing
	  the contents of an uninitialized character buffer.

	  ASTERISK-24223
	  Review: https://reviewboard.asterisk.org/r/3979/
	  ........

	  Merged revisions 422664 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-09-05 12:45 +0000 [5a1de68b9a]  Richard Mudgett <rmudgett@digium.com>

	* devicestate.c: Minor tweaks

	  * In ast_state_chan2dev() use ARRAY_LEN() instead of a sentinel value in
	  chan2dev[].

	  * Fix some comments in chan_iax2.c.
	  ........

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2014-09-05 08:29 +0000 [2362d88a18]  Kinsey Moore <kmoore@digium.com>

	* Menuselect: Fix incorrect enabling on failed deps

	  This corrects a situation where menuselect can incorrectly enable a
	  module by default that has defaultenabled set to "no" and has
	  failed/non-selected dependencies. The bug is due to an inverted test
	  when checking for whether the given module should be set to enabled by
	  default on load.

	  Review: https://reviewboard.asterisk.org/r/3975/
	  Reported by: John Bigelow
	  ........

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2014-09-04 17:05 +0000 [af75e45da1]  Jonathan Rose <jrose@digium.com>

	* Manager: Require read permission for SYSTEM in order to send FullyBooted

	  Review: https://reviewboard.asterisk.org/r/3969/
	  ........

	  Merged revisions 422584 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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	  ........

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2014-09-03 09:05 +0000 [3cd36d0e10]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI.

	  The code for changing the Contact header wrongly assumed that the Contact
	  would always contain a URI. This is incorrect.

	  ASTERISK-24271
	  Reported by: Dafi Ni
	  ........

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	  ........

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2014-09-02 15:29 +0000 [1b64f353f1]  Mark Michelson <mmichelson@digium.com>

	* Resolve race condition where channels enter dialplan application before media has been negotiated.

	  Testsuite tests will occasionally fail because on reception of a 200 OK SIP response,
	  an AST_CONTROL_ANSWER frame is queued prior to when media has finished being
	  negotiated. This is because session supplements are called into before PJSIP's
	  inv_session code has told us that media has been updated. Sometimes the queued answer
	  frame is handled by the PBX thread before the ensuing media negotiations occur, causing
	  a test failure.

	  As it turns out, there is another place that session supplements could be called into, which is
	  after media has finished getting negotiated. What this commit introduces is a means for session
	  supplements to indicate when they wish to be called into when handling an incoming SIP response.
	  By default, all session supplements will be run at the same point that they were prior to this
	  commit. However, session supplements may indicate that they wish to be handled earlier than
	  normal on redirects, or they may indicate they wish to be handled after media has been negotiated.

	  In this changeset, two session supplements have been updated to indicate a preference for when
	  they should be run: res_pjsip_diversion executes before handling redirection in order to get
	  information from the Diversion header, and chan_pjsip now handles responses to INVITEs after
	  media negotiation to fix the race condition mentioned previously.

	  ASTERISK-24212 #close
	  Reported by Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/3930
	  ........

	  Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-09-01 10:25 +0000 [897cbf6a4f]  Matt Jordan <mjordan@digium.com>

	* main/cli: Do not attempt to show CDR data for internal channels

	  Internal channels don't have CDRs. Querying the CDR engine for their variables
	  will make it cranky.
	  ........

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	  ........

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2014-09-01 09:15 +0000 [df5dbbd878]  Matt Jordan <mjordan@digium.com>

	* res_stasis: Don't play MoH to channels by default when added to holding bridges

	  When ARI manipulates a bridge, it generally doesn't care what the mixing
	  technology is. Operations on a bridge initiated through ARI should perform
	  their action in generally the same way, regardless of the bridge's mixing
	  technology. While the mixing technology may determine how media flows to
	  channels, the actual operations on a bridge themselves should be the same.

	  Currently, this isn't the case with holding bridges. When a channel joins
	  without a role, MoH is started on that channel automatically. Subsequent bridge
	  operations that would stop MoH would fail (as there is no Announcer channel
	  playing MoH to the bridge). Starting MoH on the bridge will also create two
	  MoH streams: one from the MoH being played on the participant channel, and one
	  from the announcer channel. From the perspective of ARI users, this is
	  counter-intuitive - I would not expect MoH to be started for me. The mixing
	  technology determines how media is shared between participants, not the
	  application experience.

	  This patch does the following:
	   * The Stasis bridge class now inspects channels as they are going into a
	     bridge. If the bridge has a holding capability, and the channel has no
	     roles, we give it a participant role and mark the default behaviour to have
	     no entertainment. This allows addChannel operations to continue to set a
	     participant role with an entertainment option if it felt like it (or could
	     do it).
	   * The music on hold channel is now Stasis approved (tm)

	  Review: https://reviewboard.asterisk.org/r/3929/

	  ASTERISK-24264 #close
	  Reported by: Samuel Galarneau
	  Tested by: Samuel Galarneau 
	  ........

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2014-08-30 12:33 +0000 [5aefecd81e]  gtjoseph <george.joseph@fairview5.com>

	* confbridge: Add Duration to ConfbridgeList event

	  The ConfbridgeList event doesn't include how long the user has been a
	  member of the conference.  This patch adds Duration (seconds) which
	  is based on user->chan->answertime.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3955/
	  ........

	  Merged revisions 422444 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-30 12:24 +0000 [59d4dbd3d0]  gtjoseph <george.joseph@fairview5.com>

	* manager: Make WaitEvent action respect eventfilters

	  A WaitEvent issued via an http session isn't respecting eventfilters defined
	  for the user. I just added a match_filter to the predicate that controls
	  astman_append.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3958/
	  ........

	  Merged revisions 422439 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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2014-08-29 14:40 +0000 [664f83a03b]  Jeremy Laine (License 6561)

	* doc: Add a manpage for the smsq utility

	  This patch adds a manpage for the smsq utility. Note that this is one of
	  the patches the Debian distro applies for the Asterisk project, as per
	  ASTERISK-24191.

	  Review: https://reviewboard.asterisk.org/r/3895/

	  ASTERISK-24171 #close
	  Reported by: Jeremy Laine
	  patches:
	    smsq.8 uploaded by Jeremy Laine (License 6561)
	  ........

	  Merged revisions 422376 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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	  ........

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	  ........

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2014-08-29 14:35 +0000 [81598fa082]  Jeremy Laine (License 6561)

	* doc: Add a manpage for the aelparse utility

	  This patch adds a manpage for the aelparse utility. Note that this is one of
	  the patches the Debian distro applies for the Asterisk project, as per
	  ASTERISK-24191.

	  Review: https://reviewboard.asterisk.org/r/3896/

	  ASTERISK-24171 #close
	  Reported by: Jeremy Laine
	  patches:
	    aelparse.8 uploaded by Jeremy Laine (License 6561)
	  ........

	  Merged revisions 422371 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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	  ........

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	  ........

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2014-08-29 13:46 +0000 [2df2d785b7]  Scott Griepentrog <sgriepentrog@digium.com>

	* The assertion that peer was not found on final event
	  message was being triggered on configuration reload.
	  This patch changes that case to just return instead.

	  Review: https://reviewboard.asterisk.org/r/3953/



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2014-08-28 16:54 +0000 [3194892ea2]  Matt Jordan <mjordan@digium.com>

	* LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP

	  The UniMRCP project distributes Asterisk modules that integrate Asterisk with
	  UniMRCP, and other Asterisk users use the UniMRCP library as well.
	  Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software
	  Foundation, is not a compatible license with the GPLv2.

	  "Please note that this license is not compatible with GPL version 2, because it
	  has some requirements that are not in that GPL version. These include certain
	  patent termination and indemnification provisions. The patent termination
	  provision is a good thing, which is why we recommend the Apache 2.0 license for
	  substantial programs over other lax permissive licenses."

	  On the other hand, UniMRCP is a great project and we'd like to let people use
	  it with Asterisk.

	  This patch updates the LICENSE text to allow users to link Asterisk with
	  UniMRCP and distribute the resulting binaries.
	  ........

	  Merged revisions 422293 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2014-08-28 15:31 +0000 [c5916fb39f]  Michael L. Young (license 5026)

	* chan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure

	  The reporter on the issue found some issues when upgrading from version 10 to 11
	  on 55 hosts.

	  Two situations that can occur with dynamic registrations.

	  1.  With dnsmgr disabled, if the host is not resolvable we are not trying to
	      resolve the host again when it is time to attempt to register again.  This
	      results in never registering to the host.
	  2.  With dnsmgr enabled, when the host is temporarily not resolvable the
	      address is set to 0.0.0.0:0 and then when the host is resolvable the port
	      is not being restored and stays set to 0.

	  This patch resolves these two issues by:

	  * Storing the hostname so that it can be used for resolving with DNS.
	  * Resolve the hostname on the next scheduled attempt to register.
	  * Storing the port used to reach the host so that when the hostname is
	    resolvable again, we can set the port again if the port is still unset after
	    looking up the host.

	  ASTERISK-23767 #close
	  Reported by: David Herselman
	  Tested by: David Herselman, Michael L. Young
	  Patches:
	      asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff
	                                       uploaded by Michael L. Young (license 5026)

	  Review: https://reviewboard.asterisk.org/r/3856/
	  ........

	  Merged revisions 422274 from http://svn.asterisk.org/svn/asterisk/branches/11
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2014-08-28 12:29 +0000 [4e750a26fd]  Richard Mudgett <rmudgett@digium.com>

	* Added ConfBridge AMI event note to UPGRADE.txt.
	  ........

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2014-08-28 11:06 +0000 [ef28cc0d43]  Paul Belanger

	* chan_sip.c: Add 'rtpbindaddr' setting

	  Users now have the ability to bind the rtpengine instance to a specific IP
	  address.  For example, you want chan_sip (call control) on eth0 but rtp (media)
	  on eth1.

	  ASTERISK-24280 #close
	  Reported by: Paul Belanger
	  Tested by: Paul Belanger
	  Review: https://reviewboard.asterisk.org/r/3952/
	  Patches:
	      rtpengine.diff uploaded by Paul Belanger


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2014-08-28 10:50 +0000 [327d67270f]  Mark Michelson <mmichelson@digium.com>

	* Fix bug that did not allow for multiple batched RLS notifications to be sent.

	  A misunderstanding of how the scheduler worked caused further batched notifications
	  beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after
	  the batched notification is sent. This way, further notifications can be scheduled
	  when they arise.
	  ........

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2014-08-27 19:44 +0000 [94e1b4a8a4]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage.

	  * Fix off nominal ref leak in find_or_create_contact_status().

	  * Add missing NULL check of status in update_contact_status() and
	  init_start_time().
	  ........

	  Merged revisions 422214 from http://svn.asterisk.org/svn/asterisk/branches/12
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2014-08-27 19:16 +0000 [4728c05957]  Richard Mudgett <rmudgett@digium.com>

	* sched: Fix typo and whitespace change.
	  ........

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2014-08-27 12:30 +0000 [7c1a22fba7]  gtjoseph <george.joseph@fairview5.com>

	* confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events

	  Currently there's no way to tell if a user is an admin or not when receiving
	  the join, leave, mute, unmute and talking events.  This patch adds that
	  capability.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3950/
	  ........

	  Merged revisions 422176 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-27 10:39 +0000 [bf85018107]  Kinsey Moore <kmoore@digium.com>

	* CallerID: Fix parsing of malformed callerid

	  This allows the callerid parsing function to handle malformed input
	  strings and strings containing escaped and unescaped double quotes.
	  This also adds a unittest to cover many of the cases where the parsing
	  algorithm previously failed.

	  Review: https://reviewboard.asterisk.org/r/3923/
	  Review: https://reviewboard.asterisk.org/r/3933/
	  ........

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	  ........

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2014-08-26 18:30 +0000 [d199536a04]  gtjoseph <george.joseph@fairview5.com>

	* confbridge: Make kick, mute and unmute handle channel targets consistently.

	  Kick, mute and unmute were a little inconsistent in their handling of channel
	  targets.  This patch cleans that up by insuring they all handle the 'all'
	  target consistently and adds the 'participants' target which acts on
	  non-admins.  Documentation for kick was also cleaned up as it never
	  supported partial channel names.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3944/
	  ........

	  Merged revisions 422090 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-26 17:14 +0000 [c5ab4adf17]  Mark Michelson <mmichelson@digium.com>

	* Fix race condition in the scheduler when deleting a running entry.

	  When scheduled tasks run, they are removed from the heap (or hashtab).
	  When a scheduled task is deleted, if the task can't be found in the
	  heap (or hashtab), an assertion is triggered. If DO_CRASH is enabled,
	  this assertion causes a crash.

	  The problem is, sometimes it just so happens that someone attempts
	  to delete a scheduled task at the time that it is running, leading
	  to a crash. This change corrects the issue by tracking which task
	  is currently running. If that task is attempted to be deleted,
	  then we mark the task, and then wait for the task to complete.
	  This way, we can be sure to coordinate task deletion and memory
	  freeing.

	  ASTERISK-24212
	  Reported by Matt Jordan

	  Review: https://reviewboard.asterisk.org/r/3927
	  ........

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	  ........

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2014-08-25 11:45 +0000 [fefa6fba82]  Richard Mudgett <rmudgett@digium.com>

	* res_musiconhold.c: Release any format refs before memset().

	  * Clear the channel music_state pointer before destroying the music_state
	  object for safety.
	  ........

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2014-08-25 11:16 +0000 [2b19d94a71]  Richard Mudgett <rmudgett@digium.com>

	* res_musiconhold: Fix MOH restarting where it left off from the last hold.

	  Restore code removed by https://reviewboard.asterisk.org/r/3536/ that
	  introduced a regression that prevents MOH from restarting were it left off
	  the last time.

	  ASTERISK-24019 #close
	  Reported by: Jason Richards
	  Patches:
	        jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett

	  Review: https://reviewboard.asterisk.org/r/3928/
	  ........

	  Merged revisions 421976 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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2014-08-24 14:37 +0000 [497a92d079]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs.

	  In order to alter the Contact header on in-dialog requests and responses the
	  Websocket module must be attached on outgoing INVITEs. The Contact header is
	  modified so that the PJSIP transport layer can find and use the existing
	  Websocket connection based on the source IP address, port, and transport.

	  ASTERISK-24143 #close
	  Reported by: Aleksei Kulakov
	  ........

	  Merged revisions 421955 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-24 14:21 +0000 [477e2e6edb]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_transport_websocket: Fix a progressive memory growth.

	  The packet structure used to receive messages was using the transport
	  pool. This meant that for each parsing the pool would grow accordingly.
	  Since memory can not be reclaimed without resetting it this would
	  cause the memory pool to grow and grow.

	  This change uses a specific memory pool for the packet structure and
	  resets it to a fresh state after the message has been received and
	  handled.
	  ........

	  Merged revisions 421939 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-24 13:54 +0000 [2c0cbf8e64]  Joshua Colp <jcolp@digium.com>

	* res_pjsip_transport_websocket: Ensure secure Websocket clients can be called.

	  This change enforces the transport in the Contact header for Websocket clients.
	  Previously a client may provide a transport of 'ws' when it is actually using
	  a transport of 'wss'. This would cause outgoing calls to fail as the existing
	  connection could not be found.
	  ........

	  Merged revisions 421931 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-24 12:22 +0000 [cee660dadf]  Badalian Vyacheslav (license 5249)

	* chan_sip: Use the server reflexive ICE candidate RTCP port as provided.

	  This code originally worked around an issue within res_rtp_asterisk itself.
	  The wrong socket was being used for the STUN check for RTCP, causing the
	  port to be the same as RTP. This was subsequently fixed and the RTCP port
	  provided for the ICE candidate is correct and does not need to be incremented.

	  ASTERISK-23997 #close
	  Reported by: Badalian Vyacheslav
	  Patches:
	   plus1.diff submitted by Badalian Vyacheslav (license 5249)
	  ........

	  Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-22 11:56 +0000 [dcfffce66d]  Mark Michelson <mmichelson@digium.com>

	* Fix a locking inversion in MixMonitor.

	  We need to unlock the audiohook before trying to lock
	  the channel, since the correct locking order is channel
	  then audiohook.
	  ........

	  Merged revisions 421882 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-08-22 11:52 +0000 [33835e17a0]  Jonathan Rose <jrose@digium.com>

	* ARI: Fix a crash caused by hanging during playback to a channel in a bridge

	  ASTERISK-24147 #close
	  Reported by: Edvin Vidmar
	  Review: https://reviewboard.asterisk.org/r/3908/
	  ........

	  Merged revisions 421879 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-22 09:09 +0000 [1498ae0830]  Matt Jordan <mjordan@digium.com>

	* main/message: Add a new-line to a DEBUG message
	  ........

	  Merged revisions 421859 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 17:09 +0000 [f8c4fc1121]  Richard Mudgett <rmudgett@digium.com>

	* res_musiconhold.c: Remove obsolete REF_DEBUG code.

	  Remove unneeded code that writes to the wrong file location in an obsolete
	  format.
	  ........

	  Merged revisions 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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	  ........

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2014-08-21 16:43 +0000 [644e693645]  Mark Michelson <mmichelson@digium.com>

	* Switch from hostname to an IP address in the SDP origin line.

	  Using the hostname in the SDP origin line may not satisfy the requirement
	  of RFC 4566 that we use a FQDN or IP address. This change has us use the
	  same information from the SDP connection line if possible. If not possible,
	  we'll use the configured media address. And if that's not possible, we use
	  the result of a PJLIB call to get the IP address of ourself.

	  ASTERISK-23994 #close
	  Reported by Private Name

	  Review: https://reviewboard.asterisk.org/r/3925
	  ........

	  Merged revisions 421796 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 16:37 +0000 [56a1d4930a]  Mark Michelson <mmichelson@digium.com>

	* Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge.

	  Because of the departable state of channels that enter Stasis bridges, Stasis has to
	  take responsibility for directing the channel to its intended after-bridge destination
	  if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures
	  that when such a move occurs, when the channel leaves the bridging system, any after
	  bridge gotos are honored.

	  Review: https://reviewboard.asterisk.org/r/3920
	  ........

	  Merged revisions 421792 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 16:35 +0000 [4946981646]  Jonathan Rose (license 6182)

	* res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set

	  Due to a faulty function for debugging reference decrementing, it was possible
	  to reduce the refcount on the wrong object if two moh classes of the same name
	  were in the moh class container.

	  (closes issue ASTERISK-22252)
	  Reported by: Walter Doekes
	  Patches:
	      18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182)
	  ........

	  Merged revisions 398937 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 421777 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421779 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 16:28 +0000 [12d34bb12f]  Mark Michelson <mmichelson@digium.com>

	* Let's try checking the name and number, instead of the name twice.
	  ........

	  Merged revisions 421789 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 16:19 +0000 [2150daf748]  Mark Michelson <mmichelson@digium.com>

	* Improve consistency of party ID privacy usage.

	  Prior to this change, the Remote-Party-ID header took the position of
	  "If caller name and number are not explicitly allowed, then they are private"
	  and P-Asserted-Identity took the position of
	  "Caller name and number are only private if marked explicitly so"

	  Now both mechanisms of conveying party identification use the former approach.
	  ........

	  Merged revisions 421778 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 12:35 +0000 [77ddc5b713]  Elazar Broad (License 5835)

	* chan_sip: Don't use port derived from fromdomain if it isn't set

	  If a user does not provide a port in the fromdomain setting, chan_sip will set
	  the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will
	  then get used unilaterally in certain places. This causes issues with TLS,
	  where the default port is expected to be 5061.

	  This patch modifies chan_sip such that fromdomainport is only used if it is
	  not the standard SIP port; otherwise, the port from the SIP pvt's recorded
	  self IP address is used.

	  Review: https://reviewboard.asterisk.org/r/3893/

	  ASTERISK-24178 #close
	  Reported by: Elazar Broad
	  patches:
	    fromdomainport_fix.diff uploaded by Elazar Broad (License 5835)
	  ........

	  Merged revisions 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

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	  ........

	  Merged revisions 421719 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 10:25 +0000 [f3a525e9a6]  Matt Jordan <mjordan@digium.com>

	* ARI: Fix implicit answer when playback is initiated on unanswered channel

	  When issuing a POST /channels/{channel_id}/play on a channel that is not
	  yet answered, ARI is supposed to:
	  * Queue up an AST_CONTROL_PROGRESS on the channel
	  * Start up the playback of the media

	  Instead, we sneak an answer on the channel right before starting playing media.

	  This is due to ARI's usage of control_streamfile. This function implicitly
	  answers the channel (and doesn't give ARI the option to stop it). The answering
	  of the channel here is probably unnecessary:
	  * app_voicemail, by far the biggest consumer of this function, always answers
	    the channels anyway
	  * control stream file (in res_agi) and ControlPlayback probably shouldn't be
	    implicitly answering the channel. Answering should not be tied directly to
	    playing back media.

	  As it turns out, the answering of the channel here is pretty old:
	  356042    twilson       if (ast_channel_state(chan) != AST_STATE_UP) {
	    3087      anthm               res = ast_answer(chan);
	  180259   tilghman       }

	  (As in, ancient?)

	  Note that others ran into this problem and commented about it on various
	  mailing lists.

	  Review: https://reviewboard.asterisk.org/r/3907/

	  ASTERISK-24229 #close
	  Reported by: Matt Jordan
	  ........

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2014-08-21 09:52 +0000 [085d5a2629]  Shaun Ruffell (License 5417)

	* Clean up files that do not end with newlines

	  Trivial patch to add new lines to several files missing them. This fixes
	  warnings when compiling with gcc 4.1.2 on CentOS 5.

	  ASTERISK-24245 #close
	  Reported by: Shaun Ruffell
	  patches:
	    0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417)
	  ........

	  Merged revisions 421677 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-21 09:42 +0000 [da91946df7]  Shaun Ruffell (License 5417)

	* uri: Quiet warning about type qualifiers ignored on function return type

	  This patch fixes gcc warnings that occur due to the type qualifier 'const'
	  being ignored on a return type of int.

	  ASTERISK-24246 #close
	  Reported by: Shaun Ruffell
	  patches:
	    0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417)
	  ........

	  Merged revisions 421675 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-08-20 17:52 +0000 [b7f98c3da4]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip: Update media translation paths when new SDP negotiated.

	  On a SIP reinvite that changes media strams, the PJSIP channel driver was
	  flooding the log with "Asked to transmit frame type %s, while native
	  formats is %s" warnings.

	  * Fixes PJSIP not setting up translation paths when the formats change on
	  a reinvite.  AFS-63 was effectively reintroduced because of the media
	  formats work.  res_pjsip_sdp_rtp.c:set_caps()

	  * Improved the unexpected frame format WARNING message to include more
	  information.

	  * Added protective locking while altering formats on a channel.  Reworked
	  set_format() to simplify and protect the formats under manipulation.

	  * Restored some code that got lost in the media_formats work.
	  (channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())

	  AFS-137 #close
	  Reported by: Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/3906/
	  ........

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2014-08-20 17:23 +0000 [4672c139dd]  Richard Mudgett <rmudgett@digium.com>

	* cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.

	  filename_completion_function() returns memory that was not allocated by
	  the MALLOC_DEBUG allocation tracker so the memory must be freed by
	  ast_std_free().
	  ........

	  Merged revisions 421600 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 421602 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421608 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-20 15:41 +0000 [49f8bd4ad4]  Mark Michelson <mmichelson@digium.com>

	* Set the role for inbound subscriptions correctly.

	  This was causing the AMI show_subscriptions test in
	  the testsuite to fail since all subscriptions were being
	  seen as subscribers instead of notifiers.
	  ........

	  Merged revisions 421585 from http://svn.asterisk.org/svn/asterisk/branches/13


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2014-08-20 15:04 +0000 [d0640ad7df]  Mark Michelson <mmichelson@digium.com>

	* Move evaluation of set_var options in pjsip to the end of channel initialization.

	  This allows for set_var to override certain defaults such as caller ID and codec
	  values. This also fixes a test suite regression. The "set_var" test suite test attempted
	  to use set_var to override caller ID, but a recent change caused that to no longer work.
	  ........

	  Merged revisions 421565 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-20 08:06 +0000 [36f4bff943]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Add information to blind transfer event

	  When a blind transfer occurs that is forced to create a local channel
	  pair to satisfy the transfer request, information about the local
	  channel pair is not published. This adds a field to describe that
	  channel to the blind transfer message struct so that this information
	  is conveyed properly to consumers of the blind transfer message.

	  This also fixes a bug in which Stasis() was unable to properly identify
	  the channel that was replacing an existing Stasis-controlled channel
	  due to a blind transfer.

	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/3921/
	  ........

	  Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-20 07:39 +0000 [01f1ff1f77]  Kinsey Moore <kmoore@digium.com>

	* AMI: Add AllVariables parameter to Status

	  This adds the AllVariables parameter to the Status AMI action such that
	  if defined and set to "true", all channel variables will be reported in
	  the subsequent Status event(s). This parameter does not negate the
	  functionality of the "Variables" parameter so that global variables and
	  dialplan functions can be requested.

	  Review: https://reviewboard.asterisk.org/r/3915/


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421534 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-19 15:28 +0000 [76290adf50]  Mark Michelson <mmichelson@digium.com>

	* Alter documentation for callerid_privacy to use correct values.
	  ........

	  Merged revisions 421485 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-19 14:55 +0000 [28a89e7685]  Mark Michelson <mmichelson@digium.com>

	* Fix compilation error on certain versions of GCC.
	  ........

	  Merged revisions 421447 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-19 14:43 +0000 [a85a483fcd]  Kinsey Moore <kmoore@digium.com>

	* AMI Docs: Fix Status channel parameter optionality
	  ........

	  Merged revisions 421442 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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	  Merged revisions 421443 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

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	  ........

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2014-08-19 11:36 +0000 [222b5cd036]  Krandon Bruse (license 6631)

	* ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX

	  If /channels/{channelID}/continue is called on a channel that was originated
	  without a PBX (such as the ARI command POST channel with a stasis application
	  argument), the channel will not start dialplan execution. This patch will now
	  run the PBX out of the stasis execution if the channel doesn't currently have
	  an active PBX upon continuing.

	  ASTERISK-24043 #close
	  Reported by: Krandon Bruse
	  Review: https://reviewboard.asterisk.org/r/3917/
	  Patches:
	      stasis-continue.diff submitted by Krandon Bruse (license 6631)
	  ........

	  Merged revisions 421416 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

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2014-08-19 11:16 +0000 [83a9b91da9]  Richard Mudgett <rmudgett@digium.com>

	* chan_pjsip: Fix attended transfer connected line name update.

	  A calls B
	  B answers
	  B SIP attended transfers to C
	  C answers, B and C can see each other's connected line information
	  B completes the transfer
	  A has number but no name connected line information about C
	    while C has the full information about A

	  I examined the incoming and outgoing party id information handling of
	  chan_pjsip and found several issues:

	  * Fixed ast_sip_session_create_outgoing() not setting up the configured
	  endpoint id as the new channel's caller id.  This is why party A got
	  default connected line information.

	  * Made update_initial_connected_line() use the channel's CALLERID(id)
	  information.  The core, app_dial, or predial routine may have filled in or
	  changed the endpoint caller id information.

	  * Fixed chan_pjsip_new() not setting the full party id information
	  available on the caller id and ANI party id.  This includes the configured
	  callerid_tag string and other party id fields.

	  * Fixed accessing channel party id information without the channel lock
	  held.

	  * Fixed using the effective connected line id without doing a deep copy
	  outside of holding the channel lock.  Shallow copy string pointers can
	  become stale if the channel lock is not held.

	  * Made queue_connected_line_update() also update the channel's
	  CALLERID(id) information.  Moving the channel to another bridge would need
	  the information there for the new bridge peer.

	  * Fixed off nominal memory leak in update_incoming_connected_line().

	  * Added pjsip.conf callerid_tag string to party id information from
	  enabled trust_inbound endpoint in caller_id_incoming_request().

	  AFS-98 #close
	  Reported by: Mark Michelson

	  Review: https://reviewboard.asterisk.org/r/3913/
	  ........

	  Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-18 16:18 +0000 [c4c9d4ad6c]  Damien Wedhorn <voip@facts.com.au>

	* Skinny: Fixup compile warning for non dev-mode.
	  ........

	  Merged revisions 421376 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421380 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-18 15:20 +0000 [1de8b8035e]  gtjoseph <george.joseph@fairview5.com>

	* func_config: Change 'Not Found' message from ERROR to DEBUG

	  When you call the CONFIG dialplan function with the name of a variable that
	  doesn't exist in the target context you get an ERROR.  This does nothing but
	  clutter up the logs with messages that may be perfectly acceptable.  Just
	  because a variable wasn't in the context doesn't mean it's an error.  Maybei
	  t's optional or just needs to be defaulted or ignored.

	  This patch changes the log level from ERROR to DEBUG.  If a dialplan developer
	  wants to debug their dialplan they still canby setting the console debug level 
	  as needed.

	  Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3919/
	  ........

	  Merged revisions 421327 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 421328 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421329 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421337 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-17 20:14 +0000 [bb494067a5]  Matt Jordan <mjordan@digium.com>

	* Multiple revisions 421311-421312

	  ........
	    r421311 | mjordan | 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) | 9 lines
	    
	    res/ari/resource_channels: Don't return allocation failure on failed function
	    
	    If a function fails to execute, it is most likely due to one of two reasons:
	    (1) The function doesn't exist or can't be read from
	    (2) The function is dangerous and is restricted based on the user's permissions
	    
	    Currently we return allocation failure, which is incorrect. This updates the
	    reason code to more accurately reflect why the request failed.

	    ASTERISK-24215
	  ........
	    r421312 | mjordan | 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) | 4 lines
	    
	    res/ari/resource_channels: Fix compilation issue
	    
	    Forgot a parameter. Whoops.
	  ........

	  Merged revisions 421311-421312 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421313 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-17 19:57 +0000 [ba5d5da60b]  Matt Jordan <mjordan@digium.com> (License #6283)

	* Improve call forwarding reporting, especially with regards to ARI.

	  This patch addresses a few issues:

	  1) The order of Dial events have been changed when performing a call forward.
	     The order has now been altered to
	      1) Dial begins dialing channel A.
	      2) When A forwards the call to B, we issue the dial end event to channel
	         A, indicating the dial is being canceled due to a forward to B.
	      3) When the call to channel B occurs, we then issue a new dial begin to
	         channel B.

	  2) Call forwards are now reported on the calling channel, not the peer channel.

	  3) AMI DialEnd events have been altered to display the extension the call is
	     being forwarded to when relevant.

	  4) You can now get the values of channel variables for channels that are not
	     currently in the Stasis application. This brings the retrieval of channel
	     variables more in line with the rest of channel read operations since they
	     may be performed on channels not in Stasis.

	  ASTERISK-24134 #close
	  Reported by Matt Jordan

	  ASTERISK-24138 #close
	  Reported by Matt Jordan

	  Patches:
	  	forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

	  Review: https://reviewboard.asterisk.org/r/3899
	  ........

	  Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-17 18:29 +0000 [6525f374db]  Matt Jordan <mjordan@digium.com>

	* apps/app_meetme: Fix crash when publishing MeetMe messages with no channel

	  The same function, meetme_stasis_generate_msg, handles creating and publishing
	  Stasis message both when there are channels in the MeetMe conference and when
	  there are no channels in the conference. When the performance improvement was
	  made to use cached snapshots, this created a situation where Asterisk would
	  crash: obtaining a cached snapshot is not NULL tolerant.

	  This patch restores the previous implementation, which used a NULL safe set
	  of routines to produce a blob containing the channel snapshot (if available)
	  and information about the MeetMe conference.

	  ASTERISK-24234 #close
	  Reported by: Shaun Ruffell
	  Tested by: Shaun Ruffell
	  ........

	  Merged revisions 421270 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421273 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-17 18:10 +0000 [44fc6ea6ff]  Richard Mudgett <rmudgett@digium.com> (License 5621)

	* apps/app_dial: Fix Dial 'z' option

	  The 'z' option is supposed to disable the dial timeout in the case of a call
	  forward. Unfortunately, the wrong timeout timer was passed to the do_forward
	  function, resulting in the option not working.

	  ASTERISK-24225 #close
	  Reported by: dimitripietro
	  Tested by: dimitripietro
	  patches:
	    jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621)
	    jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621)
	  ........

	  Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 421233 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421235 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421236 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-17 17:35 +0000 [98ca5c0b5f]  cloos <cloos@jhcloos.com> (License 5956)

	* configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc

	  Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
	  executed with optimization. This "help" unfortunately results in re-definition
	  warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
	  patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.

	  Review: https://reviewboard.asterisk.org/r/3912/

	  ASTERISK-24032 #close
	  Reported by: Kilburn
	  Tested by: Kilburn, wdoekes
	  patches:
	    1.8.diff uploaded by cloos (License 5956)
	    10.diff uploaded by cloos (License 5956)
	    11.diff uploaded by cloos (License 5956)
	    12.diff uploaded by cloos (License 5956)
	    13.diff uploaded by cloos (License 5956)
	  ........

	  Merged revisions 421227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 421228 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421229 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421230 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421231 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-17 11:11 +0000 [952da298ce]  Joshua Colp <jcolp@digium.com>

	* res_http_websocket: Include query parameters in client connection requests.

	  Review: https://reviewboard.asterisk.org/r/3914/
	  ........

	  Merged revisions 421210 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421211 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-15 12:26 +0000 [9b658b7c60]  Jonathan Rose <jrose@digium.com>

	* Bridging: Fix a behavioral change when checking if a channel is leaving a bridge

	  r420934 introduced some failures in the test suite.  Upon investigating, it was
	  discovered that differences in the way we were evaluating whether a channel was in
	  the process of leaving a bridge were causing some reinvites not to occur (mostly
	  reinvites back to Asterisk when ending a call). This patch fixes that behavioral
	  change.

	  ASTERISK-24027 #close
	  Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/3910/
	  ........

	  Merged revisions 421186 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421187 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421195 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-15 10:50 +0000 [0d0a616e1a]  Matt Jordan <mjordan@digium.com>

	* app_voicemail/app: Remove test events that were duplicated by r421059

	  Moving the test event raised when a file is played back (which occurred in
	  r421059) broke the ever loving snot out of the voicemail tests. This caused
	  duplicate test events to get raised, as app_voicemail and main/app were raising
	  events prior to call ast_streamfile. The voicemail tests did not enjoy getting
	  multiple events.

	  Since raising the playback event in ast_streamfile is far more useful to the
	  vast majority of tests, this patch keeps the call there and simply removes the
	  extraneous calls that duplicated the event.
	  ........

	  Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 421164 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421165 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421166 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-14 16:16 +0000 [980e49614c]  Matt Jordan <mjordan@digium.com>

	* res/res_hep_rtcp: Remove dependency on PJSIP

	  The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need
	  to be included, as the module does not using PJPROJECT any fashion.
	  Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as
	  a dependency, this also meant that res_hep_rtcp will fail to compile on a
	  system without PJPROJECT.

	  This patch removes the include.

	  Thanks to Damien Wedhorn for pointing this out in #asterisk-dev.

	  ASTERISK-24236 #close
	  Reported by: Damien Wedhorn, Matt Jordan
	  Tested by: Damien Wedhorn
	  ........

	  Merged revisions 421064 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421065 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421066 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-14 15:59 +0000 [513981c89d]  Matt Jordan <mjordan@digium.com>

	* main/file: Move test event to emit PLAYBACK event more consistently

	  This is being done in advance of the test for ASTERISK-23953
	  ........

	  Merged revisions 421059 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 421060 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 421061 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421062 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-14 14:21 +0000 [0b11c48522]  Matt Jordan <mjordan@digium.com>

	* cel: Make sure channels in extra fields include their unique IDs as well

	  CEL typically tracks a lot of information using the unique ID of the channel.
	  This is typically needed due to tying events together using the linked ID of
	  the various channels involved in a "call", which is derived from the channel ID
	  of the oldest channel involved in a bridge (or in the case of a Dial, the
	  parent channel).

	  Previously, we had updated the extra fields to include the involved channel
	  names, but forgot to put in the unique ID. This patch corrects that error.
	  ........

	  Merged revisions 421037 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421042 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421043 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-14 11:33 +0000 [79c5c08db9]  Richard Mudgett <rmudgett@digium.com>

	* ARI: Originate to app local channel subscription code optimization.

	  Reduce the scope of local_peer and only get it if the ARI originate is
	  subscribing to the channels.

	  Review: https://reviewboard.asterisk.org/r/3905/
	  ........

	  Merged revisions 421009 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 421010 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421012 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-14 11:01 +0000 [e4b32731b9]  Richard Mudgett <rmudgett@digium.com>

	* channel_internal_api.c: Replace some code with ao2_replace().

	  Use ao2_replace() instead of ao2_cleanup(); ao2_bump().

	  ao2_replace() has the advantange of not altering the ref count if the
	  replaced pointer is the same.

	  Review: https://reviewboard.asterisk.org/r/3904/
	  ........

	  Merged revisions 420992 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420993 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-13 12:05 +0000 [dd41d0ff01]  Richard Mudgett <rmudgett@digium.com>

	* res_pjsip_send_to_voicemail.c: Fix svn file properties.
	  ........

	  Merged revisions 420956 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420957 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-13 11:56 +0000 [6aa510b41f]  Kinsey Moore <kmoore@digium.com>

	* PJSIP: Prevent crash no-URI contacts

	  This prevents a crash from occurring when a contact with no URI is used
	  for the creation of an outbound out-of-dialog request with no
	  associated endpoint.
	  ........

	  Merged revisions 420949 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420950 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420953 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-13 11:24 +0000 [d4695774e7]  Jonathan Rose <jrose@digium.com>

	* Bridges: Fix feature interruption/unintended kick caused by external actions

	  If a manager or CLI user attached a mixmonitor to a call running a dynamic
	  bridge feature while in a bridge, the feature would be interrupted and the
	  channel would be forcibly kicked out of the bridge (usually ending the call
	  during a simple 1 to 1 call). This would also occur during any similar action
	  that could set the unbridge soft hangup flag, so the fix for this was to
	  remove unbridge from the soft hangup flags and make it a separate thing all
	  together.

	  ASTERISK-24027 #close
	  Reported by: mjordan
	  Review: https://reviewboard.asterisk.org/r/3900/
	  ........

	  Merged revisions 420934 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420940 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-13 09:31 +0000 [6a6702bb0f]  Kinsey Moore <kmoore@digium.com>

	* AMI: Improve documentation for Status action
	  ........

	  Merged revisions 420919 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420921 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-13 02:54 +0000 [52c94d3af4]  Walter Doekes <walter+asterisk@wjd.nu>

	* logger: Don't store verbose-magic in the log files.

	  In r399267, the verbose2magic stuff was edited. This time it results
	  in magic characters in the log files for multiline messages.

	  In trunk (and 13) this was fixed by the "stripping" of those
	  characters from multiline messages (in r414798).

	  This fix is altered to actually strip the characters and not replace
	  them with blanks.

	  Review: https://reviewboard.asterisk.org/r/3901/
	  Review: https://reviewboard.asterisk.org/r/3902/
	  ........

	  Merged revisions 420897 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 420898 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420899 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-12 18:45 +0000 [969982b878]  Richard Mudgett <rmudgett@digium.com>

	* chan_sip: Fix type mismatch when the format is changed.

	  Symptom is most likely an invalid ao2 object bad magic number message or a
	  less likely crash.
	  ........

	  Merged revisions 420881 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420882 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-12 18:36 +0000 [8526d967c9]  Richard Mudgett <rmudgett@digium.com>

	* res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup.

	  * Made use ast_copy_string() instead of strcpy() for snoop uniqueid for
	  safety.  There is no guarantee that the max channel uniqueid length will
	  remain the same as the snoop uniqueid space.
	  ........

	  Merged revisions 420879 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420880 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-12 06:18 +0000 [ca61f8ac82]  Joshua Colp <jcolp@digium.com>

	* app_voicemail: Fix the "test_voicemail_vm_info" unit test.
	  ........

	  Merged revisions 420856 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 16:04 +0000 [aba07a0f6e]  Richard Mudgett <rmudgett@digium.com>

	* res/stasis/command.c: Fix recent commit using spaces instead of tabs.
	  ........

	  Merged revisions 420836 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420837 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 13:51 +0000 [ffccae8269]  Matt Jordan <mjordan@digium.com>

	* AMI/ARI: Update version to 2.5.0/1.5.0 respectively

	  This is to support the backwards compatible changes made in the next version
	  of Asterisk.
	  ........

	  Merged revisions 420805 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420808 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420811 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 13:46 +0000 [7a4691b425]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Use the correct return value

	  Return the correct value instead of always returning 0 when setting
	  internal status on unreal channels.

	  Reported by: Richard Mudgett
	  ........

	  Merged revisions 420802 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420803 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 13:38 +0000 [6f735288b0]  Kinsey Moore <kmoore@digium.com>

	* Stasis: Allow internal channels directly into bridges

	  The patch to catch channels being shoehorned into Stasis() via external
	  mechanisms also happens to catch Announcer and Recorder channels
	  because they aren't known to be stasis-controlled channels in the usual
	  sense. This marks those channels as Stasis()-internal channels and
	  allows them directly into bridges.

	  Review: https://reviewboard.asterisk.org/r/3903/
	  ........

	  Merged revisions 420795 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420796 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420797 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 12:40 +0000 [db0a97f8ce]  Mark Michelson <mmichelson@digium.com>

	* Fix crashing unit tests with regards to RLS.

	  The unit tests require a sorcery.conf file that has been
	  set up to store resource lists in memory rather than retrieving
	  from configuration.

	  With a setup that is not conducive to running the tests, a fault
	  in sorcery currently causes Asterisk to crash when attempting to
	  run any of the tests.

	  To get around the crash, this adds a function that verifies the
	  current environment and marks the tests as "not run" if the setup
	  is not correct.
	  ........

	  Merged revisions 420779 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420780 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 11:03 +0000 [b4e33c81e3]  Mark Michelson <mmichelson@digium.com>

	* Fix crash encountered by the testsuite.

	  Running testsuite tests locally produced no errors, but when
	  run using the continuous integration framework, crashes occurred.

	  The crashes occurred due to a refcounting error that had been fixed
	  for a similar situation.
	  ........

	  Merged revisions 420758 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 08:57 +0000 [becf7c7003]  Matt Jordan <mjordan@digium.com>

	* res_hep: Remove disabling of modules

	  These modules were originally specified as being disabled, as they were
	  introduced midstream in Asterisk 12. That makes it nicer for folks who are
	  upgrading to a new release in the middle of Asterisk 12. That's not the case
	  for Asterisk 13: it's a brand new release. There's no reason to have the
	  modules disabled by default in that case.
	  ........

	  Merged revisions 420742 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420743 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 05:41 +0000 [1e0846167b]  Walter Doekes <walter+asterisk@wjd.nu>

	* general: Fix memory Corruption in __ast_string_field_ptr_build_va.

	  If the space left in a stringfield is between 0 and
	  (alignof(ast_string_field_allocation)-1) adding new data would cause
	  memory corruption, because we would assume enough space (unsigned
	  underrun).

	  Thanks Arnd Schmitter for reporting and finding out the cause!

	  ASTERISK-23508 #close
	  Reported by: Arnd Schmitter
	  Tested by: Arnd Schmitter, JoshE

	  Review: https://reviewboard.asterisk.org/r/3898/
	  ........

	  Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 420715 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 420716 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420717 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420718 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-11 04:55 +0000 [b2afbc48e4]  Walter Doekes <walter+asterisk@wjd.nu>

	* tcptls: Avoid compiler warning on non-dev-mode.
	  ........

	  Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........

	  Merged revisions 420655 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........

	  Merged revisions 420656 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420657 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420658 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-10 20:31 +0000 [6650704414]  Matt Jordan <mjordan@digium.com>

	* funcs/func_jitterbuffer: Tweak documentation

	  This patch merely reformats and cleans up a bit of the jitterbuffer
	  documentation for the wiki.
	  ........

	  Merged revisions 420639 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-10 19:14 +0000 [add46fd27c]  Michael K (License 6621)

	* app_queue: Add RealTime support for queue rules

	  This patch gives the optional ability to keep queue rules in RealTime. It is
	  important to note that with this patch:
	   (a) Queue rules in RealTime are only examined on module load/reload
	   (b) Queue rules are loaded both from the queuerules.conf file as well as the
	       RealTime backend
	  To inform app_queue to examine RealTime for queue rules, a new setting has been
	  added to queuerules.conf's general section "realtime_rules". RealTime queue
	  rules will only be used when this setting is set to "yes".

	  The schema for the database table supports a rule_name, time, min_penalty, and
	  max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or
	  '+' literal is provided. Otherwise, the penalties are treated as constants.

	  For example:
	  rule_name, time, min_penalty, max_penalty
	  'default', '10', '20', '30'
	  'test2', '20', '30', '55'
	  'test2', '25', '-11', '+1111'
	  'test2', '400', '112', '333'
	  'test3', '0', '4564', '46546'
	  'test_rule', '40', '15', '50'

	  which would result in :

	  Rule: default
	   - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust
	     QUEUE_MIN_PENALTY to 20
	  Rule: test2
	   - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
	     QUEUE_MIN_PENALTY to 30
	   - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust
	     QUEUE_MIN_PENALTY by -11
	   - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
	     QUEUE_MIN_PENALTY to 112
	  Rule: test3
	   - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust
	     QUEUE_MIN_PENALTY to 4564
	  Rule: test_rule
	   - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust
	     QUEUE_MIN_PENALTY to 15

	  If you use RealTime, the queue rules will be always reloaded on a module
	  reload, even if the underlying file did not change. With the option disabled,
	  the rules will only be reloaded if the file was modified.

	  Review: https://reviewboard.asterisk.org/r/3607/

	  ASTERISK-23823 #close
	  Reported by: Michael K
	  patches:
	    app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621)
	  ........

	  Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-10 17:02 +0000 [f7bb772804]  Matt Jordan <mjordan@digium.com>

	* Update CHANGES file
	  ........

	  Merged revisions 420609 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-10 16:35 +0000 [455243cdd4]  Matt Jordan <mjordan@digium.com>

	* Update UPGRADE-13.txt file

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-08 15:08 +0000 [3e452fa4d9]  Jason Parker <jparker@digium.com>

	* Fix build in devmode.
	  ........

	  Merged revisions 420592 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-08 14:16 +0000 [5ce4ad8031]  Jason Parker <jparker@digium.com>

	* app_voicemail: Add the ability to specify multiple email addresses.

	  ASTERISK-24045
	  Reported by: Jacob Barber
	  Review: https://reviewboard.asterisk.org/r/3833/
	  ........

	  Merged revisions 420577 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-08 12:53 +0000 [91f7b66183]  Matt Jordan <mjordan@digium.com>

	* chan_sip: Mark chan_sip and its files as extended support
	  ........

	  Merged revisions 420562 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-08 07:40 +0000 [86e927a714]  Matt Jordan <mjordan@digium.com>

	* make_ari_stubs: Update wiki prefix to '13'
	  ........

	  Merged revisions 420538 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420539 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-08 07:38 +0000 [1f35fccda1]  Matt Jordan <mjordan@digium.com>

	* res_ari_resource.c.mustache: Update template to emit module support level
	  ........

	  Merged revisions 420536 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-08 07:33 +0000 [008c1ad9bf]  Matt Jordan <mjordan@digium.com>

	* main/message: remove debug message
	  ........

	  Merged revisions 420533 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420534 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420535 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-07 22:07 +0000 [c94fef6f36]  Kinsey Moore <kmoore@digium.com>

	* CEL: Update unit tests for additional information

	  This updates the CEL unit tests for the new information contained in
	  the attended transfer CEL extra field.
	  ........

	  Merged revisions 420513 from http://svn.asterisk.org/svn/asterisk/branches/12
	  ........

	  Merged revisions 420514 from http://svn.asterisk.org/svn/asterisk/branches/13


	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-07 20:37 +0000 [96be6b2228]  Matt Jordan <mjordan@digium.com>

	* Initialize svnmerge from branches/13

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-07 20:36 +0000 [38a0df95b1]  Matt Jordan <mjordan@digium.com>

	* Remove 12 merge properties

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-08-07 20:33 +0000 [5760526f69]  Matt Jordan <mjordan@digium.com>

	* Update UPGRADE.txt for 13 branch

	  git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2014-10-24  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 13.0.0 Released.

2014-10-22 21:27 +0000 [r426097]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: codec_dahdi: Cannot use struct
	  ast_translator.core_{src,src}_codec. This fixes a Segmentation
	  fault introduced in r419044 "media formats: re-architect handling
	  of media for performance improvements". The problem is that
	  codec_dahdi was using core_src_codec and core_dst_codec in the
	  ast_translator structure when these fields were never set. Now
	  instead of trying to map the new core codec descriptions to the
	  way DAHDI defines different codecs, we will store the DAHDI
	  specific formats in 'struct translator' directly so we can refer
	  to them without mapping. This also allows us to remove the
	  "global_format_map" structure, since we can now query the list of
	  translators directly to make sure we do not ever register a DAHDI
	  based translator for a specific path more than once and eliminate
	  the need to keep the list and the map in sync. ASTERISK-24435
	  #close Reported by: Marian Koniuszko Review:
	  https://reviewboard.asterisk.org/r/4105/

2014-10-21 17:47 +0000 [r426079]  Richard Mudgett <rmudgett@digium.com>

	* main/translate.c: translage.c: Fix regression when generating
	  translation path strings. Fix the AMI Status action read and
	  write translation path strings from growing for each channel in
	  the status event list by reseting the ast string given to
	  ast_translate_path_to_str() to fill in the given translation
	  path.

2014-10-20 14:15 +0000 [r425991]  Matthew Jordan <mjordan@digium.com>

	* res/res_xmpp.c, main/tcptls.c, /: AST-2014-011: Fix POODLE
	  security issues There are two aspects to the vulnerability: (1)
	  res_jabber/res_xmpp use SSLv3 only. This patch updates the module
	  to use TLSv1+. At this time, it does not refactor
	  res_jabber/res_xmpp to use the TCP/TLS core, which should be done
	  as an improvement at a latter date. (2) The TCP/TLS core, when
	  tlsclientmethod/sslclientmethod is left unspecified, will default
	  to the OpenSSL SSLv23_method. This method allows for all
	  encryption methods, including SSLv2/SSLv3. A MITM can exploit
	  this by forcing a fallback to SSLv3, which leaves the server
	  vulnerable to POODLE. This patch adds WARNINGS if a user uses
	  SSLv2/SSLv3 in their configuration, and explicitly disables
	  SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk
	  will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly
	  chosen. For TLS servers, Asterisk will no longer support SSLv2 or
	  SSLv3. Much thanks to abelbeck for reporting the vulnerability
	  and providing a patch for the res_jabber/res_xmpp modules.
	  Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425
	  #close Reported by: abelbeck Tested by: abelbeck, opsmonitor,
	  gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by
	  abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch
	  uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff
	  uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded
	  by mjordan (License 6283) ........ Merged revisions 425987 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-19 17:07 +0000 [r425965]  George Joseph <george.joseph@fairview5.com>

	* Makefile, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, makeopts.in: build: Force -fsigned-char on
	  platforms where the default for char is unsigned gcc on the ARM
	  platform defaults 'char' to 'unsigned char' whereas Intel and
	  SPARC default to 'signed char'. This is only an issue in the rare
	  cases where negative values are assigned to a 'char' but this
	  this patch insures compatibility by detecting platforms that
	  default to 'unsigned' and adding an '-fsigned-char' flag to
	  _ASTCFLAGS. If compiling for ARM (native or cross-compile) be
	  sure to run ./bootstrap.sh and ./configure to regenerate the
	  build files. You shouldn't have to do this for Intel or SPARC.
	  Tested-by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/4091/ ........ Merged
	  revisions 425964 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-19 04:01 +0000 [r425923-425944]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert 425922
	  This patch for r425922 introduced a bug, wherein sending an
	  INVITE request with no SDP would cause Asterisk to not send an
	  SDP Offer in the 200 OK. The current structure of
	  res_pjsip_sdp_rtp is a bit hard to deal with to fix this, as
	  create_outgoing_sdp has no knowledge of whether or not it is
	  creating an SDP as a new Offer or an Answer. This is something of
	  an oversight in the callback definition, as the caller of it does
	  have this information.

	* res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Remove left over
	  reference to override_prefs The usage of the local override_prefs
	  variable in create_outgoing_sdp_stream was previously to track an
	  override format preference set by PJSIP_MEDIA_OFFER. Now,
	  however, that function simply sets the joint capabilities
	  structure, session->req_caps. During the media format rework, the
	  override_prefs was instead used to check if there were any
	  formats in session->req_caps. However, this usage isn't useful in
	  create_outgoing_sdp_stream. session->req_caps contains the
	  negotiated formats for *all* streams, not just the current one
	  being created. Thus, so long as any stream of any type has
	  provided a format, override_prefs will be non-zero. Hence, its
	  usage in checking whether or not we should look at the formats on
	  the endpoint or the joint capabilities is generally useless.
	  There's only two things useful to check: (1) Does the endpoint
	  have a format for the media type? (2) Did we negotiate a format
	  for the media type? If either of those is a 'no', then we must
	  kill the media stream.

2014-10-17 22:43 +0000 [r425905]  Jonathan Rose <jrose@digium.com>

	* configs/samples/cli_aliases.conf.sample: Sample Configurations:
	  make 'pjsip reload' reload all reloadable pjsip modules AST-1432
	  #close Reported by: John Bigelow

2014-10-17 13:35 +0000 [r425821-425879]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip_sdp_rtp.c, res/res_pjsip.c,
	  res/res_pjsip_session.c, /: res_pjsip_session/res_pjsip_sdp_rtp:
	  Be more tolerant of offers When an inbound SDP offer is received,
	  Asterisk currently makes a few incorrection assumptions: (1) If
	  the offer contains more than a single audio/video stream,
	  Asterisk will reject the entire stream with a 488. This is an
	  overly strict response; generally, Asterisk should accept the
	  media streams that it can accept and decline the others. (2) If
	  the offer contains a declined media stream, Asterisk will attempt
	  to process it anyway. This can result in attempting to match
	  format capabilities on a declined media stream, leading to a 488.
	  Asterisk should simply ignore declined media streams. (3)
	  Asterisk will currently attempt to handle offers with AVPF with
	  use_avpf=No/AVP with use_avpf=Yes. This mismatch results in
	  invalid SDP answers being sent in response. If there is a
	  mismatch between the media type being offered and the
	  configuration, Asterisk must reject the offer with a 488. This
	  patch does the following: * Asterisk will accept SDP offers with
	  at least one media stream that it can use. Some WARNING messages
	  have been dropped to NOTICEs as a result. * Asterisk will not
	  accept an offer with a media type that doesn't match its
	  configuration. * Asterisk will ignore declined media streams
	  properly. #SIPit31 Review:
	  https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close
	  Reported by: James Van Vleet ASTERISK-24381 #close Reported by:
	  Matt Jordan ........ Merged revisions 425868 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: channels/chan_sip: Respect outboundproxy
	  setting when sending qualify requests The outboundproxy setting
	  is currently ignored when sending OPTIONS requests as a result of
	  the qualify setting. This means that if an Asterisk server is
	  unable to send the packet directly to a peer, it is unable to
	  qualify any non-inbound registered peer (e.g. a peer SIP Trunk).
	  This patch grabs the outboundproxy information for a peer when a
	  qualify attempt is being constructed and, if it finds the
	  information, uses it when sending the OPTIONS request. Review:
	  https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close
	  Reported by: Damian Ivereigh patches: outboundproxy-dai.patch
	  uploaded by Damian Ivereigh (License 6632) ........ Merged
	  revisions 425818 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425819 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425820 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-17 02:41 +0000 [r425783]  Richard Mudgett <rmudgett@digium.com>

	* main/core_unreal.c, main/channel.c, /: AMI: Add missing VarSet
	  events when a channel inherits variables. There should be AMI
	  VarSet events when channel variables are inherited by an outgoing
	  channel. Also local;2 should generate VarSet events when it gets
	  all of its channel variables from channel local;1. ASTERISK-24415
	  #close Reported by: Richard Mudgett Patches:
	  jira_asterisk_24415_v12.patch (license #5621) patch uploaded by
	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/
	  ........ Merged revisions 425782 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-17 01:57 +0000 [r425736-425761]  Matthew Jordan <mjordan@digium.com>

	* /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix audio
	  issues when moving from remote bridge to softmix When a native
	  RTP bridge that is remotely bridging its participants switches to
	  a softmix bridge, it may not properly re-INVITE the media for one
	  or both participants back to Asterisk. This is due to the current
	  bridge_native_rtp code only re-INVITEs if it believes the channel
	  will survive the bridge operation. Currently, that code is
	  failing, as it expects the channels to have a soft hangup flag
	  set on it indicating that a redirect has occurred or that the
	  channel is going to leave the bridge. (The code did not take into
	  account a smart bridge operation). This patch also renames a few
	  things to be more reflective of the underlying types. Review:
	  https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close
	  ........ Merged revisions 425760 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, tests/test_cel.c: test_cel: Update pickup test to expect
	  CANCEL instead of ANSWSER The CEL pickup test previously looked
	  for a disposition of ANSWER between the original caller/peer when
	  the call is picked up. This is actually incorrect: the
	  disposition should, at the very least, not be ANSWER as the call
	  was never ANSWERed. The disposition is now CANCEL; this patch
	  updates the test accordingly. ........ Merged revisions 425757
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: main/cdr: Use 'time' when rescheduling batched
	  CDRs as opposed to 'size' When refactoring CDRs to use the
	  configuration framework, a 'whoops' was introduced where the CDR
	  batch size was used when rescheduling a batch, as opposed to the
	  time duration. This patch corrects that obvious mistake.
	  ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged
	  revisions 425735 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-16 17:30 +0000 [r425714]  George Joseph <george.joseph@fairview5.com>

	* include/asterisk/config.h, tests/test_config.c, main/config.c, /:
	  config: Fix inf loop using ast_category_browse and
	  ast_variable_retrieve Fix infinite loop when calling
	  ast_variable_retrieve inside an ast_category_browse loop when
	  there is more than 1 category with the same name. Tested-by:
	  George Joseph Review: https://reviewboard.asterisk.org/r/4089/
	  ........ Merged revisions 425713 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-16 14:35 +0000 [r425691]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip_t38.c, res/res_pjsip_registrar_expire.c,
	  res/res_pjsip_mwi_body_generator.c,
	  res/res_pjsip_endpoint_identifier_user.c,
	  res/res_pjsip_send_to_voicemail.c,
	  include/asterisk/res_pjsip_pubsub.h,
	  res/res_pjsip_outbound_authenticator_digest.c,
	  res/res_pjsip_outbound_registration.c,
	  res/res_pjsip_endpoint_identifier_anonymous.c,
	  res/res_pjsip_path.c, res/res_pjsip_one_touch_record_info.c,
	  res/res_pjsip_acl.c, res/res_pjsip_pubsub.c,
	  res/res_pjsip_diversion.c, res/res_pjsip_refer.c,
	  include/asterisk/res_pjsip.h,
	  res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c,
	  res/res_pjsip_multihomed.c, res/res_pjsip_authenticator_digest.c,
	  res/res_pjsip_sdp_rtp.c, res/res_hep_pjsip.c,
	  res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c,
	  res/res_pjsip_logger.c, res/res_pjsip_nat.c,
	  res/res_pjsip_session.c, res/res_pjsip_exten_state.c,
	  res/res_pjsip_header_funcs.c, res/res_pjsip_rfc3326.c,
	  res/res_pjsip_phoneprov_provider.c, res/res_pjsip_mwi.c,
	  res/res_pjsip_dialog_info_body_generator.c,
	  res/res_pjsip_xpidf_body_generator.c, res/res_pjsip_registrar.c,
	  channels/chan_pjsip.c, res/res_pjsip_transport_websocket.c,
	  res/res_pjsip_pidf_eyebeam_body_supplement.c,
	  include/asterisk/res_pjsip_session.h, /, res/res_pjsip_notify.c,
	  res/res_pjsip_pidf_digium_body_supplement.c,
	  res/res_pjsip_endpoint_identifier_ip.c,
	  res/res_pjsip_publish_asterisk.c: PJSIP: Enforce module load
	  dependencies This enforces that res_pjsip, res_pjsip_session, and
	  res_pjsip_pubsub have loaded properly before attempting to load
	  any modules that depend on them since the module loader system is
	  not currently capable of resolving module dependencies on its
	  own. ASTERISK-24312 #close Reported by: Dafi Ni Review:
	  https://reviewboard.asterisk.org/r/4062/ ........ Merged
	  revisions 425690 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-16 06:11 +0000 [r425669]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c, /: Fix loss of voice after second call
	  drops (on a second line) in case using multiple lines on unistim
	  phones. There is regression was introduced in r391379. Reported
	  by: Rustam Khankishyiev (closes issue ASTERISK-23846) ........
	  Merged revisions 425667 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425668 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-16 01:25 +0000 [r425646]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix a bug where ICE
	  state would get reset when it shouldn't. In the case where the
	  ICE negotiation had not yet started current state would get wiped
	  when it shouldn't. This also removes channel binding as in
	  practice this does not work well with other implementations.
	  ........ Merged revisions 425644 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425645 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-15 19:31 +0000 [r425627]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_motif.c: chan_motif: Cleanup
	  jingle_tech.capabilities only once.

2014-10-15 19:05 +0000 [r425611]  Jonathan Rose <jrose@digium.com>

	* res/parking/parking_tests.c: parking_tests: Fix assertions and
	  possibly crashes in res_parking unit tests Assertions were caused
	  by attempting to play music on hold to a channel with no formats.
	  Parking unit test channels were given formats and a technology so
	  that they would be able to pretend to read/write frames.
	  ASTERISK-24413 #close Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/4075/

2014-10-15 09:59 +0000 [r425590]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: chan_ooh323: fix rtptimeout general
	  value checking correct condition to check rtptimeout in [general]
	  config section ASTERISK-24393 #close Reported by: Dmitry Melekhov
	  Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........
	  Merged revisions 425547 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425548 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425589 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-14 20:46 +0000 [r425526]  George Joseph <george.joseph@fairview5.com>

	* /, include/asterisk/config.h, tests/test_config.c, main/config.c:
	  config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG
	  the /main/config config_basic_ops test was causing a SEGV while
	  doing an ast_category_delete in an ast_category_browse loop.
	  Apparently this never worked but was also never tested. I removed
	  the test, added 2 notes to config.h indicating that it's not
	  supported and added a few lines of code to ast_category_delete to
	  prevent the SEGV should someone attempt it in the future.
	  Tested-by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/4078/ ........ Merged
	  revisions 425525 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-14 19:00 +0000 [r425504]  Jonathan Rose <jrose@digium.com>

	* main/sched.c, /: Scheduler: Fix a nasty scheduler caching bug
	  which makes new tasks not execute Tasks that were marked for
	  pending deletion in the scheduler would be moved to the cache for
	  later reuse, but after being recycled the deleted mark wouldn't
	  be removed resulting in fresh tasks being deleted without
	  reason... and immediately moved back into the cache where they
	  could be reused again. This could cause horrendous things to
	  happen in just about anything that used a scheduler.
	  ASTERISK-24321 #close Reported by: Steve Pitts Review:
	  https://reviewboard.asterisk.org/r/4071/ ........ Merged
	  revisions 425503 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-14 18:12 +0000 [r425481]  George Joseph <george.joseph@fairview5.com>

	* res/res_phoneprov.c, include/asterisk/phoneprov.h, /,
	  res/res_pjsip_phoneprov_provider.c: res_phoneprov: Create
	  accessor for ast_phoneprov_std_variable_lookup Based on feedback
	  from Richard, I created an accessor for
	  res_phoneprov/ast_phoneprov_std_variable_lookup and added load
	  priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by:
	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/
	  ........ Merged revisions 425480 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-14 16:46 +0000 [r425459]  Corey Farrell <git@cfware.com>

	* /, res/res_fax.c: res_fax: Fix reference leak caused by gateway
	  sessions Fax gateway session objects can be re-used, causing the
	  same gateway session to be added to faxregistry.container more
	  than once. This change causes fax_session_new to remove the
	  reserved session from the container before it's id is changed,
	  ensuring it's possible for the session to be freed.
	  ASTERISK-24392 #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/4049/ ........ Merged
	  revisions 425457 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425458 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-14 16:35 +0000 [r425455]  Richard Mudgett <rmudgett@digium.com>

	* /, main/stasis_channels.c: stasis_channels.c: Resolve unfinished
	  Dials when doing masquerades (Part 2) Masquerades into and out of
	  channels that are involved in a dial operation don't create the
	  expected dial end event. The missing dial end event goes against
	  the model for things like CDRs and generating Dial end manager
	  actions and such. There are four cases: 1) A channel masquerades
	  into the caller channel. The case happens when performing a
	  blonde transfer using the channel driver's protocol. 2) A channel
	  masquerades into a callee channel. The case happens when
	  performing a directed call pickup. 3) The caller channel
	  masquerades out of dial. The case happens when using the Bridge
	  application on the caller channel. 4) A callee channel
	  masquerades out of dial. The case happens when using the Bridge
	  application on a peer channel. As it turned out, all four cases
	  need to be handled instead of just the first one. ASTERISK-24237
	  Reported by: Richard Mudgett ASTERISK-24394 #close Reported by:
	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/
	  ........ Merged revisions 425430 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-14 16:19 +0000 [r425415]  Corey Farrell <git@cfware.com>

	* /, res/res_fax.c: res_fax: Resolve module reference leak caused
	  by reserved sessions Remove reference to module providing
	  reserved session after adding a reference to the final module.
	  This re-reference is done to ensure that module references are
	  correct even if the final session selects a different module than
	  the reserved session. ASTERISK-18923 #close Reported by: Grigoriy
	  Puzankin Review: https://reviewboard.asterisk.org/r/4048/
	  ........ Merged revisions 425405 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425407 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425411 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-13 16:10 +0000 [r425384]  George Joseph <george.joseph@fairview5.com>

	* apps/app_directory.c, tests/test_sorcery.c, main/config.c,
	  tests/test_sorcery_realtime.c, res/res_sorcery_realtime.c,
	  apps/app_voicemail.c, res/res_sorcery_config.c, main/manager.c,
	  /, include/asterisk/config.h, pbx/pbx_realtime.c,
	  tests/test_config.c: manager/config: Support templates and
	  non-unique category names via AMI This patch provides the
	  capability to manipulate templates and categories with non-unique
	  names via AMI. Summary of changes: GetConfig and GetConfigJSON:
	  Added "Filter" parameter: A comma separated list of
	  name_regex=value_regex expressions which will cause only
	  categories whose variables match all expressions to be
	  considered. The special variable name TEMPLATES can be used to
	  control whether templates are included. Passing 'include' as the
	  value will include templates along with normal categories.
	  Passing 'restrict' as the value will restrict the operation to
	  ONLY templates. Not specifying a TEMPLATES expression results in
	  the current default behavior which is to not include templates.
	  UpdateConfig: NewCat now includes options for allowing duplicate
	  category names, indicating if the category should be created as a
	  template, and specifying templates the category should inherit
	  from. The rest of the actions now accept a filter string as
	  defined above. If there are non-unique category names, you can
	  now update specific ones based on variable values. To facilitate
	  the new capabilities in manager, corresponding changes had to be
	  made to config, most notably the addition of filter criteria to
	  many of the APIs. In some cases it was easy to change the
	  references to use the new prototype but others would have
	  required touching too many files for this patch so a wrapper with
	  the original prototype was created. Macros couldn't be used in
	  this case because it would break binary compatibility with
	  modules such as res_digium_phone that are linked to real symbols.
	  Tested-by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/4033/ ........ Merged
	  revisions 425383 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-12 21:09 +0000 [r425362]  Joshua Colp <jcolp@digium.com>

	* /, res/res_rtp_asterisk.c: res_rtp_asterisk: Make the ICE
	  transport check case insensitive as some implementations use
	  'udp'. ........ Merged revisions 425360 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425361 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-12 08:15 +0000 [r425289-425299]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: chan_sip: Fix so asterisk won't send
	  reINVITE after a BYE. After a reINVITE glare situation, Asterisk
	  would re-send the reINVITE even though the call had been hung up
	  in the mean time. This patch unschedules the reinvite when
	  handling the BYE. ASTERISK-22791 #close Reported by: Paolo
	  Compagnini Tested by: Paolo Compagnini Review:
	  https://reviewboard.asterisk.org/r/4056/ (testcase is in review
	  r4055) ........ Merged revisions 425296 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425297 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425298 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, Makefile: build: Relax badshell tilde test to allow for ~ in
	  middle of DESTDIR. The main Makefile has a target test called
	  'badshell' that tests if DESTDIR does not happen to have an
	  an-expanded tilde (~). This might be the case if you run: make
	  install DESTDIR=~/somewhere/ That test also disallowed valid
	  tildes in directory names. The test is now changed to only
	  trigger on a tilde at the start of the path. ASTERISK-13797
	  #close Reported by: Tzafrir Cohen Review:
	  https://reviewboard.asterisk.org/r/4064/ ........ Merged
	  revisions 425291 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425292 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425293 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_calendar_ews.c: res_calendar_ews: Relax neon version
	  check to work with 0.30 too. Allow res_calendar_ews to work not
	  only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close
	  Reported by: Tzafrir Cohen Review:
	  https://reviewboard.asterisk.org/r/4068/ ........ Merged
	  revisions 425286 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425287 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425288 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-11 21:08 +0000 [r425265]  George Joseph <george.joseph@fairview5.com>

	* /, res/res_phoneprov.c: res_phoneprov: Cleanup module load error
	  handling Tested module load/reload interaction between
	  res_phoneprov and res_pjsip_phoneprov_provider in cases where
	  res_phoneprov didn't load correctly (usually misconfiguration or
	  missing phoneprov.conf) Tested-by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/4069/ ........ Merged
	  revisions 425264 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-10 20:48 +0000 [r425243]  Joshua Colp <jcolp@digium.com>

	* /, main/bridge.c, bridges/bridge_native_rtp.c: bridge: During a
	  smart bridge operation provide a more complete bridge to the old
	  technology. When a smart bridge operation occurs and a bridge
	  transitions from one technology to another the old technology is
	  provided the channels formerly in it and told that they are
	  leaving. Unfortunately the bridge provided along with them is
	  incomplete. The bridge, despite there being channels in it,
	  contains none. This forces technology implementations to have
	  additional logic when channels are leaving or to store their own
	  duplicated state. This change makes the bridge more complete so
	  it contains the expected channels. Now that the bridge is
	  complete special logic within bridge_native_rtp is no longer
	  needed and has been removed. Review:
	  https://reviewboard.asterisk.org/r/4057/ ........ Merged
	  revisions 425242 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-10 14:31 +0000 [r425221]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_phoneprov.c: res/res_phoneprov: Bail on registration
	  if res_phoneprov didn't load If res_phoneprov failed to fully
	  load (due to not being configured), the providers container will
	  be NULL. If a module attempts to register a phone provisioning
	  provider, it should check for the presence of the container. If
	  there is no providers container, it should return an error. This
	  patch makes the ast_phoneprov_provider_register function do
	  that... otherwise this would be a silly commit message. ........
	  Merged revisions 425220 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-10 14:23 +0000 [r425217]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_phoneprov_provider.c:
	  res_pjsip_phoneprov_provider: Add missing dependency on
	  pjproject. ........ Merged revisions 425216 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-10 13:01 +0000 [r425155]  Kinsey Moore <kmoore@digium.com>

	* /, tests/test_callerid.c, main/callerid.c: CallerID: Fix parsing
	  regression This fixes a regression in callerid parsing introduced
	  when another bug was fixed. This bug occurred when the name was
	  composed entirely of DTMF keys and quoted without a number
	  section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard
	  Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by
	  Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/
	  ........ Merged revisions 425152 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425153 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425154 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-10 12:10 +0000 [r425132]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_nat.c, /: res_pjsip_nat: Place source port into
	  rport of responses if 'force_rport' is on. When the 'force_rport'
	  option is enabled the behavior should be the same as if the
	  remote side placed rport into the message themselves. Therefore
	  any responses we send should include the source port of the
	  request in the rport of the Via header. #SIPit31 ASTERISK-24387
	  #close Reported by: Matt Jordan ........ Merged revisions 425131
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-10 07:32 +0000 [r425071]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: chan_sip: Fix dialog leak resulting from
	  missing ACK to re-INVITE. If a device re-INVITEs at the same time
	  as the dialog is hung up, and if then the ACK to the re-INVITE
	  never reaches Asterisk, chan_sip would fail to destroy the dialog
	  after a while. This resulted in (most prominently) file handle
	  leaks. (Patch reindented by me.) ASTERISK-20784 #close
	  ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal
	  Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle
	  (License #5334) patch_asterisk_20784.txt uploaded by Nitesh
	  Bansal (License #6418) Reviewboard:
	  https://reviewboard.asterisk.org/r/4052/ (testcase can be found
	  at r4051) ........ Merged revisions 425068 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 425069 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 425070 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-09 23:35 +0000 [r425052]  George Joseph <george.joseph@fairview5.com>

	* res/res_pjsip_phoneprov_provider.c: res_pjsip_phoneprov_provider:
	  fix compile breakage on AST_VECTOR endpoint->inbound_auths was
	  changed to a vector in 13 and I committed the 12 patch instead of
	  the 13 patch. Tested-by: George Joseph

2014-10-09 21:38 +0000 [r425031]  Kevin Harwell <kharwell@digium.com>

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Crash if no
	  candidates received for component When starting ice if there is
	  not at least one remote ice candidate with an RTP component
	  asterisk will crash. This is due to an assertion in pjnath as it
	  expects at least one candidate with an RTP component. Added a
	  check to make sure at least one candidate contains an RTP
	  component and at least one candidate has an RTCP component.
	  ASTERISK-24383 #close Review:
	  https://reviewboard.asterisk.org/r/4039/ ........ Merged
	  revisions 425030 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-09 20:54 +0000 [r425008]  George Joseph <george.joseph@fairview5.com>

	* /, res/res_pjsip_phoneprov_provider.c (added),
	  configs/samples/pjsip.conf.sample: res_pjsip_phoneprov_provider:
	  Provides pjsip integration with res_phoneprov This module allows
	  res_pjsip to integrate with res_phoneprov. It handles the pjsip
	  'phoneprov' object type. Tested-by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3976/ ........ Merged
	  revisions 425007 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-09 18:37 +0000 [r424986]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_phoneprov.c: res/res_phoneprov: Don't cancel Asterisk
	  load on module load failure ........ Merged revisions 424985 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-09 17:45 +0000 [r424964]  George Joseph <george.joseph@fairview5.com>

	* include/asterisk/phoneprov.h (added), /,
	  configs/samples/phoneprov.conf.sample,
	  include/asterisk/chanvars.h, res/res_phoneprov.c,
	  res/res_phoneprov.exports.in (added), main/chanvars.c:
	  res_phoneprov: Refactor phoneprov to allow pluggable config
	  providers This patch makes res_phoneprov more modular so other
	  modules (like pjsip) can provide configuration information
	  instead of res_phoneprov relying solely on users.conf and
	  sip.conf. To accomplish this a new ast_phoneprov public API is
	  now exposed which allows config providers to register themselves,
	  set defaults (server profile, etc) and add user extensions. *
	  ast_phoneprov_provider_register registers the provider and
	  provides callbacks for loading default settings and loading
	  users. * ast_phoneprov_provider_unregister clears the defaults
	  and users. * ast_phoneprov_add_extension should be called once
	  for each user/extension by the provider's load_users callback to
	  add them. * ast_phoneprov_delete_extension deletes one extension.
	  * ast_phoneprov_delete_extensions deletes all extensions for the
	  provider. Tested-by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3970/ ........ Merged
	  revisions 424963 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-09 16:36 +0000 [r424942]  Richard Mudgett <rmudgett@digium.com>

	* /, main/cdr.c: cdr.c: Make turning on CDR debug a one step
	  process instead of two. Now "cdr set debug on" doesn't also
	  require "core set verbose 1" to see CDR debug output. ........
	  Merged revisions 424941 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-09 08:08 +0000 [r424880]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, contrib/scripts/safe_asterisk: safe_asterisk: Don't
	  automatically exceed MAXFILES value of 2^20. On systems with lots
	  of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can
	  exceed the per-process file limit of 2^20. This patch ensures the
	  value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close
	  Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff
	  uploaded by Michael Myles (License #6626) ........ Merged
	  revisions 424875 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 424878 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424879 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-08 18:46 +0000 [r424854]  Joshua Colp <jcolp@digium.com>

	* /, res/res_rtp_asterisk.c: res_rtp_asterisk: Allow only UDP ICE
	  candidates. The underlying library, pjnath, that res_rtp_asterisk
	  uses for ICE support does not have support for ICE-TCP. As
	  candidates are passed through directly to it this can cause error
	  messages to occur when it receives something unexpected (such as
	  a TCP candidate). This change merely ignores all non-UDP
	  candidates so they never reach pjnath. ASTERISK-24326 #close
	  Reported by: Joshua Colp ........ Merged revisions 424852 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424853 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-08 18:24 +0000 [r424769-424850]  Kinsey Moore <kmoore@digium.com>

	* main/stasis.c: Stasis: Relegate log message to dev-mode This
	  error message primarily applies to development tasks and will now
	  only show up when dev-mode is enabled via configure.

	* main/sounds_index.c: Indexer: Format message types may not exist
	  In Asterisk 13+, any given message type is not guaranteed to
	  exist even if Asterisk comes up correctly since creation of the
	  message type could be declined. The indexer should not prevent
	  Asterisk from starting under these conditions.

	* main/stasis.c: Stasis: Only log errors for non-declined types
	  When message type creation is declined via stasis.conf, certain
	  operations log errors assuming that the declined type is being
	  used before initialization or after destruction. These error
	  messages get quite spammy for oft used message types and should
	  not be logged in the first place since the message type is
	  validly NULL. Reported by: Matt DiMeo

2014-10-07 18:33 +0000 [r424752]  Joshua Colp <jcolp@digium.com>

	* main/data.c: data: Properly access formats in capabilities
	  structure when adding codecs. Formats within a capabilities
	  structure are addressed starting at 0, not 1. Assuming 1 causes
	  it to exceed an array. ASTERISK-24389 #close Reported by: Kevin
	  Harwell

2014-10-07 17:41 +0000 [r424692-424731]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_pjsip_outbound_registration.c:
	  res/res_pjsip_outbound_registration: Initialize
	  auth_reject_permanent parameter Prior to this patch, the
	  auth_reject_permanent parameter was not initialized on the
	  registration client state, leading to the parameter being
	  disabled regardless of the value specified in pjsip.conf. This
	  patch initialized the setting on the registration client state to
	  the provided configuration value. ASTERISK-24398 #close ........
	  Merged revisions 424730 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Fix typo in WARNING
	  message

	* main/message.c, /: message: Don't close an AMI connection on
	  SendMessage action error If SendMessage encounters an error (such
	  as incorrect input provided to the action), it will currently
	  return -1. Actions should only return -1 if the connection to the
	  AMI client should be closed. In this case, SendMessage causing
	  the client to disconnect is inappropriate. This patch causes the
	  action to return 0, which simply causes the action to fail.
	  Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354
	  #close Reported by: Peter Katzmann patches: sendMessage.patch
	  uploaded by Peter Katzmann (License 5968) ........ Merged
	  revisions 424690 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424691 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-06 15:38 +0000 [r424669]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c, /: features.c: Fix lingering channel ref while
	  Bridge() application is active. Using the Bridge application to
	  bridge a channel that is executing an applicaiton such as Wait
	  results in a lingering Surrogate channel in the CLI "core show
	  channels" output even though it has already hungup. * Fix
	  bridge_exec() to not hold onto the current_dest_chan ref once it
	  has been put into the bridge. * Eliminated bridge_exec()'s use of
	  RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged
	  revisions 424668 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-06 12:38 +0000 [r424601-424647]  Matthew Jordan <mjordan@digium.com>

	* /, main/sdp_srtp.c: sdp_srtp: Add new lines to some WARNING
	  messages ........ Merged revisions 424646 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip/pjsip_options.c: res_pjsip/pjsip_options: Do not
	  404 an OPTIONS request not sent to an endpoint An OPTIONS request
	  that is sent to Asterisk but not to a specific endpoint is
	  currently sent a 404 in response. This is because, not
	  surprisingly, an empty extension is never going to be found in
	  the dialplan. This patch makes it so that we only attempt to look
	  up the endpoint in the dialplan if it is specified in the OPTIONS
	  request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt
	  Jordan ........ Merged revisions 424624 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions:
	  Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels Calling
	  PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your
	  health. It will treat the channels as a PJSIP channel, eventually
	  hitting an ao2 error, FRACKing on assertion error, and quite
	  likely crashing. This patch adds checks to the read/write
	  callbacks that ensure that the channel technology is of type
	  'PJSIP' before attempting to operate on the channel. #SIPit31
	  ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged
	  revisions 424621 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_hep_pjsip.c, res/res_pjsip/pjsip_distributor.c,
	  res/res_pjsip_logger.c: res_pjsip: Prevent crashes when PJPROJECT
	  presents an rdata with no message When a message that exceeds the
	  PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible
	  (although it shouldn't occur) for pjproject to pass up an rdata
	  object with a NULL msg in the msg_info. Needless to say, things
	  that attempt to dereference this are in for a rough ride. In
	  particular, this caused crashes in three different locations, all
	  of which are 'low level' enough to intercept an rdata object
	  early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3)
	  res_pjsip/distributor Anything that can intercept an rdata object
	  before res_pjsip/distributor should be defensive when looking at
	  the received packet. #SIPit31 ASTERISK-24369 #close Reported by:
	  Matt Jordan ........ Merged revisions 424618 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Gracefully handle
	  errors when re-creating subscriptions A subscription that has
	  been persisted can - for various reasons - fail to be re-created
	  on startup. This patch resolves a number of crashes that occurred
	  when a subscription cannot be re-created on several off-nominal
	  paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan

2014-10-05 00:48 +0000 [r424552-424580]  Corey Farrell <git@cfware.com>

	* main/manager.c, /: Release AMI connections on shutdown.
	  ASTERISK-24378 #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/4037/ ........ Merged
	  revisions 424578 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424579 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_motif.c: chan_motif: Correct last commit to use
	  ao2_cleanup to free format cap This fix applies to 13 and trunk.
	  ASTERISK-24384 #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/4043/

	* /, channels/chan_motif.c: chan_motif: Release format capabilities
	  and config on module load error ASTERISK-24384 #close Reported
	  by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/4043/ ........ Merged
	  revisions 424550 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424551 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-03 21:56 +0000 [r424472-424529]  Richard Mudgett <rmudgett@digium.com>

	* /, CHANGES, res/res_pjsip.c: res_pjsip: Fix XML typo and update
	  CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, /,
	  main/framehook.c: audiohooks: Reevaluate the bridge technology
	  when an audiohook is added or removed. Adding a mixmonitor to a
	  channel causes the bridge to change technologies from native to
	  simple_bridge so the call can be recorded. However, when the
	  mixmonitor is stopped the bridge does not switch back to the
	  native technology. * Added unbridge requests to reevaluate the
	  bridge when a channel audiohook is removed. * Moved the unbridge
	  request into ast_audiohook_attach() ensure that the bridge
	  reevaluates whenever an audiohook is attached. This simplified
	  the mixmonitor and chan_spy start code as well. * Added defensive
	  code to stop_mixmonitor_full() in case additional arguments are
	  ever added to the StopMixMonitor application. * Made
	  ast_framehook_detach() not do an unbridge request if the
	  framehook does not exist. * Made ast_framehook_list_fixup() do an
	  unbridge request if there are any framehooks. Also simplified the
	  loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/4046/ ........ Merged
	  revisions 424506 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/core_unreal.c, main/taskprocessor.c, channels/chan_iax2.c,
	  res/res_pjsip_session.c, main/channel.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, funcs/func_frame_trace.c,
	  channels/chan_motif.c, include/asterisk/frame.h,
	  main/bridge_channel.c, channels/chan_pjsip.c,
	  channels/chan_unistim.c, include/asterisk/res_pjsip_session.h,
	  addons/chan_ooh323.c, /, include/asterisk/taskprocessor.h,
	  channels/chan_sip.c, res/res_pjsip_session.exports.in:
	  chan_pjsip: Fix deadlock when masquerading PJSIP channels.
	  Performing a directed call pickup resulted in a deadlock when
	  PJSIP channels were involved. A masquerade needs to hold onto the
	  channel locks while it swaps channel information between the two
	  channels involved in the masquerade. With PJSIP channels, the
	  fixup routine needed to push a fixup task onto the PJSIP
	  channel's serializer. Unfortunately, if the serializer was also
	  processing a task that needed to lock the channel, you get
	  deadlock. * Added a new control frame that is used to notify the
	  channels that a masquerade is about to start and when it has
	  completed. * Added the ability to query taskprocessors if the
	  current thread is the taskprocessor thread. * Added the ability
	  to suspend/unsuspend the PJSIP serializer thread so a masquerade
	  could fixup the PJSIP channel without using the serializer.
	  ASTERISK-24356 #close Reported by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/4034/ ........ Merged
	  revisions 424471 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-03 15:54 +0000 [r424448]  George Joseph <george.joseph@fairview5.com>

	* /, main/sorcery.c: sorcery: Prevent SEGV in sorcery_wizard_create
	  when there's no create function When you call
	  ast_sorcery_create() you don't necessarily know which wizard is
	  going to be invoked. If it happens to be a wizard like 'config'
	  that doesn't have a 'create' virtual function you get a segfault
	  in the sorcery_wizard_create callback. This patch catches the
	  null function pointer, does an ast_assert, and logs an error.
	  Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged
	  revisions 424447 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-03 13:58 +0000 [r424424-424427]  Kinsey Moore <kmoore@digium.com>

	* configs/samples/pjsip.conf.sample, /,
	  res/res_pjsip/pjsip_configuration.c: PJSIP: Restore functional
	  default for callerid_privacy The pjsip config option default
	  fixups from r424263 altered the functional default from
	  "allowed_not_screened" to "allowed". This change restores the
	  functional default value when none is provided. ........ Merged
	  revisions 424426 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/manager.c, /: Manager: Add missing fields and documentation
	  for CoreShowChannels This corrects some issues introduced in the
	  responses to the CoreShowChannels AMI command as well as adding
	  documentation for the responses. The command in Asterisk 12 was
	  missing the following fields: Duration, Application,
	  ApplicationData, and BridgedChannel and BridgedUniqueID (replaced
	  with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn
	  Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged
	  revisions 424423 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-03 07:54 +0000 [r424415]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_session.c, /: res_pjsip_session: Reduce SDP size by
	  removing duplicate connection lines. Due to the architecture of
	  how media streams are handled each individual handler adds
	  connection details (IP address) for it. The first media stream is
	  then used as the top level SDP connection line. In practice each
	  line ends up being the same so to reduce the SDP size
	  stream-level connection information is also added to the SDP if
	  it differs from the top level SDP connection line. ........
	  Merged revisions 424414 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-02 21:52 +0000 [r424394]  Richard Mudgett <rmudgett@digium.com>

	* /, configs/samples/pjsip.conf.sample, res/res_pjsip.c,
	  res/res_pjsip/config_transport.c: res_pjsip: Make transport
	  cipher option accept a comma separated list of cipher names.
	  Improvements to the res_pjsip transport cipher option. * Made the
	  cipher option accept a comma separated list of OpenSSL cipher
	  names. Users of realtime will be glad if they have more than one
	  name to list. * Added the CLI command 'pjsip list ciphers' so a
	  user can know what OpenSSL names are available for the cipher
	  option. * Updated the cipher option online XML documentation to
	  specify what is expected for the value. * Updated
	  pjsip.conf.sample to not indicate that ALL is acceptable since
	  ALL does not imply a preference order for the ciphers and PJSIP
	  does not simply pass the string to OpenSSL for interpretation.
	  ASTERISK-24199 #close Reported by: Joshua Colp Review:
	  https://reviewboard.asterisk.org/r/4018/ ........ Merged
	  revisions 424393 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-02 20:15 +0000 [r424373]  Jonathan Rose <jrose@digium.com>

	* /,
	  contrib/ast-db-manage/config/versions/10aedae86a32_add_outgoing_enum_va.py
	  (added): Alembic: Add enumerator value to sippeers -> directmedia
	  - 'outgoing' The 'outgoing' value was left off of the enumerator
	  when first creating the column. This patch adds it, and should
	  gracefully upgrade keeping the existing data in tact.
	  ASTERISK-23781 #close Reported by: Stephen More Review:
	  https://reviewboard.asterisk.org/r/4013/ ........ Merged
	  revisions 424372 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-02 13:35 +0000 [r424338]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, configs/samples/pjsip.conf.sample: res_pjsip: document use of
	  rewrite_contact in sample conf Without setting rewrite_contact,
	  an invite to an endpoint behind NAT will not reach it - unless
	  the endpoint itself uses STUN or TURN to discover it's public
	  URI. Thus, the use of this should be in the sample documentation.
	  Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged
	  revisions 424337 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-01 22:52 +0000 [r424333]  Jonathan Rose <jrose@digium.com>

	* channels/chan_pjsip.c: chan_pjsip: Fix an assertion for channels
	  that lack formats on creation ASTERISK-24222 #close Reported by:
	  Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/

2014-10-01 20:36 +0000 [r424313]  Corey Farrell <git@cfware.com>

	* res/res_hep.c, /: res_hep: Release allocation reference to
	  configuration. ASTERISK-24362 #close Reported by: Corey Farrell
	  Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged
	  revisions 424312 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-01 16:37 +0000 [r424288-424291]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip/pjsip_configuration.c,
	  configs/samples/pjsip.conf.sample, res/res_pjsip.c: res_pjsip:
	  Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.
	  During the latest update to DTLS-SRTP support the ability to
	  configure the hash used for fingerprints was added. This gave us
	  two supported ones: SHA-1 and SHA-256. The default was
	  accordingly updated to SHA-256. Unfortunately this configuration
	  ability was not exposed within res_pjsip. This change adds a
	  dtls_fingerprint option that controls it. #SIPit31 ........
	  Merged revisions 424290 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Accept DTLS
	  attributes in top level, not just media session. #SIPit31
	  ........ Merged revisions 424287 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-01 12:27 +0000 [r424245-424266]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip/config_transport.c, /, res/res_pjsip/location.c,
	  res/res_pjsip_endpoint_identifier_ip.c,
	  res/res_pjsip/pjsip_configuration.c,
	  configs/samples/pjsip.conf.sample: PJSIP: Handle defaults
	  properly This updates the code behind PJSIP configuration options
	  with custom handlers to deal with the assigned default values
	  properly where it makes sense and adjusting the default value
	  where it doesn't. Before applying this patch, there were several
	  cases where the default value for an option would prevent that
	  config section from loading properly. Reported by: Thomas
	  Thompson Review: https://reviewboard.asterisk.org/r/4019/
	  ........ Merged revisions 424263 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_nat.c: PJSIP: Force transport on contact rewrite
	  If contact rewriting is enabled but the contact differs in
	  transport from what is actually being used, messages after the
	  initial INVITE transaction can be sent to an incorrect
	  transport/port combination. In the case where this bug occurred
	  the remote party never received a BYE since it was sent to the
	  remote party's TCP port over UDP. Review:
	  https://reviewboard.asterisk.org/r/4032/ ........ Merged
	  revisions 424244 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-10-01 10:09 +0000 [r424179-424184]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: chan_sip: Simplify some unref code by
	  removing unlink_peer_from_tables. ASTERISK-22945 #related
	  Reported by: ibercom Patches:
	  asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License
	  #6599) ........ Merged revisions 424181 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 424182 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424183 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: chan_sip: Remove excess ref of realtime
	  peer before sip_poke_peer. The peer is referenced at the end of
	  sip_poke_peer, it should not get an extra ref before the call to
	  sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close
	  Reported by: ibercom Tested by: Yuriy Gorlichenko Patches:
	  asterisk11.patch uploaded by ibercom (License #6599) Review:
	  https://reviewboard.asterisk.org/r/4031/ ........ Merged
	  revisions 424176 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 424177 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424178 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-30 11:40 +0000 [r424153-424156]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't place an
	  extra whitespace before 'rport' and don't put IPv6 addresses in
	  brackets. #SIPit31 ........ Merged revisions 424155 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the base
	  and mapped address for candidates is present in SDP. This change
	  fixes an issue where ICE candidates put into the SDP did not
	  contain the 'raddr' and 'rport' information for server reflexive
	  and relay candidates. #SIPit31 ........ Merged revisions 424151
	  from http://svn.asterisk.org/svn/asterisk/branches/11 ........
	  Merged revisions 424152 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-29 21:59 +0000 [r424129]  George Joseph <george.joseph@fairview5.com>

	* /, res/res_pjsip/pjsip_cli.c: pjsip_cli: Suppress header print on
	  error or no objects If there's an error on the pjsip command line
	  or there are no objects, don't print the column headers.
	  ASTERISK-24350 #close Reported-by: Brad Latus Tested-by: George
	  Joseph Tested-by: Brad Latus Review:
	  https://reviewboard.asterisk.org/r/4025/ ........ Merged
	  revisions 424128 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-29 21:26 +0000 [r424126]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, contrib/scripts/autosupport: autosupport: Fix bashism. '==' is
	  bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
	  'case' works better there. Originally committed in r375059 and
	  r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported
	  by: Tzafrir Cohen ........ Merged revisions 424117 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 424125 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-29 21:17 +0000 [r424097-424105]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c,
	  /, res/res_pjsip_authenticator_digest.c: Simplify UUID generation
	  in several places. Replace code using ast_uuid_generate() with
	  simpler and faster code using ast_uuid_generate_str(). The new
	  code avoids a malloc(), free(), and copy. ........ Merged
	  revisions 424103 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/threadpool.c: threadpool.c: Minor cleanup fixes. * Fix
	  threadpool_alloc() prototype. * Add missing off-nominal NULL
	  check of pool in threadpool_alloc(). * searializer_create() does
	  not need to create the object with a lock as the lock is not
	  used. ........ Merged revisions 424096 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-27 12:43 +0000 [r424057]  Joshua Colp <jcolp@digium.com>

	* channels/chan_pjsip.c, res/res_pjsip_session.c, /:
	  res_pjsip_session: Add additional checks for delaying session
	  refreshes. There are certain situations which no checks existed
	  for which need to prevent session refreshes. This includes
	  sending a session refresh with SDP before SDP negotiation has
	  completed and sending a session refresh before the dialog itself
	  has been established. Checks for these have been added.
	  Additionally COLP related UPDATEs were including SDP when it is
	  not needed. Review: https://reviewboard.asterisk.org/r/4008/
	  ........ Merged revisions 424056 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-26 15:21 +0000 [r423992]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_fax.c: res_fax: Fix out of bounds error in
	  update_modem_bits(). ASTERISK-24357 #close Reported by: Jeremy
	  Laine Patches: res_fax_bounds.patch (license #6561) patch
	  uploaded by Jeremy Laine Modified patch to not use magic numbers.
	  ........ Merged revisions 423979 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 423983 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423987 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-26 08:25 +0000 [r423918]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, doc/asterisk.8: docs: Escape unescaped minus sign in
	  asterisk.8 manpage. ASTERISK-23768 #close Reported by: Jeremy
	  Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy
	  Lainé (License #6561) ........ Merged revisions 423915 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 423916 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423917 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-25 21:01 +0000 [r423895]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip.c, /: res_pjsip.c: Add missing off nominal cleanup
	  in ast_sip_push_task_synchronous(). * Made memset the std struct
	  in ast_sip_push_task_synchronous() because if DEBUG_THREADS is
	  enabled then uninitialized lock tracking data is used. ........
	  Merged revisions 423894 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-24 18:32 +0000 [r423867]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c:
	  pjsip_options.c: Fix race condition stopping periodic out of
	  dialog OPTIONS request. The crash on the issues is a result of an
	  invalid transport configuration change when asterisk is
	  restarted. The attempt to send the qualify request fails and we
	  cleaned up. However, the callback is also called which results in
	  a double unref of the objects involved. * Put a wrapper around
	  pjsip_endpt_send_request() to detect when the passed in callback
	  is called because of an error so callers can know to not cleanup.
	  * Made send_request_cb() able to handle repeated challenges (Up
	  to 10). * Fix periodic endpoint qualify OPTIONS sched deletion
	  race by avoiding it. The sched entry will no longer self stop and
	  must be externally stopped. * Added REF_DEBUG description tags to
	  struct sched_data in pjsip_options.c. * Fix some off-nominal ref
	  leaks in schedule_qualify(), qualify_and_schedule(). * Reordered
	  pjsip_options.c module start/stop code to cleanup better on
	  error. ASTERISK-24295 #close Reported by: Rogger Padilla Review:
	  https://reviewboard.asterisk.org/r/3954/ ........ Merged
	  revisions 423866 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-24 08:53 +0000 [r423803]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: chan_sip: Unref outbound proxy structure
	  on dialog/pvt destruction. Make sure outbound proxy refs are
	  always unreffed on dialog destruction. Review:
	  https://reviewboard.asterisk.org/r/4016/ ........ Merged
	  revisions 423800 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 423801 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423802 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-23 14:29 +0000 [r423783]  Mark Michelson <mmichelson@digium.com>

	* tests/test_cel.c, tests/test_cdr.c: Make CDR and CEL unit tests
	  less FRACKy. Prior to this commit, CDR and CEL tests were
	  expected to trigger FRACKs (i.e. assertions) due to the fact that
	  the channels they create have no formats on them. Some code was
	  independently added recently that attempts to prevent FRACKs from
	  occurring by failing early when attempting to set up translation
	  paths if one or both channels support no formats. Unfortunately,
	  this attempt to be helpful made the CDR and CEL tests go from
	  simply FRACKing to outright failing and in some cases, failing so
	  badly as to crash Asterisk. This commit seeks to correct past
	  mistakes by adding the ulaw format to channels created by the CDR
	  and CEL unit tests. This makes setting up translation paths
	  succeed, eliminates previously-seen FRACKs, and ultimately causes
	  the unit tests to succeed again. Review:
	  https://reviewboard.asterisk.org/r/4014

2014-09-22 19:48 +0000 [r423660-423723]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: chan_sip: On INVITE retransmission, don't
	  add an extra 503 response. INVITE arrives to asterisk, asterisk
	  responds Busy(). If the INVITE is retransmitted, asterisk would
	  generate a 503 in addition to the 486. Thanks Torrey Searle for
	  providing a working regression test. ASTERISK-24335 #close
	  Review: https://reviewboard.asterisk.org/r/4003/ Patches:
	  retrans_486_invite.patch uploaded by Torrey Searle (License
	  #5334) ........ Merged revisions 423720 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 423721 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423722 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/editline/readline.c: cli.c: Fix tab completion "module
	  load" when MALLOC_DEBUG is enabled. r421600 conflicted with
	  r155763. ASTERISK-24348 #close ........ Merged revisions 423657
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 423658 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423659 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-21 01:15 +0000 [r423618-423641]  Matthew Jordan <mjordan@digium.com>

	* main/channel.c: main/channel: Unlock channel in off-nominal path
	  In r423414 (13) / r423415 (trunk), an API call that determines if
	  a format capability structure is empty was added. This returns
	  true if the format capability structure is completely empty or
	  "none". A check for this was added in channel.c's set_format
	  call. Unfortunately, when this check was true, it returned from
	  the function while still holding the channel lock. This caused
	  the CDR unit tests - which have a tendency to create channels
	  with no formats - to deadlock. Whoops. This patch unlocks the
	  channel on the off-nominal path.

	* rest-api/api-docs/events.json, /: rest-api/api-docs/events.json:
	  Remove non-compliant 'extends' attribute Prior to the release of
	  Swagger 1.2, the attribute 'extends' was being promoted as a
	  possible way to show that a particular object extends an existing
	  object. Instead, the Swagger specification went with the
	  'subTypes' attribute in the base object. This patch removes the
	  unsupported attribute; the object that the offending objects
	  proposed to extend already lists them in its 'subTypes'
	  attribute. ASTERISK-24300 #close Reported by: Bradley Watkins
	  ........ Merged revisions 423620 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
	  rest-api/api-docs/bridges.json,
	  rest-api/api-docs/recordings.json,
	  rest-api/api-docs/deviceStates.json,
	  rest-api/api-docs/endpoints.json,
	  rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
	  /, rest-api/api-docs/asterisk.json,
	  rest-api/api-docs/applications.json,
	  rest-api/api-docs/playbacks.json: rest-api/api-docs: Correct
	  basePath in resources to match top resources file The
	  resources.json file that defines the resource JSON files used
	  with ARI references a basePath of 'http://localhost:8088/ari'.
	  This does not match what is defined in the resource files
	  themselves, 'http://localhost:8088/stasis'. The correct base path
	  is the one that includes 'ari' in the URL; this patch updates the
	  various resource JSON files to have the correct basePath.
	  ASTERISK-24339 #close Reported by: Bradley Watkins ........
	  Merged revisions 423617 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-19 19:51 +0000 [r423580]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_notify.c: res_pjsip_notify: Fix crash on
	  unload/load and don't say the module doesn't exist on reload.
	  When unloading the module did not unregister the CLI commands
	  causing a crash upon load when they were registered again. When
	  reloading the module the return value from the config options
	  framework was not checked to determine if an error occurred or
	  not. This caused a message to be output saying the module did not
	  exist when reloading if no changes were present. AST-1433 #close
	  AST-1434 #close ........ Merged revisions 423579 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-19 17:08 +0000 [r423561]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_pjsip.c, res/res_pjsip_sdp_rtp.c:
	  res_pjsip_sdp_rtp.c: Fix native formats containing formats that
	  were not negotiated. Outgoing PJSIP calls can result in
	  non-negotiated formats listed in the channel's native formats if
	  video formats are listed in the endpoint's configuration. The
	  resulting call could then use a non-negotiated format resulting
	  in one way audio. * Simplified the update of session->req_caps in
	  set_caps(). Why do something in five steps when only one is
	  needed? AFS-162 #close Review:
	  https://reviewboard.asterisk.org/r/4000/

2014-09-19 15:18 +0000 [r423524-423530]  Jonathan Rose <jrose@digium.com>

	* /, main/stasis_channels.c: Stasis_channels: Resolve unfinished
	  Dials when doing masquerades Masquerades into channels that are
	  in the dialing state don't end their dial and this goes against
	  the model for things like CDRs and generating Dial end manager
	  actions and such. ASTERISK-24237 #close Reported by: Richard
	  Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........
	  Merged revisions 423525 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_iax2.c: chan_iax2: Fix a crash when using chan_iax2
	  jitterbuffer settings Caused by format changes in Asterisk 13
	  ASTERISK-24265 #close Reported by: Dafi Ni Review:
	  https://reviewboard.asterisk.org/r/3999/

2014-09-19 12:45 +0000 [r423504]  Kinsey Moore <kmoore@digium.com>

	* include/asterisk/framehook.h, /, main/framehook.c,
	  res/res_pjsip_t38.c: PJSIP: Prevent T38 framehook being put on
	  wrong channel This change gives framehooks a reverse-direction
	  masquerade callback in addition to chan_fixup_cb similar to the
	  callback added to datastores to handle the same situation. The
	  new callback provides the same parameters as the fixup callback,
	  but is called on the new channel's framehooks before moving
	  framehooks from the old channel to the new channel. This gives
	  the framehooks an oppurtunity to decide whether they should
	  remain on the new channel or be removed. This new callback is
	  used to prevent the PJSIP T.38 framehook from remaining on a
	  masqueraded channel if the new channel is not also a PJSIP
	  channel. This was causing a crash when a local channel was
	  masqueraded into a PJSIP channel and the framehook was executed
	  on the local channel since the channel's tech private data was
	  not structured as expected. Review:
	  https://reviewboard.asterisk.org/r/4001/ ........ Merged
	  revisions 423503 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-18 19:30 +0000 [r423482]  Sean Bright <sean@malleable.com>

	* res/res_pjsip/config_auth.c, /: res_pjsip: Don't require a
	  password when doing userpass authentication. An empty password is
	  valid for username/password authentication so we should allow
	  password to be empty/not supplied. Review:
	  https://reviewboard.asterisk.org/r/3988 ........ Merged revisions
	  423481 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-18 19:22 +0000 [r423478]  George Joseph <george.joseph@fairview5.com>

	* tests/test_strings.c, /, main/utils.c,
	  include/asterisk/strings.h: utils: Create ast_strsep function
	  that ignores separators inside quotes This function acts like
	  strsep with three exceptions... * The separator is a single
	  character instead of a string. * Separators inside quotes are
	  treated literally instead of like separators. * You can elect to
	  have leading and trailing whitespace and quotes stripped from the
	  result and have '\' sequences unescaped. Like strsep, ast_strsep
	  maintains no internal state and you can call it recursively using
	  different separators on the same storage. Also like strsep, for
	  consistent results, consecutive separators are not collapsed so
	  you may get an empty string as a valid result. Tested by: George
	  Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........
	  Merged revisions 423476 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-18 18:31 +0000 [r423462]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pubsub.c: Add subscription state test events. These
	  are needed for a set of batched notification RLS tests that are
	  about to be committed to the testsuite. Review:
	  https://reviewboard.asterisk.org/r/3967

2014-09-18 17:11 +0000 [r423425]  Jonathan Rose <jrose@digium.com>

	* res/res_pjsip_endpoint_identifier_ip.c, /:
	  res_pjsip_endpoint_identifier_ip: Fix parsing of match value with
	  CIDR Also fixes comma separates match lists ASTERISK-24290 #close
	  Reported by: Ray Crumrine Review:
	  https://reviewboard.asterisk.org/r/3995/ ........ Merged
	  revisions 423417 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-18 17:09 +0000 [r423418-423423]  Richard Mudgett <rmudgett@digium.com>

	* bridges/bridge_softmix.c: bridge_softmix.c: Made use
	  ao2_replace() instead of the inline equivalent. * Clarified some
	  read/write format comments. * Fixed a doxygen tag typo.

	* main/astobj2.c, contrib/scripts/refcounter.py, /:
	  astobj2.c/refcounter.py: Fix to deal with invalid object refs. *
	  Make astob2 REF_DEBUG output an invalid object line when an
	  invalid ao2 object ref/unref is attempted. This is similar to the
	  constructor/destructor lines. * Fixed refcounter.py to handle
	  skewed objects that have constructor/destructor states. * Made
	  refcounter.py highlight the invalid ao2 object refs by putting
	  them in their own section of the processed output file. * Made
	  refcounter.py highlight unreffing an object by more than one that
	  results in a negative ref count and the object being destroyed.
	  The abnormally destroyed object is reported in the invalid and
	  finalized object sections of the output. Review:
	  https://reviewboard.asterisk.org/r/3971/ ........ Merged
	  revisions 423349 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 423400 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423416 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-18 16:37 +0000 [r423348-423414]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/format_cap.h, main/channel.c, main/format_cap.c,
	  main/translate.c: Add API call to determine if format capability
	  structure is "empty". Empty here means that there are no formats
	  in the format_cap structure or the only format in it is the
	  "none" format. I've added calls to check the emptiness of a
	  format_cap in a few places in order to short-circuit operations
	  that would otherwise be pointless as well as to prevent some
	  assertions from being triggered in cases where channels with no
	  formats are used.

	* /, res/res_fax_spandsp.c: res_fax_spandsp: Properly handle
	  cleanup before starting FAXes. If faxing fails at a very early
	  stage, then it is possible for us to pass a NULL t30 state
	  pointer to spandsp, which spandsp is none too pleased with. This
	  patch ensures that we pass the correct pointer to spandsp in the
	  situation where we have not yet set our local t30 state pointer.
	  ASTERISK-24301 #close Reported by Matt Jordan Patches:
	  ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License
	  #5049) ........ Merged revisions 423360 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423365 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_mwi.c,
	  res/res_pjsip_dialog_info_body_generator.c,
	  res/res_pjsip_xpidf_body_generator.c,
	  res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c,
	  res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
	  res/res_pjsip_pidf_body_generator.c: res_pjsip_pubsub: Add some
	  type safety when generating NOTIFY bodies. res_pjsip_pubsub has
	  two separate checks that it makes when a SUBSCRIBE arrives. * It
	  checks that there is a subscription handler for the Event * It
	  checks that there are body generators for the types in the Accept
	  header The problem is, there's nothing that ensures that these
	  two things will actually mesh with each other. For instance,
	  Asterisk will accept a subscription to MWI that accepts pidf+xml
	  bodies. That doesn't make sense. With this commit, we add some
	  type information to the mix. Subscription handlers state they
	  generate data of type X, and body generators state that they
	  consume data of type X. This way, Asterisk doesn't end up in some
	  hilariously mismatched situation like the one in the previous
	  paragraph. ASTERISK-24136 #close Reported by Mark Michelson
	  Review: https://reviewboard.asterisk.org/r/3877 Review:
	  https://reviewboard.asterisk.org/r/3878 ........ Merged revisions
	  423344 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-18 15:13 +0000 [r423284]  George Joseph <george.joseph@fairview5.com>

	* /, res/res_pjsip/location.c,
	  res/res_pjsip_endpoint_identifier_ip.c,
	  res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c,
	  include/asterisk/res_pjsip.h, res/res_pjsip/config_auth.c:
	  res_pjsip: ami: Fix error in AMI output when an endpoint has no
	  transport When no transport is associated to an endpoint, the AMI
	  output for PJSIPShowEndpoint indicates an error instead of
	  silently ignoring the missing transport. This patch causes the
	  error to appear only if a transport was specified on the endpoint
	  and the transport doesn't exist. It also fixes an issue with
	  counting the objects that were actually found. ASTERISK-24161
	  #close ASTERISK-24331 #close Tested by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3998/ ........ Merged
	  revisions 423282 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-18 15:00 +0000 [r423281]  David M. Lee <dlee@digium.com>

	* makeopts.in, Makefile: Only install dahdi_span_config_hook if
	  DAHDI is enabled This patch changes the install to only install
	  the hook script if DAHDI is enabled. It also adds the script to
	  the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so
	  that it's not between the _MAKEOPTS variables and their comment.
	  This allows installs which specify a --prefix to work normally,
	  as long as they don't enable DAHDI. Review:
	  https://reviewboard.asterisk.org/r/3972/

2014-09-18 14:45 +0000 [r423279]  George Joseph <george.joseph@fairview5.com>

	* main/manager.c, /, include/asterisk/config.h, main/config.c:
	  config: bug: Fix SEGV in ast_category_insert when matching
	  category isn't found If you call ast_category_insert with a match
	  category that doesn't exist, the list traverse runs out of 'next'
	  categories and you get a SEGV. This patch adds check for the
	  end-of-list condition and changes the signature to return an int
	  for success/failure indication instead of a void. The only
	  consumer of this function is manager and it was also changed to
	  use the return value. Tested by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3993/ ........ Merged
	  revisions 423276 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 423277 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423278 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-17 18:05 +0000 [r423209-423255]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the
	  thread terminating pj stuff is registered. ........ Merged
	  revisions 423253 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423254 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix 100% CPU usage
	  due to timer heap thread spinning. Side note: I need a vacation.
	  ........ Merged revisions 423210 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423211 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix building when
	  pjproject is not used. ........ Merged revisions 423207 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423208 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-16 16:32 +0000 [r423192]  Scott Griepentrog <sgriepentrog@digium.com>

	* apps/app_voicemail.c, include/asterisk/file.h, main/file.c:
	  Voicemail: get correct duration when copying file to vm Changes
	  made during format improvements resulted in the recording to
	  voicemail option 'm' of the MixMonitor app writing a zero length
	  duration in the msgXXXX.txt file. This change introduces a new
	  function ast_ratestream(), which provides the sample rate of the
	  format associated with the stream, and updates the app_voicemail
	  function for ast_app_copy_recording_to_vm to calculate the right
	  duration. Review: https://reviewboard.asterisk.org/r/3996/
	  ASTERISK-24328 #close

2014-09-16 12:12 +0000 [r423152-423173]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_session.c, /: res_pjsip_session: Fix usage of wrong
	  memory pool when creating local SDP. ........ Merged revisions
	  423172 from http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c, /:
	  res_rtp_asterisk: Fix a myriad of TURN client issues. 1. The
	  number of file descriptors an ioqueue instance can handle is
	  fixed, so we now spawn the required number to handle the load. 2.
	  Our transport identifiers were exceeding the range supported by
	  pjnath. 3. The TURN client did not set up client binding causing
	  needless bandwidth usage. 4. The code no longer updates address
	  information on each packet. 5. STUN traffic was getting looped
	  back to Asterisk instead of going through the TURN server. 6.
	  Synchronization now ensures things are completely setup or
	  destroyed. 7. Logging now reflects the target the TURN server is
	  sending to/receiving from on our behalf. ASTERISK-23577 #close
	  Reported by: Jay Jideliov ASTERISK-23634 #close Reported by:
	  Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/
	  ........ Merged revisions 423150 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423151 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-15 10:49 +0000 [r423069-423129]  Walter Doekes <walter+asterisk@wjd.nu>

	* /,
	  contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py
	  (added): contrib: Fix verifyi typo in alembic DB script
	  ps_transport table. Reported by: Zogot (on IRC) Patches: tmp.diff
	  uploaded by Zogot, cleaned up by me. ........ Merged revisions
	  423128 from http://svn.asterisk.org/svn/asterisk/branches/12

	* configs/samples/sip.conf.sample, /: chan_sip: Clarify that
	  sipdebug=yes cannot be undone by the CLI. Document it in
	  sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod
	  Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged
	  revisions 423066 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 423067 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 423068 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-12 16:09 +0000 [r422985]  Jonathan Rose <jrose@digium.com>

	* main/config.c, /: Realtime: Fix a bug that caused realtime
	  destroy command to crash Also has could affect with anything that
	  goes through ast_destroy_realtime. If a CLI user used the command
	  'realtime destroy <family>' with only a single column/value pair,
	  Asterisk would crash when trying to create a variable list from a
	  NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson
	  Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged
	  revisions 422984 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-11 22:16 +0000 [r422965]  Mark Michelson <mmichelson@digium.com>

	* /, main/app.c: Remove undocumented default behavior of
	  ast_play_and_record_full acceptdtmf. ast_play_and_record_full()
	  has a parameter called "acceptdtmf" that is a string of
	  acceptable DTMF digits that may be pressed by a caller to end and
	  accept the recording. ARI uses this function in order to perform
	  recording, and it provides options for what is passed as
	  acceptdtmf to ast_play_and_record_full(). By default, ARI passes
	  an empty string, with the intention that no DTMF can be used to
	  end the recording. The problem is that ast_play_and_record_full()
	  attempts to be "helpful" by setting "#" as the acceptdtmf if an
	  empty string or NULL pointer has been passed in. With ARI, this
	  results in unexpected behavior occurring if you have attempted to
	  intercept "#" yourself in order to perform some other
	  manipulation of the live recording. This change removes the
	  "helpful" behavior by no longer accepting "#" as a default
	  acceptdtmf if none is specified by the caller of
	  ast_play_and_record_full(). This makes the ARI scenario work as
	  expected. The other callers of ast_play_and_record_full() are
	  app_voicemail and app_minivm, and in both cases, they pass an
	  explicit "#" to ast_play_and_record_full() as acceptdtmf, so they
	  are unaffected by this change. ........ Merged revisions 422964
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-10 16:04 +0000 [r422905]  George Joseph <george.joseph@fairview5.com>

	* /, main/config.c: config: bug: fix truncation of included config
	  files on permissions error ast_config_text_file_save() currently
	  truncates include files as they are processed. If a subsequent
	  include file or the main config file has a permissions error that
	  prevents writing, earlier include files are left truncated
	  resulting in a frantic search for backups. This patch causes
	  ast_config_text_file_save to check for write access on all files
	  before it truncates any of them. Will be applied 1.8 > trunk.
	  Tested by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3986/ ........ Merged
	  revisions 422900 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 422903 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422904 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-10 15:59 +0000 [r422901]  Sean Bright <sean@malleable.com>

	* res/res_pjsip/config_auth.c, /: pjsip/config_auth.c: Add missing
	  whitespace to log messages. The errors generated when validating
	  'auth' settings are missing a space which makes the messages a
	  little confusing. ........ Merged revisions 422899 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-09 20:01 +0000 [r422883]  Rusty Newton <rnewton@digium.com>

	* /, sounds/sounds.xml, sounds/Makefile: Sounds/BuildSystem:
	  Modifications to include new releases and Japanese language.
	  Modifying Makefile and sounds.xml to include new core 1.4.26 and
	  extra 1.4.15 sound prompt releases, plus the new Japanese core
	  sound prompts contributed by QLOOG. ASTERISK-23324 Reported by:
	  Kevin McCoy Tested by: Rusty Newton ........ Merged revisions
	  422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 422790 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422791 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-08 18:03 +0000 [r422851-422855]  Mark Michelson <mmichelson@digium.com>

	* configs/samples/pjsip.conf.sample: Add note about configuring
	  list_items on a single line.

	* configs/samples/pjsip.conf.sample: Add sample configuration for
	  resource lists. On review /r/3977, it was recommended to note in
	  the sample configuration about the size limitation for resource
	  lists. However, since there was no section in the sample
	  configuration at all for resource list subscriptions, I decided
	  to make a separate commit where I have added the necessary sample
	  configuration as well as the size limitation warning.

	* res/res_pjsip_pubsub.c: Pre-allocate transmission data buffer for
	  RLS NOTIFY requests. PJSIP, unless a constant is modified at
	  compilation time, limits SIP requests to 4000 bytes. Full-state
	  RLS notifications can easily exceed this limit with moderately
	  small lists. This changeset allows for Asterisk to work around
	  this size limit by performing its own allocation of the
	  transmission data buffer. This way, Asterisk can allocate a
	  buffer that exceeds the built-in maximum. We still impose our own
	  limit of 64000 bytes, mainly because making allocations larger
	  than that is a bit absurd. ASTERISK-24181 #close Reported by Mark
	  Michelson Review: https://reviewboard.asterisk.org/r/3977

2014-09-08 15:41 +0000 [r422836]  Jonathan Rose <jrose@digium.com>

	* res/res_pjsip_pubsub.c: res_pjsip_pubsub: Check supported headers
	  for eventlist when subscribing to resource list
	  https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan
	  According to the off-nominal plan, if evenlist support is not
	  specified in a SUBSCRIBE's supported header(s), that subscription
	  should be rejected with an error. ASTERISK-23871 Reported by:
	  Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/3960/diff/#index_header

2014-09-06 22:49 +0000 [r422767-422770]  Matthew Jordan <mjordan@digium.com>

	* /, main/cdr.c: main/cdr: Copy over location information during a
	  fork When a CDR is forked, a new CDR is created and appended to
	  the CDR chain for the Party A. The forked CDR starts life off as
	  a clone of the last non-finalized for the particular Party A. In
	  the past, merely copying over the snapshots for Party A/Party B
	  would be sufficient. However, as the CDRs now contain cached
	  information from Party A - specifically application/data,
	  context, and extension - we need to copy that over during a fork
	  as well. Huzzah for unit tests catching this when the
	  context/extension were derived from a cached value on the CDR
	  instead of on Party A. ........ Merged revisions 422769 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/rtp_engine.c, /: main/rtp_engine: Format NTP timestamps as
	  unsigned ints On some systems, a timeval's tv_sec/tv_usec will be
	  unsigned lont ints, as opposed to long ints. When the RTP engine
	  formats these as strings, it was previously formatting them as
	  signed integers, which can result in some odd negative timestamp
	  values (particularly on 32-bit systems). This patch formats the
	  values as unsigned long integers. ........ Merged revisions
	  422766 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-06 19:12 +0000 [r422747]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix retrieval of
	  "ice-pwd" attribute if in session and not media stream. ........
	  Merged revisions 422746 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-05 22:03 +0000 [r422716-422719]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /, apps/app_macro.c, include/asterisk/channel.h,
	  apps/app_stack.c: main/cdrs: Preserve context/extension when
	  executing a Macro or GoSub The context/extension in a CDR is
	  generally considered the destination of a call. When looking at a
	  2-party call CDR, users will typically be presented with the
	  following: context exten channel dest_channel app data default
	  1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial
	  actually takes place in a Macro, the current behaviour in 12 will
	  result in the following CDR: context exten channel dest_channel
	  app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The
	  same is true of a GoSub: context exten channel dest_channel app
	  data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This
	  generally makes the context/exten fields less than useful. It
	  isn't hard to preserve these values in the CDR state machine;
	  however, we need to have something that informs us when a channel
	  is executing a subroutine. Prior to this patch, there isn't
	  anything that does this. This patch solves this problem by adding
	  a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on
	  a channel when it executes a Macro or a GoSub. The CDR engine
	  looks for this value when updating a Party A snapshot; if the
	  flag is present, we don't override the context/exten on the main
	  CDR object. In a funny quirk, executing a hangup handler must
	  *not* abide by this logic, as the endbeforehexten logic assumes
	  that the user wants to see data that occurs in hangup logic,
	  which includes those subroutines. Since those execute outside of
	  a typical Dial operation (and will typically have their own
	  dedicated CDR anyway), this is unlikely to cause any heartburn.
	  Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254
	  #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis
	  ........ Merged revisions 422718 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: main/cdr: Fix crash/memory consumption in CDRs in
	  multi-party bridge scenarios This patch fixes an issue where CDRs
	  would get stuck generating an infinite number of CDRs, eventually
	  crashing Asterisk (and consuming a lot of memory along the way).
	  When a channel enters into a multi-party bridge, the CDR engine
	  creates mappings of each participant to each other participant,
	  picking the 'A' party as it goes. So, if we have four channels in
	  a multi-party bridge (Alice, Bob, Charlie, Denise), we would have
	  something like: Alice => Bob Alice => Charlie Alice => Denise Bob
	  => Charlie Bob => Denise Charlie => Denise This works fine when
	  participants enter the bridge a single time. When a participant
	  leaves a bridge, the CDRs for that channel are transitioned to a
	  finalized state. The bug occurs if Bob rejoins. When the CDR
	  engine creates mappings between the channels, it walks through
	  all the participants currently in the bridge, and realizes that
	  no one in the bridge can create a CDR with the channel (Bob). As
	  such it creates a new CDR for the candidate and appends it to
	  that candidate's chain. Unfortunately, on this particular code
	  path, it doesn't stop traversing the candidate's chain. Since we
	  just added ourselves to the chain, this causes the loop to keep
	  going, constantly adding new CDRs. This patch makes it so the
	  engine bails when it creates a CDR match in this case. Review:
	  https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close
	  Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat
	  ASTERISK-24208 Reported by: Frankie Chin ........ Merged
	  revisions 422715 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-05 20:35 +0000 [r422700]  Richard Mudgett <rmudgett@digium.com>

	* funcs/func_channel.c: func_channel.c: Add missing locking to some
	  CHANNEL() requests. * The CHANNEL() audionativeformat,
	  videonativeformat, audioreadformat, and audiowriteformat now need
	  locking since the media format rework when accessing the
	  channel's format pointers. * Increased the buffer size for
	  CHANNEL() audionativeformat and videonativeformat output strings
	  since the allow=all can be a lengthy list. * Tweaked the
	  CHANNEL() XML documentation for secure_bridge_signaling,
	  secure_bridge_media, and state. * Ensured the output buffer is
	  initialized for secure_bridge_signaling and secure_bridge_media.
	  * Made use the locked_copy_string() macro instead of inlining it
	  for trace and checkhangup.

2014-09-05 20:11 +0000 [r422665-422684]  Jonathan Rose <jrose@digium.com>

	* main/dial.c, include/asterisk/dial.h: Dial API: Add a dial option
	  to indicate the dialed channel will replace dialer Adds an option
	  to the dial API that marks an outgoing dial as replacing the
	  dialing channel for the purpose of propagating accountcode. When
	  it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of
	  AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on
	  the involved channels with ast_channel_req_accountcodes. Review:
	  https://reviewboard.asterisk.org/r/3968/

	* main/cli.c, /: Call IDs: Fix appearance of call ID in core show
	  channels when NULL NULL call IDs were meant to appear as '(none)'
	  but instead were showing the contents of an uninitialized
	  character buffer. ASTERISK-24223 Review:
	  https://reviewboard.asterisk.org/r/3979/ ........ Merged
	  revisions 422664 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-05 17:36 +0000 [r422661]  Richard Mudgett <rmudgett@digium.com>

	* main/devicestate.c, channels/chan_iax2.c: devicestate.c: Minor
	  tweaks * In ast_state_chan2dev() use ARRAY_LEN() instead of a
	  sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c.

2014-09-05 13:28 +0000 [r422646]  Kinsey Moore <kmoore@digium.com>

	* menuselect/menuselect.c: Menuselect: Fix incorrect enabling on
	  failed deps This corrects a situation where menuselect can
	  incorrectly enable a module by default that has defaultenabled
	  set to "no" and has failed/non-selected dependencies. The bug is
	  due to an inverted test when checking for whether the given
	  module should be set to enabled by default on load. Review:
	  https://reviewboard.asterisk.org/r/3975/ Reported by: John
	  Bigelow

2014-09-04 21:23 +0000 [r422631]  Jonathan Rose <jrose@digium.com>

	* main/manager.c, /: Manager: Require read permission for SYSTEM in
	  order to send FullyBooted Review:
	  https://reviewboard.asterisk.org/r/3969/ ........ Merged
	  revisions 422584 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 422625 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422626 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-03 14:05 +0000 [r422558]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_transport_websocket.c, /:
	  res_pjsip_transport_websocket: Fix crash when the Contact header
	  is not a URI. The code for changing the Contact header wrongly
	  assumed that the Contact would always contain a URI. This is
	  incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged
	  revisions 422557 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-02 20:29 +0000 [r422542]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_pjsip.c, res/res_pjsip_diversion.c,
	  res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h:
	  Resolve race condition where channels enter dialplan application
	  before media has been negotiated. Testsuite tests will
	  occasionally fail because on reception of a 200 OK SIP response,
	  an AST_CONTROL_ANSWER frame is queued prior to when media has
	  finished being negotiated. This is because session supplements
	  are called into before PJSIP's inv_session code has told us that
	  media has been updated. Sometimes the queued answer frame is
	  handled by the PBX thread before the ensuing media negotiations
	  occur, causing a test failure. As it turns out, there is another
	  place that session supplements could be called into, which is
	  after media has finished getting negotiated. What this commit
	  introduces is a means for session supplements to indicate when
	  they wish to be called into when handling an incoming SIP
	  response. By default, all session supplements will be run at the
	  same point that they were prior to this commit. However, session
	  supplements may indicate that they wish to be handled earlier
	  than normal on redirects, or they may indicate they wish to be
	  handled after media has been negotiated. In this changeset, two
	  session supplements have been updated to indicate a preference
	  for when they should be run: res_pjsip_diversion executes before
	  handling redirection in order to get information from the
	  Diversion header, and chan_pjsip now handles responses to INVITEs
	  after media negotiation to fix the race condition mentioned
	  previously. ASTERISK-24212 #close Reported by Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3930 ........ Merged revisions
	  422536 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-09-01 14:16 +0000 [r422504-422507]  Matthew Jordan <mjordan@digium.com>

	* main/cli.c, /: main/cli: Do not attempt to show CDR data for
	  internal channels Internal channels don't have CDRs. Querying the
	  CDR engine for their variables will make it cranky. ........
	  Merged revisions 422506 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis.c, /, res/stasis/stasis_bridge.c: res_stasis:
	  Don't play MoH to channels by default when added to holding
	  bridges When ARI manipulates a bridge, it generally doesn't care
	  what the mixing technology is. Operations on a bridge initiated
	  through ARI should perform their action in generally the same
	  way, regardless of the bridge's mixing technology. While the
	  mixing technology may determine how media flows to channels, the
	  actual operations on a bridge themselves should be the same.
	  Currently, this isn't the case with holding bridges. When a
	  channel joins without a role, MoH is started on that channel
	  automatically. Subsequent bridge operations that would stop MoH
	  would fail (as there is no Announcer channel playing MoH to the
	  bridge). Starting MoH on the bridge will also create two MoH
	  streams: one from the MoH being played on the participant
	  channel, and one from the announcer channel. From the perspective
	  of ARI users, this is counter-intuitive - I would not expect MoH
	  to be started for me. The mixing technology determines how media
	  is shared between participants, not the application experience.
	  This patch does the following: * The Stasis bridge class now
	  inspects channels as they are going into a bridge. If the bridge
	  has a holding capability, and the channel has no roles, we give
	  it a participant role and mark the default behaviour to have no
	  entertainment. This allows addChannel operations to continue to
	  set a participant role with an entertainment option if it felt
	  like it (or could do it). * The music on hold channel is now
	  Stasis approved (tm) Review:
	  https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close
	  Reported by: Samuel Galarneau Tested by: Samuel Galarneau
	  ........ Merged revisions 422503 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-30 17:32 +0000 [r422442-422445]  George Joseph <george.joseph@fairview5.com>

	* apps/app_confbridge.c, /: confbridge: Add Duration to
	  ConfbridgeList event The ConfbridgeList event doesn't include how
	  long the user has been a member of the conference. This patch
	  adds Duration (seconds) which is based on user->chan->answertime.
	  Tested by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3955/ ........ Merged
	  revisions 422444 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/manager.c, /: manager: Make WaitEvent action respect
	  eventfilters A WaitEvent issued via an http session isn't
	  respecting eventfilters defined for the user. I just added a
	  match_filter to the predicate that controls astman_append. Tested
	  by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3958/ ........ Merged
	  revisions 422439 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 422440 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422441 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-29 19:40 +0000 [r422374-422379]  Matthew Jordan <mjordan@digium.com>

	* doc/smsq.8 (added), /: doc: Add a manpage for the smsq utility
	  This patch adds a manpage for the smsq utility. Note that this is
	  one of the patches the Debian distro applies for the Asterisk
	  project, as per ASTERISK-24191. Review:
	  https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close
	  Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy
	  Laine (License 6561) ........ Merged revisions 422376 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 422377 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422378 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* doc/aelparse.8 (added), /: doc: Add a manpage for the aelparse
	  utility This patch adds a manpage for the aelparse utility. Note
	  that this is one of the patches the Debian distro applies for the
	  Asterisk project, as per ASTERISK-24191. Review:
	  https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close
	  Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy
	  Laine (License 6561) ........ Merged revisions 422371 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 422372 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422373 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-29 19:05 +0000 [r422359]  Scott Griepentrog <sgriepentrog@digium.com>

	* channels/chan_sip.c: The assertion that peer was not found on
	  final event message was being triggered on configuration reload.
	  This patch changes that case to just return instead. Review:
	  https://reviewboard.asterisk.org/r/3953/ Commited in trunk
	  revision 422358

2014-08-28 21:54 +0000 [r422296]  Matthew Jordan <mjordan@digium.com>

	* LICENSE, /: LICENSE: Clarify language in Asterisk's LICENSE to
	  allow for linking to UniMRCP The UniMRCP project distributes
	  Asterisk modules that integrate Asterisk with UniMRCP, and other
	  Asterisk users use the UniMRCP library as well. Unfortunately,
	  the UniMRCP license is Apache 2.0, which per the Free Software
	  Foundation, is not a compatible license with the GPLv2. "Please
	  note that this license is not compatible with GPL version 2,
	  because it has some requirements that are not in that GPL
	  version. These include certain patent termination and
	  indemnification provisions. The patent termination provision is a
	  good thing, which is why we recommend the Apache 2.0 license for
	  substantial programs over other lax permissive licenses." On the
	  other hand, UniMRCP is a great project and we'd like to let
	  people use it with Asterisk. This patch updates the LICENSE text
	  to allow users to link Asterisk with UniMRCP and distribute the
	  resulting binaries. ........ Merged revisions 422293 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 422294 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422295 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-28 20:30 +0000 [r422276]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_iax2.c: chan_iax2: Fix Dynamic IAX2
	  Registrations After Temporary DNS Failure The reporter on the
	  issue found some issues when upgrading from version 10 to 11 on
	  55 hosts. Two situations that can occur with dynamic
	  registrations. 1. With dnsmgr disabled, if the host is not
	  resolvable we are not trying to resolve the host again when it is
	  time to attempt to register again. This results in never
	  registering to the host. 2. With dnsmgr enabled, when the host is
	  temporarily not resolvable the address is set to 0.0.0.0:0 and
	  then when the host is resolvable the port is not being restored
	  and stays set to 0. This patch resolves these two issues by: *
	  Storing the hostname so that it can be used for resolving with
	  DNS. * Resolve the hostname on the next scheduled attempt to
	  register. * Storing the port used to reach the host so that when
	  the hostname is resolvable again, we can set the port again if
	  the port is still unset after looking up the host. ASTERISK-23767
	  #close Reported by: David Herselman Tested by: David Herselman,
	  Michael L. Young Patches:
	  asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/3856/ ........ Merged
	  revisions 422274 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422275 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-28 17:25 +0000 [r422256]  Richard Mudgett <rmudgett@digium.com>

	* /, UPGRADE.txt: Added ConfBridge AMI event note to UPGRADE.txt.
	  ........ Merged revisions 422255 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-28 15:49 +0000 [r422239]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pubsub.c: Fix bug that did not allow for multiple
	  batched RLS notifications to be sent. A misunderstanding of how
	  the scheduler worked caused further batched notifications beyond
	  the first not to get scheduled. Now we reset our scheduler ID to
	  -1 after the batched notification is sent. This way, further
	  notifications can be scheduled when they arise.

2014-08-28 00:36 +0000 [r422200-422215]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip/pjsip_options.c, /: res/res_pjsip/pjsip_options.c:
	  Eliminate excessive RAII_VAR usage. * Fix off nominal ref leak in
	  find_or_create_contact_status(). * Add missing NULL check of
	  status in update_contact_status() and init_start_time(). ........
	  Merged revisions 422214 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/sched.c, include/asterisk/sched.h: sched: Fix typo and
	  whitespace change.

2014-08-27 17:29 +0000 [r422177]  George Joseph <george.joseph@fairview5.com>

	* /, apps/confbridge/confbridge_manager.c, apps/app_confbridge.c:
	  confbridge: Add 'Admin' param to join, leave, mute, unmute and
	  talking events Currently there's no way to tell if a user is an
	  admin or not when receiving the join, leave, mute, unmute and
	  talking events. This patch adds that capability. Tested by:
	  George Joseph Review: https://reviewboard.asterisk.org/r/3950/
	  ........ Merged revisions 422176 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-27 15:31 +0000 [r422154]  Kinsey Moore <kmoore@digium.com>

	* include/asterisk/utils.h, /, channels/chan_sip.c,
	  tests/test_callerid.c (added), tests/test_utils.c,
	  main/callerid.c, main/utils.c, res/res_pjsip_caller_id.c:
	  CallerID: Fix parsing of malformed callerid This allows the
	  callerid parsing function to handle malformed input strings and
	  strings containing escaped and unescaped double quotes. This also
	  adds a unittest to cover many of the cases where the parsing
	  algorithm previously failed. Review:
	  https://reviewboard.asterisk.org/r/3923/ Review:
	  https://reviewboard.asterisk.org/r/3933/ ........ Merged
	  revisions 422112 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 422113 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 422114 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-26 23:28 +0000 [r422091]  George Joseph <george.joseph@fairview5.com>

	* apps/app_confbridge.c, /: confbridge: Make kick, mute and unmute
	  handle channel targets consistently. Kick, mute and unmute were a
	  little inconsistent in their handling of channel targets. This
	  patch cleans that up by insuring they all handle the 'all' target
	  consistently and adds the 'participants' target which acts on
	  non-admins. Documentation for kick was also cleaned up as it
	  never supported partial channel names. Tested by: George Joseph
	  Review: https://reviewboard.asterisk.org/r/3944/ ........ Merged
	  revisions 422090 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-26 22:13 +0000 [r422071]  Mark Michelson <mmichelson@digium.com>

	* main/sched.c, /: Fix race condition in the scheduler when
	  deleting a running entry. When scheduled tasks run, they are
	  removed from the heap (or hashtab). When a scheduled task is
	  deleted, if the task can't be found in the heap (or hashtab), an
	  assertion is triggered. If DO_CRASH is enabled, this assertion
	  causes a crash. The problem is, sometimes it just so happens that
	  someone attempts to delete a scheduled task at the time that it
	  is running, leading to a crash. This change corrects the issue by
	  tracking which task is currently running. If that task is
	  attempted to be deleted, then we mark the task, and then wait for
	  the task to complete. This way, we can be sure to coordinate task
	  deletion and memory freeing. ASTERISK-24212 Reported by Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/3927 ........
	  Merged revisions 422070 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-25 16:44 +0000 [r421979-422037]  Richard Mudgett <rmudgett@digium.com>

	* res/res_musiconhold.c: res_musiconhold.c: Release any format refs
	  before memset(). * Clear the channel music_state pointer before
	  destroying the music_state object for safety.

	* res/res_musiconhold.c, /: res_musiconhold: Fix MOH restarting
	  where it left off from the last hold. Restore code removed by
	  https://reviewboard.asterisk.org/r/3536/ that introduced a
	  regression that prevents MOH from restarting were it left off the
	  last time. ASTERISK-24019 #close Reported by: Jason Richards
	  Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch
	  uploaded by rmudgett Review:
	  https://reviewboard.asterisk.org/r/3928/ ........ Merged
	  revisions 421976 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421977 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421978 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-24 19:36 +0000 [r421911-421956]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_transport_websocket.c, /:
	  res_pjsip_transport_websocket: Attach the Websocket module on
	  outgoing INVITEs. In order to alter the Contact header on
	  in-dialog requests and responses the Websocket module must be
	  attached on outgoing INVITEs. The Contact header is modified so
	  that the PJSIP transport layer can find and use the existing
	  Websocket connection based on the source IP address, port, and
	  transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov
	  ........ Merged revisions 421955 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_transport_websocket.c:
	  res_pjsip_transport_websocket: Fix a progressive memory growth.
	  The packet structure used to receive messages was using the
	  transport pool. This meant that for each parsing the pool would
	  grow accordingly. Since memory can not be reclaimed without
	  resetting it this would cause the memory pool to grow and grow.
	  This change uses a specific memory pool for the packet structure
	  and resets it to a fresh state after the message has been
	  received and handled. ........ Merged revisions 421939 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_transport_websocket.c:
	  res_pjsip_transport_websocket: Ensure secure Websocket clients
	  can be called. This change enforces the transport in the Contact
	  header for Websocket clients. Previously a client may provide a
	  transport of 'ws' when it is actually using a transport of 'wss'.
	  This would cause outgoing calls to fail as the existing
	  connection could not be found. ........ Merged revisions 421931
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: chan_sip: Use the server reflexive ICE
	  candidate RTCP port as provided. This code originally worked
	  around an issue within res_rtp_asterisk itself. The wrong socket
	  was being used for the STUN check for RTCP, causing the port to
	  be the same as RTP. This was subsequently fixed and the RTCP port
	  provided for the ICE candidate is correct and does not need to be
	  incremented. ASTERISK-23997 #close Reported by: Badalian
	  Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav
	  (license 5249) ........ Merged revisions 421909 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421910 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-22 16:56 +0000 [r421882]  Mark Michelson <mmichelson@digium.com>

	* apps/app_mixmonitor.c: Fix a locking inversion in MixMonitor. We
	  need to unlock the audiohook before trying to lock the channel,
	  since the correct locking order is channel then audiohook.

2014-08-22 16:44 +0000 [r421880]  Jonathan Rose <jrose@digium.com>

	* res/res_stasis_answer.c, res/res_stasis.c, res/stasis/command.c,
	  res/res_stasis_playback.c, /, res/stasis/control.c,
	  res/stasis/stasis_bridge.c, res/stasis/command.h,
	  include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c:
	  ARI: Fix a crash caused by hanging during playback to a channel
	  in a bridge ASTERISK-24147 #close Reported by: Edvin Vidmar
	  Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged
	  revisions 421879 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-22 14:08 +0000 [r421860]  Matthew Jordan <mjordan@digium.com>

	* main/message.c, /: main/message: Add a new-line to a DEBUG
	  message ........ Merged revisions 421859 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-21 22:07 +0000 [r421802]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_musiconhold.c: res_musiconhold.c: Remove obsolete
	  REF_DEBUG code. Remove unneeded code that writes to the wrong
	  file location in an obsolete format. ........ Merged revisions
	  421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 421800 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421801 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-21 21:42 +0000 [r421790-421797]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_session.c, /: Switch from hostname to an IP address
	  in the SDP origin line. Using the hostname in the SDP origin line
	  may not satisfy the requirement of RFC 4566 that we use a FQDN or
	  IP address. This change has us use the same information from the
	  SDP connection line if possible. If not possible, we'll use the
	  configured media address. And if that's not possible, we use the
	  result of a PJLIB call to get the IP address of ourself.
	  ASTERISK-23994 #close Reported by Private Name Review:
	  https://reviewboard.asterisk.org/r/3925 ........ Merged revisions
	  421796 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/stasis/control.c: Ensure after-bridge behavior is correct
	  when moving from Stasis to a non-Stasis bridge. Because of the
	  departable state of channels that enter Stasis bridges, Stasis
	  has to take responsibility for directing the channel to its
	  intended after-bridge destination if the channel moves from a
	  Stasis bridge to a non-Stasis bridge. This change ensures that
	  when such a move occurs, when the channel leaves the bridging
	  system, any after bridge gotos are honored. Review:
	  https://reviewboard.asterisk.org/r/3920 ........ Merged revisions
	  421792 from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_caller_id.c, /: Let's try checking the name and
	  number, instead of the name twice. ........ Merged revisions
	  421789 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-21 21:25 +0000 [r421788]  Jonathan Rose <jrose@digium.com>

	* /, res/res_musiconhold.c: res_musiconhold: Fix reference leaks
	  caused when reloading with REF_DEBUG set Due to a faulty function
	  for debugging reference decrementing, it was possible to reduce
	  the refcount on the wrong object if two moh classes of the same
	  name were in the moh class container. (closes issue
	  ASTERISK-22252) Reported by: Walter Doekes Patches:
	  18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
	  6182) ........ Merged revisions 398937 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421777 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421779 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-21 21:18 +0000 [r421783]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip_caller_id.c: Improve consistency of party ID
	  privacy usage. Prior to this change, the Remote-Party-ID header
	  took the position of "If caller name and number are not
	  explicitly allowed, then they are private" and
	  P-Asserted-Identity took the position of "Caller name and number
	  are only private if marked explicitly so" Now both mechanisms of
	  conveying party identification use the former approach. ........
	  Merged revisions 421778 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-21 17:34 +0000 [r421675-421720]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: chan_sip: Don't use port derived from
	  fromdomain if it isn't set If a user does not provide a port in
	  the fromdomain setting, chan_sip will set the fromdomainport to
	  STANDARD_SIP_PORT (5060). The fromdomainport value will then get
	  used unilaterally in certain places. This causes issues with TLS,
	  where the default port is expected to be 5061. This patch
	  modifies chan_sip such that fromdomainport is only used if it is
	  not the standard SIP port; otherwise, the port from the SIP pvt's
	  recorded self IP address is used. Review:
	  https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close
	  Reported by: Elazar Broad patches: fromdomainport_fix.diff
	  uploaded by Elazar Broad (License 5835) ........ Merged revisions
	  421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 421718 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421719 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, UPGRADE.txt, main/app.c: ARI: Fix implicit answer when
	  playback is initiated on unanswered channel When issuing a POST
	  /channels/{channel_id}/play on a channel that is not yet
	  answered, ARI is supposed to: * Queue up an AST_CONTROL_PROGRESS
	  on the channel * Start up the playback of the media Instead, we
	  sneak an answer on the channel right before starting playing
	  media. This is due to ARI's usage of control_streamfile. This
	  function implicitly answers the channel (and doesn't give ARI the
	  option to stop it). The answering of the channel here is probably
	  unnecessary: * app_voicemail, by far the biggest consumer of this
	  function, always answers the channels anyway * control stream
	  file (in res_agi) and ControlPlayback probably shouldn't be
	  implicitly answering the channel. Answering should not be tied
	  directly to playing back media. As it turns out, the answering of
	  the channel here is pretty old: 356042 twilson if
	  (ast_channel_state(chan) != AST_STATE_UP) { 3087 anthm res =
	  ast_answer(chan); 180259 tilghman } (As in, ancient?) Note that
	  others ran into this problem and commented about it on various
	  mailing lists. Review: https://reviewboard.asterisk.org/r/3907/
	  ASTERISK-24229 #close Reported by: Matt Jordan ........ Merged
	  revisions 421695 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/stasis/messaging.h, main/dns.c, /, main/format_cache.c: Clean
	  up files that do not end with newlines Trivial patch to add new
	  lines to several files missing them. This fixes warnings when
	  compiling with gcc 4.1.2 on CentOS 5. ASTERISK-24245 #close
	  Reported by: Shaun Ruffell patches:
	  0002-Trivial-addition-of-newlines-at-end-of-three-files.patch
	  uploaded by Shaun Ruffell (License 5417) ........ Merged
	  revisions 421677 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/uri.h, main/uri.c: uri: Quiet warning about type
	  qualifiers ignored on function return type This patch fixes gcc
	  warnings that occur due to the type qualifier 'const' being
	  ignored on a return type of int. ASTERISK-24246 #close Reported
	  by: Shaun Ruffell patches:
	  0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch
	  uploaded by Shaun Ruffell (License 5417)

2014-08-20 22:49 +0000 [r421616-421645]  Richard Mudgett <rmudgett@digium.com>

	* main/bridge.c, res/res_pjsip_sdp_rtp.c, main/file.c,
	  main/bridge_channel.c, channels/chan_pjsip.c, main/channel.c:
	  chan_pjsip: Update media translation paths when new SDP
	  negotiated. On a SIP reinvite that changes media strams, the
	  PJSIP channel driver was flooding the log with "Asked to transmit
	  frame type %s, while native formats is %s" warnings. * Fixes
	  PJSIP not setting up translation paths when the formats change on
	  a reinvite. AFS-63 was effectively reintroduced because of the
	  media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the
	  unexpected frame format WARNING message to include more
	  information. * Added protective locking while altering formats on
	  a channel. Reworked set_format() to simplify and protect the
	  formats under manipulation. * Restored some code that got lost in
	  the media_formats work. (channel.c:set_format() and
	  res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark
	  Michelson Review: https://reviewboard.asterisk.org/r/3906/

	* /, main/cli.c: cli.c: Fix tab completion of "module load" when
	  MALLOC_DEBUG is enabled. filename_completion_function() returns
	  memory that was not allocated by the MALLOC_DEBUG allocation
	  tracker so the memory must be freed by ast_std_free(). ........
	  Merged revisions 421600 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421602 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421608 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-20 20:40 +0000 [r421566-421585]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pubsub.c: Set the role for inbound subscriptions
	  correctly. This was causing the AMI show_subscriptions test in
	  the testsuite to fail since all subscriptions were being seen as
	  subscribers instead of notifiers.

	* /, channels/chan_pjsip.c: Move evaluation of set_var options in
	  pjsip to the end of channel initialization. This allows for
	  set_var to override certain defaults such as caller ID and codec
	  values. This also fixes a test suite regression. The "set_var"
	  test suite test attempted to use set_var to override caller ID,
	  but a recent change caused that to no longer work. ........
	  Merged revisions 421565 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-20 13:04 +0000 [r421538]  Kinsey Moore <kmoore@digium.com>

	* include/asterisk/stasis_bridges.h, tests/test_cel.c,
	  res/ari/ari_model_validators.c, main/stasis_bridges.c,
	  res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /,
	  res/stasis/app.c, main/bridge.c: Stasis: Add information to blind
	  transfer event When a blind transfer occurs that is forced to
	  create a local channel pair to satisfy the transfer request,
	  information about the local channel pair is not published. This
	  adds a field to describe that channel to the blind transfer
	  message struct so that this information is conveyed properly to
	  consumers of the blind transfer message. This also fixes a bug in
	  which Stasis() was unable to properly identify the channel that
	  was replacing an existing Stasis-controlled channel due to a
	  blind transfer. Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3921/ ........ Merged
	  revisions 421537 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-19 20:28 +0000 [r421448-421488]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip.c: Alter documentation for callerid_privacy to
	  use correct values. ........ Merged revisions 421485 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis.c, /: Fix compilation error on certain versions of
	  GCC. ........ Merged revisions 421447 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-19 19:42 +0000 [r421445]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c, /: AMI Docs: Fix Status channel parameter
	  optionality ........ Merged revisions 421442 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421443 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421444 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-19 16:28 +0000 [r421423]  Jonathan Rose <jrose@digium.com>

	* res/res_stasis.c, /: ARI: Fix a bug where
	  /channels/{channelID}/continue doesn't execute PBX If
	  /channels/{channelID}/continue is called on a channel that was
	  originated without a PBX (such as the ARI command POST channel
	  with a stasis application argument), the channel will not start
	  dialplan execution. This patch will now run the PBX out of the
	  stasis execution if the channel doesn't currently have an active
	  PBX upon continuing. ASTERISK-24043 #close Reported by: Krandon
	  Bruse Review: https://reviewboard.asterisk.org/r/3917/ Patches:
	  stasis-continue.diff submitted by Krandon Bruse (license 6631)
	  ........ Merged revisions 421416 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-19 16:11 +0000 [r421403]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_pjsip_caller_id.c, channels/chan_pjsip.c,
	  res/res_pjsip_session.c: chan_pjsip: Fix attended transfer
	  connected line name update. A calls B B answers B SIP attended
	  transfers to C C answers, B and C can see each other's connected
	  line information B completes the transfer A has number but no
	  name connected line information about C while C has the full
	  information about A I examined the incoming and outgoing party id
	  information handling of chan_pjsip and found several issues: *
	  Fixed ast_sip_session_create_outgoing() not setting up the
	  configured endpoint id as the new channel's caller id. This is
	  why party A got default connected line information. * Made
	  update_initial_connected_line() use the channel's CALLERID(id)
	  information. The core, app_dial, or predial routine may have
	  filled in or changed the endpoint caller id information. * Fixed
	  chan_pjsip_new() not setting the full party id information
	  available on the caller id and ANI party id. This includes the
	  configured callerid_tag string and other party id fields. * Fixed
	  accessing channel party id information without the channel lock
	  held. * Fixed using the effective connected line id without doing
	  a deep copy outside of holding the channel lock. Shallow copy
	  string pointers can become stale if the channel lock is not held.
	  * Made queue_connected_line_update() also update the channel's
	  CALLERID(id) information. Moving the channel to another bridge
	  would need the information there for the new bridge peer. * Fixed
	  off nominal memory leak in update_incoming_connected_line(). *
	  Added pjsip.conf callerid_tag string to party id information from
	  enabled trust_inbound endpoint in caller_id_incoming_request().
	  AFS-98 #close Reported by: Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/3913/ ........ Merged
	  revisions 421400 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-18 21:10 +0000 [r421376]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Skinny: Fixup compile warning for non
	  dev-mode.

2014-08-18 20:19 +0000 [r421337]  George Joseph <george.joseph@fairview5.com>

	* funcs/func_config.c, /: func_config: Change 'Not Found' message
	  from ERROR to DEBUG When you call the CONFIG dialplan function
	  with the name of a variable that doesn't exist in the target
	  context you get an ERROR. This does nothing but clutter up the
	  logs with messages that may be perfectly acceptable. Just because
	  a variable wasn't in the context doesn't mean it's an error.
	  Maybei t's optional or just needs to be defaulted or ignored.
	  This patch changes the log level from ERROR to DEBUG. If a
	  dialplan developer wants to debug their dialplan they still canby
	  setting the console debug level as needed. Tested by: George
	  Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........
	  Merged revisions 421327 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421328 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421329 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-18 01:13 +0000 [r421230-421312]  Matthew Jordan <mjordan@digium.com>

	* res/ari/resource_channels.c: res/ari/resource_channels: Fix
	  compilation issue Forgot a parameter. Whoops.

	* res/ari/resource_channels.c: res/ari/resource_channels: Don't
	  return allocation failure on failed function If a function fails
	  to execute, it is most likely due to one of two reasons: (1) The
	  function doesn't exist or can't be read from (2) The function is
	  dangerous and is restricted based on the user's permissions
	  Currently we return allocation failure, which is incorrect. This
	  updates the reason code to more accurately reflect why the
	  request failed. ASTERISK-24215

	* /, apps/app_meetme.c: apps/app_meetme: Fix crash when publishing
	  MeetMe messages with no channel The same function,
	  meetme_stasis_generate_msg, handles creating and publishing
	  Stasis message both when there are channels in the MeetMe
	  conference and when there are no channels in the conference. When
	  the performance improvement was made to use cached snapshots,
	  this created a situation where Asterisk would crash: obtaining a
	  cached snapshot is not NULL tolerant. This patch restores the
	  previous implementation, which used a NULL safe set of routines
	  to produce a blob containing the channel snapshot (if available)
	  and information about the MeetMe conference. ASTERISK-24234
	  #close Reported by: Shaun Ruffell Tested by: Shaun Ruffell
	  ........ Merged revisions 421270 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_dial.c, /: apps/app_dial: Fix Dial 'z' option The 'z'
	  option is supposed to disable the dial timeout in the case of a
	  call forward. Unfortunately, the wrong timeout timer was passed
	  to the do_forward function, resulting in the option not working.
	  ASTERISK-24225 #close Reported by: dimitripietro Tested by:
	  dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by
	  rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by
	  rmudgett (License 5621) ........ Merged revisions 421232 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421233 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421234 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, configure, configure.ac: configure: Undefine FORTIFY_SOURCE
	  prior to defining it for patched gcc Some distributions of Linux
	  patch gcc to define FORTIFY_SOURCE when gcc is executed with
	  optimization. This "help" unfortunately results in re-definition
	  warnings when FORTIFY_SOURCE is later defined in Asterisk's build
	  system. This patch undefines FORTIFY_SOURCE prior to defining it
	  to prevent this warning. Review:
	  https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close
	  Reported by: Kilburn Tested by: Kilburn, wdoekes patches:
	  1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by
	  cloos (License 5956) 11.diff uploaded by cloos (License 5956)
	  12.diff uploaded by cloos (License 5956) 13.diff uploaded by
	  cloos (License 5956) ........ Merged revisions 421227 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421228 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421229 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-17 16:10 +0000 [r421210]  Joshua Colp <jcolp@digium.com>

	* res/res_http_websocket.c: res_http_websocket: Include query
	  parameters in client connection requests. Review:
	  https://reviewboard.asterisk.org/r/3914/

2014-08-15 17:08 +0000 [r421187]  Jonathan Rose <jrose@digium.com>

	* main/channel.c, /: Bridging: Fix a behavioral change when
	  checking if a channel is leaving a bridge r420934 introduced some
	  failures in the test suite. Upon investigating, it was discovered
	  that differences in the way we were evaluating whether a channel
	  was in the process of leaving a bridge were causing some
	  reinvites not to occur (mostly reinvites back to Asterisk when
	  ending a call). This patch fixes that behavioral change.
	  ASTERISK-24027 #close Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3910/ ........ Merged
	  revisions 421186 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-15 15:45 +0000 [r421042-421166]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c, /, main/app.c: app_voicemail/app: Remove
	  test events that were duplicated by r421059 Moving the test event
	  raised when a file is played back (which occurred in r421059)
	  broke the ever loving snot out of the voicemail tests. This
	  caused duplicate test events to get raised, as app_voicemail and
	  main/app were raising events prior to call ast_streamfile. The
	  voicemail tests did not enjoy getting multiple events. Since
	  raising the playback event in ast_streamfile is far more useful
	  to the vast majority of tests, this patch keeps the call there
	  and simply removes the extraneous calls that duplicated the
	  event. ........ Merged revisions 421125 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421164 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421165 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_hep_rtcp.c, /: res/res_hep_rtcp: Remove dependency on
	  PJSIP The res_hep_rtcp module was incorrectly including
	  <pjsip.h>. This didn't need to be included, as the module does
	  not using PJPROJECT any fashion. Unfortunately, because
	  res_hep_rtcp did not include pjsip in its MODULEINFO as a
	  dependency, this also meant that res_hep_rtcp will fail to
	  compile on a system without PJPROJECT. This patch removes the
	  include. Thanks to Damien Wedhorn for pointing this out in
	  #asterisk-dev. ASTERISK-24236 #close Reported by: Damien Wedhorn,
	  Matt Jordan Tested by: Damien Wedhorn ........ Merged revisions
	  421064 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/file.c, main/app.c: main/file: Move test event to emit
	  PLAYBACK event more consistently This is being done in advance of
	  the test for ASTERISK-23953 ........ Merged revisions 421059 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 421060 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 421061 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* tests/test_cel.c, main/cel.c, /: cel: Make sure channels in extra
	  fields include their unique IDs as well CEL typically tracks a
	  lot of information using the unique ID of the channel. This is
	  typically needed due to tying events together using the linked ID
	  of the various channels involved in a "call", which is derived
	  from the channel ID of the oldest channel involved in a bridge
	  (or in the case of a Dial, the parent channel). Previously, we
	  had updated the extra fields to include the involved channel
	  names, but forgot to put in the unique ID. This patch corrects
	  that error. ........ Merged revisions 421037 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-14 16:32 +0000 [r420957-421010]  Richard Mudgett <rmudgett@digium.com>

	* /, res/ari/resource_channels.c: ARI: Originate to app local
	  channel subscription code optimization. Reduce the scope of
	  local_peer and only get it if the ARI originate is subscribing to
	  the channels. Review: https://reviewboard.asterisk.org/r/3905/
	  ........ Merged revisions 421009 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/channel_internal_api.c, main/channel.c:
	  channel_internal_api.c: Replace some code with ao2_replace(). Use
	  ao2_replace() instead of ao2_cleanup(); ao2_bump(). ao2_replace()
	  has the advantange of not altering the ref count if the replaced
	  pointer is the same. Review:
	  https://reviewboard.asterisk.org/r/3904/

	* /, res/res_pjsip_send_to_voicemail.c:
	  res_pjsip_send_to_voicemail.c: Fix svn file properties. ........
	  Merged revisions 420956 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-13 16:53 +0000 [r420950]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip.c, /: PJSIP: Prevent crash no-URI contacts This
	  prevents a crash from occurring when a contact with no URI is
	  used for the creation of an outbound out-of-dialog request with
	  no associated endpoint. ........ Merged revisions 420949 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-13 16:07 +0000 [r420940]  Jonathan Rose <jrose@digium.com>

	* main/bridge_after.c, main/channel_internal_api.c,
	  include/asterisk/channel.h, apps/app_chanspy.c,
	  apps/app_mixmonitor.c, apps/app_stack.c, main/bridge_channel.c,
	  main/channel.c, main/pbx.c, /, main/framehook.c: Bridges: Fix
	  feature interruption/unintended kick caused by external actions
	  If a manager or CLI user attached a mixmonitor to a call running
	  a dynamic bridge feature while in a bridge, the feature would be
	  interrupted and the channel would be forcibly kicked out of the
	  bridge (usually ending the call during a simple 1 to 1 call).
	  This would also occur during any similar action that could set
	  the unbridge soft hangup flag, so the fix for this was to remove
	  unbridge from the soft hangup flags and make it a separate thing
	  all together. ASTERISK-24027 #close Reported by: mjordan Review:
	  https://reviewboard.asterisk.org/r/3900/ ........ Merged
	  revisions 420934 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-13 14:24 +0000 [r420919]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c: AMI: Improve documentation for Status action

2014-08-13 07:52 +0000 [r420899]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, main/logger.c: logger: Don't store verbose-magic in the log
	  files. In r399267, the verbose2magic stuff was edited. This time
	  it results in magic characters in the log files for multiline
	  messages. In trunk (and 13) this was fixed by the "stripping" of
	  those characters from multiline messages (in r414798). This fix
	  is altered to actually strip the characters and not replace them
	  with blanks. Review: https://reviewboard.asterisk.org/r/3901/
	  Review: https://reviewboard.asterisk.org/r/3902/ ........ Merged
	  revisions 420897 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 420898 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-12 23:43 +0000 [r420879-420881]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: chan_sip: Fix type mismatch when the format
	  is changed. Symptom is most likely an invalid ao2 object bad
	  magic number message or a less likely crash.

	* res/res_stasis_snoop.c: res_stasis_snoop.c: Fix off nominial exit
	  path leaving Snoop channel locked and not hungup. * Made use
	  ast_copy_string() instead of strcpy() for snoop uniqueid for
	  safety. There is no guarantee that the max channel uniqueid
	  length will remain the same as the snoop uniqueid space.

2014-08-12 11:17 +0000 [r420856]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: app_voicemail: Fix the
	  "test_voicemail_vm_info" unit test.

2014-08-11 20:53 +0000 [r420837]  Richard Mudgett <rmudgett@digium.com>

	* res/stasis/command.c, /: res/stasis/command.c: Fix recent commit
	  using spaces instead of tabs. ........ Merged revisions 420836
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-11 18:50 +0000 [r420808]  Matthew Jordan <mjordan@digium.com>

	* rest-api/api-docs/playbacks.json,
	  rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
	  rest-api/resources.json, include/asterisk/manager.h,
	  rest-api/api-docs/bridges.json,
	  rest-api/api-docs/recordings.json,
	  rest-api/api-docs/deviceStates.json,
	  rest-api/api-docs/endpoints.json,
	  rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
	  /, rest-api/api-docs/asterisk.json,
	  rest-api/api-docs/applications.json: AMI/ARI: Update version to
	  2.5.0/1.5.0 respectively This is to support the backwards
	  compatible changes made in the next version of Asterisk. ........
	  Merged revisions 420805 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-11 18:46 +0000 [r420796-420803]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_stasis.c: Stasis: Use the correct return value Return
	  the correct value instead of always returning 0 when setting
	  internal status on unreal channels. Reported by: Richard Mudgett
	  ........ Merged revisions 420802 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis.c, res/ari/resource_bridges.c, /,
	  res/stasis/stasis_bridge.c, include/asterisk/stasis_app.h:
	  Stasis: Allow internal channels directly into bridges The patch
	  to catch channels being shoehorned into Stasis() via external
	  mechanisms also happens to catch Announcer and Recorder channels
	  because they aren't known to be stasis-controlled channels in the
	  usual sense. This marks those channels as Stasis()-internal
	  channels and allows them directly into bridges. Review:
	  https://reviewboard.asterisk.org/r/3903/ ........ Merged
	  revisions 420795 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-11 18:32 +0000 [r420758-420794]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/stasis_app.h, main/stasis_channels.c,
	  res/ari/resource_channels.c, CHANGES, res/res_pjsip_pubsub.c,
	  main/manager_channels.c, apps/app_dial.c, res/stasis/app.c,
	  res/stasis/control.c: Improve call forwarding reporting,
	  especially with regards to ARI. This patch addresses a few
	  issues: 1) The order of Dial events have been changed when
	  performing a call forward. The order has now been altered to 1)
	  Dial begins dialing channel A. 2) When A forwards the call to B,
	  we issue the dial end event to channel A, indicating the dial is
	  being canceled due to a forward to B. 3) When the call to channel
	  B occurs, we then issue a new dial begin to channel B. 2) Call
	  forwards are now reported on the calling channel, not the peer
	  channel. 3) AMI DialEnd events have been altered to display the
	  extension the call is being forwarded to when relevant. 4) You
	  can now get the values of channel variables for channels that are
	  not currently in the Stasis application. This brings the
	  retrieval of channel variables more in line with the rest of
	  channel read operations since they may be performed on channels
	  not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan
	  ASTERISK-24138 #close Reported by Matt Jordan Patches:
	  forward-shenanigans.diff uploaded by Matt Jordan (License #6283)
	  Review: https://reviewboard.asterisk.org/r/3899

	* res/res_pjsip_pubsub.c: Fix crashing unit tests with regards to
	  RLS. The unit tests require a sorcery.conf file that has been set
	  up to store resource lists in memory rather than retrieving from
	  configuration. With a setup that is not conducive to running the
	  tests, a fault in sorcery currently causes Asterisk to crash when
	  attempting to run any of the tests. To get around the crash, this
	  adds a function that verifies the current environment and marks
	  the tests as "not run" if the setup is not correct.

	* res/res_pjsip_pubsub.c: Fix crash encountered by the testsuite.
	  Running testsuite tests locally produced no errors, but when run
	  using the continuous integration framework, crashes occurred. The
	  crashes occurred due to a refcounting error that had been fixed
	  for a similar situation.

2014-08-11 13:57 +0000 [r420742]  Matthew Jordan <mjordan@digium.com>

	* res/res_hep.c, res/res_hep_pjsip.c, res/res_hep_rtcp.c: res_hep:
	  Remove disabling of modules These modules were originally
	  specified as being disabled, as they were introduced midstream in
	  Asterisk 12. That makes it nicer for folks who are upgrading to a
	  new release in the middle of Asterisk 12. That's not the case for
	  Asterisk 13: it's a brand new release. There's no reason to have
	  the modules disabled by default in that case.

2014-08-11 10:40 +0000 [r420657-420717]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, main/utils.c: general: Fix memory Corruption in
	  __ast_string_field_ptr_build_va. If the space left in a
	  stringfield is between 0 and
	  (alignof(ast_string_field_allocation)-1) adding new data would
	  cause memory corruption, because we would assume enough space
	  (unsigned underrun). Thanks Arnd Schmitter for reporting and
	  finding out the cause! ASTERISK-23508 #close Reported by: Arnd
	  Schmitter Tested by: Arnd Schmitter, JoshE Review:
	  https://reviewboard.asterisk.org/r/3898/ ........ Merged
	  revisions 420680 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 420715 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 420716 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode.
	  ........ Merged revisions 420654 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 420655 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 420656 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-11 01:31 +0000 [r420607-420639]  Matthew Jordan <mjordan@digium.com>

	* funcs/func_jitterbuffer.c: funcs/func_jitterbuffer: Tweak
	  documentation This patch merely reformats and cleans up a bit of
	  the jitterbuffer documentation for the wiki.

	* UPGRADE.txt, configs/samples/extconfig.conf.sample, CHANGES,
	  apps/app_queue.c,
	  contrib/ast-db-manage/config/versions/d39508cb8d8_create_queue_rules.py
	  (added), configs/samples/queuerules.conf.sample: app_queue: Add
	  RealTime support for queue rules This patch gives the optional
	  ability to keep queue rules in RealTime. It is important to note
	  that with this patch: (a) Queue rules in RealTime are only
	  examined on module load/reload (b) Queue rules are loaded both
	  from the queuerules.conf file as well as the RealTime backend To
	  inform app_queue to examine RealTime for queue rules, a new
	  setting has been added to queuerules.conf's general section
	  "realtime_rules". RealTime queue rules will only be used when
	  this setting is set to "yes". The schema for the database table
	  supports a rule_name, time, min_penalty, and max_penalty columns.
	  min_penalty and max_penalty can be relative, if a '-' or '+'
	  literal is provided. Otherwise, the penalties are treated as
	  constants. For example: rule_name, time, min_penalty, max_penalty
	  'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2',
	  '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0',
	  '4564', '46546' 'test_rule', '40', '15', '50' which would result
	  in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY
	  to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20
	  seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
	  QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust
	  QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 -
	  After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
	  QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust
	  QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564
	  Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to
	  50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the
	  queue rules will be always reloaded on a module reload, even if
	  the underlying file did not change. With the option disabled, the
	  rules will only be reloaded if the file was modified. Review:
	  https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close
	  Reported by: Michael K patches: app_queue.c_realtime_trunk.patch
	  uploaded by Michael K (License 6621)

	* CHANGES: Update CHANGES file

	* UPGRADE.txt: Update UPGRADE.txt file

2014-08-08 20:08 +0000 [r420577-420592]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix build in devmode.

	* CHANGES, configs/samples/voicemail.conf.sample,
	  apps/app_voicemail.c: app_voicemail: Add the ability to specify
	  multiple email addresses. ASTERISK-24045 Reported by: Jacob
	  Barber Review: https://reviewboard.asterisk.org/r/3833/

2014-08-08 17:53 +0000 [r420534-420562]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c, channels/sip/security_events.c,
	  channels/sip/dialplan_functions.c, channels/sip/reqresp_parser.c,
	  channels/sip/route.c, channels/sip/utils.c,
	  channels/sip/config_parser.c: chan_sip: Mark chan_sip and its
	  files as extended support

	* rest-api-templates/make_ari_stubs.py: make_ari_stubs: Update wiki
	  prefix to '13'

	* rest-api-templates/res_ari_resource.c.mustache:
	  res_ari_resource.c.mustache: Update template to emit module
	  support level

	* main/message.c, /: main/message: remove debug message ........
	  Merged revisions 420533 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-08 03:03 +0000 [r420514]  Kinsey Moore <kmoore@digium.com>

	* tests/test_cel.c, /: CEL: Update unit tests for additional
	  information This updates the CEL unit tests for the new
	  information contained in the attended transfer CEL extra field.
	  ........ Merged revisions 420513 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-08 01:31 +0000 [r420494-420496]  Matthew Jordan <mjordan@digium.com>

	* UPGRADE.txt: Update UPGRADE file for 13 branch

	* /: Remove old properties

	* / (added): ___ _ _ _ __ _____ / _ \ | | (_) | | / ||____ | / /_\
	  \___| |_ ___ _ __ _ ___| | __ `| | / / | _ / __| __/ _ | '__| /
	  __| |/ / | | \ \ | | | \__ | || __| | | \__ | < _| |.___/ / \_|
	  |_|___/\__\___|_| |_|___|_|\_\ \___\____/

2014-08-07 21:58 +0000 [r420437]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and
	  resolve the large SDP poll issue. Replace sip_tls_read() and
	  sip_tcp_read() with a single function and resolve the poll/wait
	  issue with large SDP payloads. ASTERISK-18345 #close Reported by:
	  Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835)
	  patch uploaded by Elazar Broad Review:
	  https://reviewboard.asterisk.org/r/3882/ ........ Merged
	  revisions 420434 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 420435 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 420436 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-07 21:17 +0000 [r420389-420415]  Kinsey Moore <kmoore@digium.com>

	* main/stasis_bridges.c, /: Stasis: Correct blind transfer message
	  generation This fixes the json object creation format string and
	  key name for the BridgeBlindTransfer Stasis event allowing it to
	  be published properly. ........ Merged revisions 420414 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis_bridges.c, /: Stasis: Ensure transfer messages follow
	  validation rules This makes Stasis() event generation for
	  transfer messages follow validation rules. Currently,
	  ast_json_null() is being used in place of omitting a key entirely
	  which falls afoul of these validation rules.
	  https://reviewboard.asterisk.org/r/3892/ ........ Merged
	  revisions 420408 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_pubsub.c: Fix build in dev mode

2014-08-07 19:44 +0000 [r420384-420388]  Mark Michelson <mmichelson@digium.com>

	* /, main/bridge.c: Ensure bridges exist when trying to determine
	  bridged parties when publishing transfer information. ........
	  Merged revisions 420387 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/strings.c, include/asterisk/res_pjsip_presence_xml.h,
	  res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c,
	  res/res_pjsip_xpidf_body_generator.c, include/asterisk/strings.h,
	  res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c,
	  include/asterisk/res_pjsip_pubsub.h,
	  res/res_pjsip_pidf_body_generator.c: Add support for RFC 4662
	  resource list subscriptions. This commit adds the ability for a
	  user to configure a resource list in pjsip.conf. Subscribing to
	  this list simultaneously subscribes the subscriber to all
	  resources listed. This has the potential to reduce the amount of
	  SIP traffic when loads of subscribers on a system attempt to
	  subscribe to each others' states.

2014-08-07 18:51 +0000 [r420364]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/format_compatibility.h,
	  channels/iax2/format_compatibility.c,
	  channels/iax2/include/codec_pref.h, main/format_compatibility.c,
	  channels/chan_iax2.c, channels/iax2/codec_pref.c,
	  channels/iax2/include/format_compatibility.h: chan_iax2: Several
	  media format fixes. * Fixed the iax.conf bandwidth option. This
	  is the root cause of ASTERISK-24150. * Added checks in
	  iax2_request() to ensure that there are actual formats requested
	  for the new channel to prevent any more fracks from issues like
	  ASTERISK-24150. This is a consequence of the iax.conf bandwidth
	  option not working. * Fixed struct iax2_codec_pref.order member
	  size mismatch issue when converting to and from the codec
	  preference order list passed over the wire. In addition the
	  values sent over the wire are now compatible with previous
	  Asterisk versions. * Fixed several issues dealing with the struct
	  iax2_codec_pref members. Off-by-one, array limit errors, and the
	  order/framing members always need to be updated together. * Made
	  iax2_request() setup the channel's native format preference order
	  according to the user's wishes. The new media format strategy
	  needs the order specified earler. * Fixed usage of
	  ast_format_compatibility_bitfield2format(). The function can
	  return NULL if the bitfield was not associated with a function. *
	  Deleted dead code iax2_codec_pref_getsize() and
	  iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and
	  iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of
	  inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH,
	  IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants
	  again as they were in Asterisk v1.8. * Renamed prefs to
	  prefs_global so it won't get confused with the local pref
	  versions. * Fixed too small buffer in
	  handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in
	  handle_cli_iax2_show_peer() to output complete lines. * Changed
	  struct create_addr_info.prefs to be struct iax2_codec_pref as an
	  optimization so iax2_request() and iax2_call() do less work. *
	  Fixed a potential deadlock in ast_iax2_new() on an off-nominal
	  path when the pbx could not get started. * Made set_config()
	  setup a local prefs list along side the local capability format
	  bitfield. Once the config is loaded, then the local copies are
	  put into the global versions. * Fix unininialized codec_buf in
	  function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott
	  Griepentrog Review: https://reviewboard.asterisk.org/r/3890/

2014-08-07 15:30 +0000 [r420338]  Kinsey Moore <kmoore@digium.com>

	* include/asterisk/bridge_features.h, res/res_stasis.c,
	  res/stasis/command.c, rest-api/api-docs/events.json, /,
	  res/stasis/app.c, res/stasis/control.c, main/bridge.c,
	  main/bridge_basic.c, res/stasis/stasis_bridge.c,
	  include/asterisk/stasis_bridges.h, res/stasis/command.h,
	  include/asterisk/stasis_app.h, res/stasis/app.h,
	  res/stasis/control.h, apps/app_queue.c,
	  res/ari/ari_model_validators.c, main/cel.c,
	  main/stasis_bridges.c, res/ari/ari_model_validators.h,
	  main/channel.c, include/asterisk/datastore.h, tests/test_cel.c:
	  Stasis: Convey transfer information to applications This fixes a
	  class of issues where Stasis applications were not made aware
	  that their channels were being manipulated or replaced by
	  external entitiessuch as transfers, AMI commands, or dialplan
	  applications such as Bridge(). Inconsistent information such as
	  StasisEnd events with unknown channels as a result of masquerades
	  has also been corrected. To accomplish these fixes, several new
	  fields were added to blind and attended transfer messages as well
	  as StasisStart and BridgeAttendedTransfer Stasis events.
	  ASTERISK-23941 #close Review:
	  https://reviewboard.asterisk.org/r/3865/ Review:
	  https://reviewboard.asterisk.org/r/3857/ Review:
	  https://reviewboard.asterisk.org/r/3852/ Review:
	  https://reviewboard.asterisk.org/r/3816/ Review:
	  https://reviewboard.asterisk.org/r/3731/ Review:
	  https://reviewboard.asterisk.org/r/3729/ Review:
	  https://reviewboard.asterisk.org/r/3728/ ........ Merged
	  revisions 420325 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-07 14:37 +0000 [r420314-420315]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/res_pjsip_pubsub.h,
	  res/res_pjsip_pubsub.exports.in, res/res_pjsip_publish_asterisk.c
	  (added), res/res_pjsip_pubsub.c: res_pjsip_publish_asterisk: Add
	  support for exchanging device and mailbox state using SIP. This
	  module uses the inbound and outbound PUBLISH support to exchange
	  device and mailbox state between Asterisk instances. Each
	  instance is configured to publish to the other and requires no
	  intermediary server. The functionality provided is similar to the
	  XMPP and Corosync support. Review:
	  https://reviewboard.asterisk.org/r/3780/

	* include/asterisk/res_pjsip_outbound_publish.h (added),
	  res/res_pjsip_outbound_publish.exports.in (added),
	  res/res_pjsip_outbound_publish.c (added):
	  res_pjsip_outbound_publish: Add module which provides outbound
	  PUBLISH support. This module implements the core parts required
	  for doing outbound PUBLISH. It takes care of configuration,
	  lifetime management, and authentication. Additional modules
	  implement the specific events that are published. Review:
	  https://reviewboard.asterisk.org/r/3780/

2014-08-07 14:17 +0000 [r420289-420309]  Matthew Jordan <mjordan@digium.com>

	* main/pbx.c: pbx: Filter out pattern matching hints in responses
	  sent to ExtensionStateList Hints that are a pattern match are
	  technically stored in the hint container in the same fashion as
	  concrete implementations of hints. The pattern matching hints,
	  however, are not "real" in the sense that things can subscribe to
	  them: rather, they are stored in the hints container so that when
	  a subscription is made a "real" hint can be generated for the
	  subscription if one does not yet exist. The extension state core
	  takes care of this correctly by matching against non-pattern
	  matching extensions prior to pattern matching extensions. Because
	  of this, however, the ExtensionStateList AMI action was returning
	  pattern matching hints when executed. These hints are meaningless
	  from the perspective of AMI clients: their state will never
	  change, they cannot be subscribed to, and events would never
	  normally be generated from them. As such, we now filter these out
	  of the response.

	* build_tools/post_process_documentation.py: build_tools: Skip
	  managerEvent combining for AMI action responses AMI action
	  responses can (and will) reference AMI events that they return.
	  These event references and definitions should not be combined
	  with AMI events raised elsewhere in the code, as they are
	  specifically tied to the AMI action that raised them.
	  ASTERISK-24156 #close Reported by: Rusty Newton

2014-08-06 18:12 +0000 [r420212-420237]  Richard Mudgett <rmudgett@digium.com>

	* contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py,
	  /: Fix alembic script to work properly in offline mode. When run
	  in offline mode, this would attempt to check the database for the
	  presence of a type it was going to try to create. I now check the
	  context to see if we're running in offline mode and change a
	  parameter accordingly. ........ Merged revisions 407567 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py
	  (added), /: Add alembic script that adds contact user_agent and
	  endpoint message_context. ........ Merged revisions 411514 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* contrib/ast-db-manage/voicemail/versions/39428242f7f5_increase_recording_column_size.py
	  (added), /,
	  contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py,
	  contrib/ast-db-manage/config.ini.sample,
	  contrib/ast-db-manage/config/versions/1758e8bbf6b_increase_useragent_column_size.py
	  (added),
	  contrib/ast-db-manage/config/versions/5139253c0423_make_q_member_uniqueid_autoinc.py
	  (added), contrib/ast-db-manage/cdr.ini.sample,
	  contrib/ast-db-manage/voicemail.ini.sample: alembic: Adjust
	  sippeers, queue_members, and voicemail_messages tables. *
	  Increased the sippeers useragent max string size to 255. *
	  Changed the queue_members uniqueid to an auto incremented integer
	  instead of a string. * Increased the voicemail_messages BLOB size
	  to LONGBLOB on mysql. * Fixed the add_tables_for_pjsip config
	  change version downgrade actions to drop a table it created. *
	  Adjusted the sample alembic.ini files cdr.ini.sample,
	  config.ini.sample, and voicemail.ini.sample to give a mysql and
	  postgres sqlalchemy.url lines. ASTERISK-23847 #close Reported by:
	  Stephen More ASTERISK-23825 #close Reported by: Stephen More
	  ASTERISK-23909 #close Reported by: Stephen More Review:
	  https://reviewboard.asterisk.org/r/3870/ ........ Merged
	  revisions 420211 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-06 16:12 +0000 [r420149]  George Joseph <george.joseph@fairview5.com>

	* /, pbx/pbx_lua.c, main/pbx.c: pbx_lua: fix regression with global
	  sym export and context clash by pbx_config. ASTERISK-23818 (lua
	  contexts being overwritten by contexts of the same name in
	  pbx_config) surfaced because pbx_lua, having the
	  AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before
	  pbx_config. Since I couldn't find any reason for pbx_lua to
	  export it's symbols to the rest of Asterisk, I simply changed the
	  flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't
	  realize was that the symbols need to be exported not because
	  Asterisk needs them but because any external Lua modules like
	  luasql.mysql need the base Lua language APIs exported
	  (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's
	  an issue in pbx.c where context_merge was only merging includes,
	  switches and ignore patterns if the context was already existing
	  AND has extensions, or if the context was brand new. If pbx_lua
	  is loaded before pbx_config, the context will exist BUT pbx_lua,
	  being implemented as a switch, will never place extensions in it,
	  just the switch statement. The result is that when pbx_config
	  loads, it never merges the switch statement created by pbx_lua
	  into the final context. This patch sets pbx_lua's modflag back to
	  AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge
	  that catches the case where an existing context has includes,
	  switchs or ingore patterns but no actual extensions.
	  ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo
	  Teräs Tested by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3891/ ........ Merged
	  revisions 420146 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 420147 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 420148 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-06 15:32 +0000 [r420144]  Walter Doekes <walter+asterisk@wjd.nu>

	* funcs/func_channel.c: Add documentation to the ability to
	  retrieve the source port of a SIP call. (belongs with r419970)
	  ASTERISK-24040 #close Patches: func_channel.c.diff uploaded by
	  dtryba Review: https://reviewboard.asterisk.org/r/3781/

2014-08-06 12:55 +0000 [r420124]  Kinsey Moore <kmoore@digium.com>

	* configs/samples/stasis.conf.sample (added), main/named_acl.c,
	  apps/app_queue.c, main/stasis_bridges.c, main/loader.c,
	  main/stasis.c, apps/app_forkcdr.c, main/stasis_message.c,
	  funcs/func_cdr.c, res/res_corosync.c, res/res_stun_monitor.c,
	  res/res_stasis_test.c, res/res_stasis.c, apps/app_chanspy.c,
	  main/stasis_cache.c, main/pickup.c, main/security_events.c,
	  include/asterisk/stasis.h, main/devicestate.c, main/core_local.c,
	  res/res_stasis_snoop.c, main/endpoints.c, main/presencestate.c,
	  main/cdr.c, main/channel.c, main/stasis_system.c, main/manager.c,
	  main/test.c, main/file.c, main/app.c, pbx/pbx_realtime.c,
	  main/stasis_channels.c, tests/test_stasis.c,
	  res/parking/parking_manager.c, main/stasis_endpoints.c,
	  main/rtp_engine.c, main/ccss.c, main/bridge.c,
	  tests/test_stasis_channels.c: Stasis: Allow message types to be
	  blocked This introduces stasis.conf and a mechanism to prevent
	  certain message types from being published. Internally, this
	  works by preventing the chosen message types from being created
	  which ensures that those message types can never be published.
	  This patch also adjusts message publishers such that message
	  payloads are not created if the related message type is not
	  available. ASTERISK-23943 #close Review:
	  https://reviewboard.asterisk.org/r/3823/

2014-08-05 21:48 +0000 [r420098-420100]  Matthew Jordan <mjordan@digium.com>

	* res/stasis/messaging.c, /: stasis: Fix compilation issue with ao2
	  tagged objects ........ Merged revisions 420099 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/ari/resource_endpoints.c, rest-api/api-docs/events.json, /,
	  channels/chan_sip.c, res/stasis/app.c, res/stasis/messaging.h
	  (added), res/ari/resource_endpoints.h, res/res_pjsip_messaging.c,
	  tests/test_message.c (added), res/res_xmpp.c,
	  include/asterisk/json.h, CHANGES, include/asterisk/manager.h,
	  res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
	  main/json.c, res/res_ari_endpoints.c, include/asterisk/message.h,
	  res/ari/resource_channels.c, main/message.c, res/res_stasis.c,
	  res/stasis/messaging.c (added), rest-api/api-docs/endpoints.json:
	  Multiple revisions 420089-420090,420097 ........ r420089 |
	  mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
	  ARI: Add channel technology agnostic out of call text messaging
	  This patch adds the ability to send and receive text messages
	  from various technology stacks in Asterisk through ARI. This
	  includes chan_sip (sip), res_pjsip_messaging (pjsip), and
	  res_xmpp (xmpp). Messages are sent using the endpoints resource,
	  and can be sent directly through that resource, or to a
	  particular endpoint. For example, the following would send the
	  message "Hello there" to PJSIP endpoint alice with a display URI
	  of sip:asterisk@mycooldomain.org:
	  ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
	  This is equivalent to the following as well:
	  ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
	  Both forms are available for message technologies that allow for
	  arbitrary destinations, such as chan_sip. Inbound messages can
	  now be received over ARI as well. An ARI application that
	  subscribes to endpoints will receive messages from those
	  endpoints: { "type": "TextMessageReceived", "timestamp":
	  "2014-07-12T22:53:13.494-0500", "endpoint": { "technology":
	  "PJSIP", "resource": "alice", "state": "online", "channel_ids":
	  [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>",
	  "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.",
	  "variables": [] }, "application": "testsuite" } The above was
	  made possible due to some rather major changes in the message
	  core. This includes (but is not limited to): - Users of the
	  message API can now register message handlers. A handler has two
	  callbacks: one to determine if the handler has a destination for
	  the message, and another to handle it. - All dialplan
	  functionality of handling a message was moved into a message
	  handler provided by the message API. - Messages can now have the
	  technology/endpoint associated with them. Various other
	  properties are also now more easily accessible. - A number of ao2
	  containers that weren't really needed were replaced with vectors.
	  Iteration over ao2_containers is expensive and pointless when the
	  lifetime of things is well defined and the number of things is
	  very small. res_stasis now has a new file that makes up its
	  structure, messaging. The messaging functionality implements a
	  message handler, and passes received messages that match an
	  interested endpoint over to the app for processing. Note that
	  inadvertently while testing this, I reproduced ASTERISK-23969.
	  res_pjsip_messaging was incorrectly parsing out the 'to' field,
	  such that arbitrary SIP URIs mangled the endpoint lookup. This
	  patch includes the fix for that as well. Review:
	  https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close
	  Reported by: Matt Jordan ASTERISK-23969 #close Reported by:
	  Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37
	  -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties
	  :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue,
	  05 Aug 2014) | 2 lines test_message: Fix strict-aliasing
	  compilation issue ........ Merged revisions 420089-420090,420097
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-05 13:59 +0000 [r420028]  Jonathan Rose <jrose@digium.com>

	* main/format.c: chan_iax2: Fix a crash that occurs when using
	  allow=all for an IAX2 peer Or any combination of codecs that
	  includes Opus. ASTERISK-24107 #close Review:
	  https://reviewboard.asterisk.org/r/3885/

2014-08-04 21:00 +0000 [r420007]  Richard Mudgett <rmudgett@digium.com>

	* main/format_cache.c, include/asterisk/format_cache.h: Remove
	  duplicate definitions of ast_format_vp8.

2014-08-04 20:25 +0000 [r419970]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/dialplan_functions.c: Add the ability to retrieve
	  the source port of a SIP call. This adds the ability to call
	  CHANNEL(recvport) on chan_sip channels to see the port on which
	  an INVITE was received. ASTERISK-24040 #close Reported by dtryba
	  Patches: dialplan_functions.patch uploaded by dtryba (License
	  #6628) Review: https://reviewboard.asterisk.org/r/3781

2014-08-04 19:47 +0000 [r419945]  Rusty Newton <rnewton@digium.com>

	* main/manager.c, /: Manager - Improve documentation for manager
	  commands Getvar and Setvar. The documentation for these commands
	  did not make it clear that they could accept expressions and
	  functions. Modified to make this clear, but tried not to be
	  overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton
	  Tested by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/3854 ........ Merged revisions
	  419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 419943 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 419944 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-08-02 03:37 +0000 [r419914]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip.c: Manager: Add PJSIPShowEndpoint[s] documentation
	  This adds a large swath of response documentation for
	  PJSIPShowEndpoint and PJSIPShowEndpoints AMI commands. It relies
	  heavily on the existing text in the configInfo documentation via
	  xi:include tags to avoid documentation duplication. Review:
	  https://reviewboard.asterisk.org/r/3888/

2014-08-01 14:48 +0000 [r419888]  Mark Michelson <mmichelson@digium.com>

	* CHANGES, res/res_pjsip/pjsip_options.c: Add ContactStatusDetail
	  to PJSIPShowEndpoint AMI output. Now when running
	  PJSIPShowEndpoint, you will receive a ContactStatusDetail for
	  each bound contact that Asterisk is qualifying. This information
	  includes the URI of the contact, current reachability, and
	  roundtrip time. AFS-91 #close Reported by Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/3797

2014-07-31 16:19 +0000 [r419851]  Jonathan Rose <jrose@digium.com>

	* CHANGES, res/res_pjsip_notify.c: PJSIP: Send Notify AMI and CLI
	  commands can now send to URI instead of endpoint Review:
	  https://reviewboard.asterisk.org/r/3817/

2014-07-31 11:57 +0000 [r419822-419825]  Matthew Jordan <mjordan@digium.com>

	* main/rtp_engine.c, /, res/res_hep_rtcp.c (added), CHANGES,
	  channels/chan_pjsip.c, res/res_rtp_asterisk.c: res_hep_rtcp: Add
	  module that sends RTCP information to a Homer Server This patch
	  adds a new module to Asterisk, res_hep_rtcp. The module
	  subscribes to the RTCP topics in Stasis and receives RTCP
	  information back from the message bus. It encodes into HEPv3
	  packets and sends the information to the res_hep module for
	  transmission. Using this, someone with a Homer server can get
	  live call quality monitoring for all RTP-based channels in their
	  Asterisk 12+ systems. In addition, there were a few bugs in the
	  RTP engine, res_rtp_asterisk, and chan_pjsip that were uncovered
	  by the tests written for the Asterisk Test Suite. This patch
	  fixes the following: 1) chan_pjsip failed to set its channel
	  unique ids on its RTP instance on outbound calls. It now does
	  this in the appropriate location, in the serialized call
	  callback. 2) The rtp_engine was overflowing some values when
	  packed into JSON. Specifically, some longs and unsigned ints
	  can't be be packed into integer values, for obvious reasons.
	  Since libjansson only supports integers, floats, strings,
	  booleans, and objects, we print these values into strings. 3)
	  res_rtp_asterisk had a few problems: (a) it would emit a source
	  IP address of 0.0.0.0 if bound to that IP address. We now use
	  ast_find_ourip to get a better IP address, and properly marshal
	  the result into an ast_strdupa'd string. (b) Reports can be
	  generated with no report bodies. In particular, this occurs when
	  a sender is transmitting information to a receiver (who will send
	  no RTP back to the sender). As such, the sender has no report
	  body for what it received. We now properly handle this case, and
	  the sender will emit SR reports with no body. Likewise, if we
	  receive an RTCP packet with no report body, we will still
	  generate the appropriate events. ASTERISK-24119 #close ........
	  Merged revisions 419823 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* funcs/func_jitterbuffer.c, doc/appdocsxml.dtd, main/xmldoc.c:
	  xmldocs: Add support for an <example> tag in the Asterisk XML
	  Documentation This patch adds support for an <example /> tag in
	  the XML documentation schema. For CLI help, this doesn't change
	  the formatting too much: - Preceeding white space is removed -
	  Unlike with para elements, new lines are preserved However,
	  having an <example /> tag in the XML schema allows for the wiki
	  documentation generation script to surround the documentation
	  with {code} or {noformat} tags, generating much better content
	  for the wiki - and allowing us to put dialplan examples (and
	  other code snippets, if desired) into the documentation for an
	  application/function/AMI command/etc. Review:
	  https://reviewboard.asterisk.org/r/3807/

2014-07-30 18:32 +0000 [r419806]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c, res/res_manager_presencestate.c,
	  res/res_manager_devicestate.c, main/pbx.c: manager: Add state
	  list commands This patch adds three new AMI commands: *
	  ExtensionStateList (pbx.c) - list all known extension state hints
	  and their current statuses. Events emitted by the list action are
	  equivalent to the ExtensionStatus events. * PresenceStateList
	  (res_manager_presencestate) - list all known presence state
	  values. Events emitted are generated by the stasis message type,
	  and hence are PresenceStateChange events. * DeviceStateList
	  (res_manager_devicestate) - list all known device state values.
	  Events emitted are generated by the stasis message type, and
	  hence are DeviceStateChange events. Patch-by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3799/

2014-07-29 19:41 +0000 [r419789]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Do not omit the first header of a UserEvent AMI
	  action from the corresponding emitted UserEvent. ASTERISK-24124
	  #close Reported by Matt Jordan AFS-131 #close Reported by Matt
	  Jordan Patches: userevent.patch uploaded by Matt Jordan (License
	  #6283)

2014-07-29 10:56 +0000 [r419751-419766]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_session.c, /: res_pjsip_session: Fix race condition
	  where redirecting information may not be set. Since the PJSIP
	  INVITE session module is invoked before any session supplements
	  it was possible for it to handle a redirect before the
	  res_pjsip_diversion module interpreted and set redirecting
	  information on the channel. This would cause the redirecting
	  information to get lost. This patch ensures that session
	  supplements are *always* invoked before a redirect occurs by
	  explicitly calling them in the redirect handler. Review:
	  https://reviewboard.asterisk.org/r/3850/ ........ Merged
	  revisions 419764 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_xpidf_body_generator.c,
	  res/res_pjsip_pidf_body_generator.c:
	  res_pjsip_pidf_body_generator / res_pjsip_xpidf_body_generator:
	  Ensure local entity is unquoted. The local entity as provided by
	  PJSIP is quoted within '<' and '>'. As a result placing this
	  value into XML will result in malformed XML being produced. This
	  patch now unquotes the local entity so it can go safely into the
	  XML. Review: https://reviewboard.asterisk.org/r/3851/ ........
	  Merged revisions 419750 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-28 18:58 +0000 [r419688]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_speech_utils.c, main/channel.c, /,
	  funcs/func_frame_trace.c, main/abstract_jb.c: datastores: Audit
	  ast_channel_datastore_remove usage. Audit of v1.8 usage of
	  ast_channel_datastore_remove() for datastore memory leaks. *
	  Fixed leaks in app_speech_utils and func_frame_trace. * Fixed
	  app_speech_utils not locking the channel when accessing the
	  channel datastore list. Review:
	  https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of
	  ast_channel_datastore_remove() for datastore memory leaks. *
	  Fixed leak in func_jitterbuffer. (Was not in v12) Review:
	  https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of
	  ast_channel_datastore_remove() for datastore memory leaks. *
	  Fixed leaks in abstract_jb. * Fixed leak in
	  ast_channel_unsuppress(). Used by ARI mute control and
	  res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used
	  by ARI mute control and res_mutestream. Review:
	  https://reviewboard.asterisk.org/r/3861/ ........ Merged
	  revisions 419684 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 419685 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 419686 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-25 18:09 +0000 [r419612]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: loader: Fix an infinite loop when printing modules
	  using "module show". When creating the alphabetical sorted list
	  each module is added to a list temporarily. On the second
	  iteration each module already has a pointer to another module,
	  causing stuff to go into a loop. ASTERISK-24123 #close Reported
	  by: Malcolm Davenport

2014-07-25 16:47 +0000 [r419592]  Mark Michelson <mmichelson@digium.com>

	* res/res_ari_sounds.c, res/res_stasis.c, res/res_fax_spandsp.c,
	  res/res_timing_kqueue.c, res/res_odbc.c,
	  res/res_pjsip_transport_websocket.c, apps/app_voicemail.c,
	  res/res_calendar.c, channels/chan_unistim.c, cel/cel_radius.c,
	  channels/chan_multicast_rtp.c, res/res_pjsip_notify.c,
	  res/res_snmp.c, formats/format_sln.c, apps/app_meetme.c,
	  apps/app_dictate.c, codecs/codec_gsm.c, res/res_stasis_snoop.c,
	  res/res_musiconhold.c, res/res_format_attr_h264.c,
	  res/res_http_websocket.c, apps/app_followme.c,
	  res/res_config_sqlite.c, formats/format_siren7.c, cdr/cdr_csv.c,
	  formats/format_ilbc.c, channels/chan_phone.c,
	  apps/app_setcallerid.c, apps/app_osplookup.c, cel/cel_custom.c,
	  apps/app_mp3.c, res/res_agi.c, channels/chan_motif.c,
	  res/res_timing_timerfd.c, apps/app_confbridge.c,
	  res/res_format_attr_silk.c, formats/format_siren14.c,
	  res/res_sorcery_realtime.c, channels/chan_mgcp.c,
	  apps/app_jack.c, codecs/codec_lpc10.c,
	  res/res_pjsip_pidf_body_generator.c, res/res_config_pgsql.c,
	  funcs/func_dialplan.c, apps/app_nbscat.c, cdr/cdr_syslog.c,
	  res/res_pjsip_authenticator_digest.c, apps/app_festival.c,
	  res/res_fax.c, apps/app_waitforsilence.c, res/res_adsi.c,
	  res/res_crypto.c, res/res_ari_applications.c,
	  res/res_hep_pjsip.c, pbx/pbx_lua.c, res/res_pjsip_messaging.c,
	  res/res_pjsip_caller_id.c, channels/chan_console.c,
	  apps/app_getcpeid.c, res/res_stasis_answer.c,
	  channels/chan_oss.c, res/res_pjsip_nat.c,
	  res/res_pjsip_session.c, cdr/cdr_tds.c,
	  res/res_pjsip_header_funcs.c, res/res_parking.c,
	  formats/format_vox.c, res/res_pjsip_rfc3326.c,
	  res/res_ari_endpoints.c, res/res_stun_monitor.c,
	  res/res_pjsip_mwi.c, res/res_stasis_recording.c,
	  res/res_pjsip_xpidf_body_generator.c, apps/app_sms.c,
	  codecs/codec_ulaw.c, channels/chan_nbs.c, apps/app_stack.c,
	  channels/chan_pjsip.c, formats/format_g729.c, cel/cel_pgsql.c,
	  res/res_sorcery_config.c, res/res_ari.c, addons/chan_ooh323.c,
	  cdr/cdr_sqlite3_custom.c, codecs/codec_adpcm.c,
	  res/res_ari_asterisk.c, res/res_calendar_caldav.c,
	  apps/app_image.c, apps/app_ices.c, formats/format_wav_gsm.c,
	  main/cli.c, res/res_format_attr_celt.c, res/res_rtp_multicast.c,
	  channels/chan_dahdi.c, funcs/func_pitchshift.c, res/res_smdi.c,
	  res/res_pjsip_one_touch_record_info.c, pbx/pbx_ael.c,
	  pbx/pbx_realtime.c, apps/app_amd.c, channels/chan_alsa.c,
	  formats/format_h263.c, apps/app_url.c, res/res_pjsip_acl.c,
	  apps/app_externalivr.c, res/res_curl.c, formats/format_gsm.c,
	  res/res_speech.c, cdr/cdr_manager.c, res/res_calendar_exchange.c,
	  codecs/codec_g722.c, res/res_pjsip_multihomed.c,
	  res/res_ari_mailboxes.c, cel/cel_tds.c, res/res_sorcery_memory.c,
	  apps/app_fax.c, codecs/codec_speex.c, res/res_pjsip_sdp_rtp.c,
	  codecs/codec_g726.c, formats/format_ogg_vorbis.c,
	  apps/app_talkdetect.c, res/res_ari_channels.c,
	  res/res_pjsip_exten_state.c, apps/app_speech_utils.c,
	  apps/app_agent_pool.c, apps/app_waitforring.c, res/res_statsd.c,
	  addons/cdr_mysql.c, formats/format_g726.c, res/res_ari_bridges.c,
	  addons/app_mysql.c, res/res_stasis_playback.c,
	  addons/format_mp3.c, res/res_pjsip_endpoint_identifier_ip.c,
	  res/res_phoneprov.c, res/res_pjsip_t38.c,
	  res/res_pjsip_registrar_expire.c, cdr/cdr_pgsql.c,
	  cdr/cdr_radius.c, res/res_chan_stats.c,
	  res/res_format_attr_opus.c, res/res_config_odbc.c,
	  funcs/func_audiohookinherit.c,
	  res/res_pjsip_outbound_registration.c, cel/cel_manager.c,
	  funcs/func_odbc.c, res/res_pjsip_endpoint_identifier_anonymous.c,
	  funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c,
	  apps/app_minivm.c, res/res_pjsip_log_forwarder.c,
	  formats/format_h264.c, res/res_config_ldap.c, apps/app_ivrdemo.c,
	  addons/chan_mobile.c, apps/app_stasis.c,
	  res/res_pjsip_diversion.c, cdr/cdr_custom.c, apps/app_adsiprog.c,
	  res/res_pjsip_dtmf_info.c, res/res_rtp_asterisk.c,
	  res/res_calendar_icalendar.c, res/res_hep.c, channels/chan_sip.c,
	  channels/chan_bridge_media.c, codecs/codec_alaw.c,
	  apps/app_queue.c, res/res_srtp.c, funcs/func_presencestate.c,
	  res/res_timing_pthread.c, res/res_manager_presencestate.c,
	  res/res_corosync.c, apps/app_celgenuserevent.c,
	  cel/cel_sqlite3_custom.c, res/snmp/agent.c, pbx/pbx_dundi.c,
	  formats/format_g723.c, funcs/func_devstate.c,
	  res/res_pjsip_registrar.c,
	  res/res_pjsip_pidf_eyebeam_body_supplement.c,
	  addons/res_config_mysql.c,
	  res/res_pjsip_pidf_digium_body_supplement.c, apps/app_test.c,
	  res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c,
	  apps/app_alarmreceiver.c, apps/app_chanisavail.c,
	  res/res_format_attr_h263.c, res/res_pjsip_mwi_body_generator.c,
	  res/res_xmpp.c, res/res_http_post.c, channels/chan_iax2.c,
	  res/res_pjsip_endpoint_identifier_user.c, res/res_pjsip.c,
	  res/res_pktccops.c, res/res_pjsip_send_to_voicemail.c,
	  main/loader.c, cel/cel_odbc.c, res/res_ari_model.c,
	  channels/chan_skinny.c,
	  res/res_pjsip_outbound_authenticator_digest.c,
	  res/res_mwi_external.c, apps/app_skel.c, formats/format_pcm.c,
	  include/asterisk/module.h, res/res_pjsip_path.c,
	  res/res_ari_playbacks.c, res/res_pjsip_pubsub.c, cdr/cdr_odbc.c,
	  funcs/func_periodic_hook.c, res/res_stasis_test.c,
	  formats/format_jpeg.c, res/res_pjsip_refer.c,
	  formats/format_g719.c, res/res_clialiases.c,
	  res/res_config_sqlite3.c, res/res_ari_device_states.c,
	  formats/format_wav.c, apps/app_saycounted.c, apps/app_dahdiras.c,
	  apps/app_morsecode.c, res/res_stasis_mailbox.c,
	  res/res_ael_share.c, res/res_mwi_external_ami.c,
	  res/res_pjsip_logger.c, res/res_stasis_device_state.c,
	  res/res_calendar_ews.c, res/res_monitor.c, apps/app_playback.c,
	  res/res_ari_recordings.c, res/res_manager_devicestate.c,
	  res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c,
	  res/res_ari_events.c, res/res_pjsip_dialog_info_body_generator.c,
	  res/res_sorcery_astdb.c, codecs/codec_dahdi.c,
	  apps/app_zapateller.c, pbx/pbx_config.c: Add module support level
	  to ast_module_info structure. Print it in CLI "module show" .
	  ASTERISK-23919 #close Reported by Malcolm Davenport Review:
	  https://reviewboard.asterisk.org/r/3802

2014-07-25 14:47 +0000 [r419563-419567]  Matthew Jordan <mjordan@digium.com>

	* CHANGES, res/ari/ari_model_validators.c,
	  rest-api/api-docs/recordings.json,
	  res/ari/ari_model_validators.h, /, res/res_stasis_recording.c:
	  Multiple revisions 419565-419566 ........ r419565 | mjordan |
	  2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) | 21 lines ARI:
	  report duration values in LiveRecording objects This patch adds
	  three new fields to the LiveRecording model: - total_duration:
	  the total length of the live recording - talking_duration:
	  optional. The duration of talking energy that was detected while
	  the recording was made. - silence_duration: optional. The
	  duration of silence that was detected while the recording was
	  made. These values are reported in the RecordingFinished ARI
	  event. When a DSP is enabled on the channel during the recording
	  - which occurs when the recording is created with
	  max_silence_seconds (indicating that the user actually cares
	  about how much silence is in the file), we will report the
	  talking_duration and silence_duration in addition to the
	  total_duration. Review: https://reviewboard.asterisk.org/r/3770/
	  ASTERISK-24037 #close Reported by: Samuel Galarneau ........
	  r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014)
	  | 1 line Update CHANGES for r419565 ........ Merged revisions
	  419565-419566 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/loader.c, res/res_calendar.c: module loader: Unload modules
	  in reverse order of their start order When Asterisk starts a
	  module (calling its load_module function), it re-orders the
	  module list, sorting it alphabetically. Ostensibly, this was done
	  so that the output of 'module show' listed modules in alphabetic
	  order. This had the unfortunate side effect of making modules
	  with complex usage patterns unloadable. A module that has a large
	  number of modules that depend on it is typically abandoned during
	  the unloading process. This results in its memory not being
	  reclaimed during exit. Generally, this isn't harmful - when the
	  process is destroyed, the operating system will reclaim all
	  memory allocated by the process. Prior to Asterisk 12, we also
	  didn't have many modules with complex dependencies. However, with
	  the advent of ARI and PJSIP, this can make make unloading those
	  modules successfully nearly impossible, and thus tracking memory
	  leaks or ref debug leaks a real pain. While this patch is not a
	  complete overhaul of the module loader - such an effort would be
	  beyond the scope of what could be done for Asterisk 13 - this
	  does make some marginal improvements to the loader such that
	  modules like res_pjsip or res_stasis *may* be made properly
	  un-loadable in the future. 1. The linked list of modules has been
	  replaced with a doubly linked list. This allows traversal of the
	  module list to occur backwards. The module shutdown routine now
	  walks the global list backwards when it attempts to unload
	  modules. 2. The alphabetic reorganization of the module list on
	  startup has been removed. Instead, a started module is placed at
	  the end of the module list. 3. The ast_update_module_list
	  function - which is used by the CLI to display the modules - now
	  does the sorting alphabetically itself. It creates its own linked
	  list and inserts the modules into it in alphabetic order. This
	  allows for the intent of the previous code to be maintained. This
	  patch also contains a fix for res_calendar. Without
	  calendar.conf, the calendar modules were improperly bumping the
	  use count of res_calendar, then failing to load themselves. This
	  patch makes it so that we detect whether or not calendaring is
	  enabled before altering the use count. Review:
	  https://reviewboard.asterisk.org/r/3777/

2014-07-25 10:54 +0000 [r419537-419539]  Joshua Colp <jcolp@digium.com>

	* apps/app_bridgewait.c, /: app_bridgewait: Remove possibility of
	  race condition between channels leaving/joining. Bridges created
	  by app_bridgewait previously had the "dissolve when empty" flag
	  set. This caused the bridge core to destroy them when the last
	  channel had left. This introduced a race condition where we may
	  have a reference to the bridge but it is not actually joinable
	  when we try to join it. This flag has now been removed and the
	  bridge is guaranteed to be joinable at all times. ASTERISK-23987
	  #close Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3836/ ........ Merged
	  revisions 419538 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/bridge.c: bridge: Make "bridge destroy" only available in
	  developer mode and add "all" to "bridge kick". The "bridge
	  destroy" CLI command is invasive to bridges and can leave them in
	  an unexpected state for the users of them. Since this command may
	  be useful for developers it is now only available when developer
	  mode is available. To take its place "all" has been added as a
	  valid option to the "bridge kick" CLI command. It will kick all
	  of the channels in the bridge out. ASTERISK-23987 Reported by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/3840/
	  ........ Merged revisions 419536 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-24 22:48 +0000 [r419520]  Richard Mudgett <rmudgett@digium.com>

	* main/bridge.c, main/bridge_basic.c, main/core_unreal.c,
	  UPGRADE.txt, include/asterisk/channel.h, CHANGES,
	  apps/app_followme.c, apps/app_queue.c, main/cel.c,
	  res/parking/parking_bridge_features.c, apps/app_dial.c,
	  main/channel.c, main/dial.c, main/pbx.c: accountcode: Slightly
	  change accountcode propagation. The previous behavior was to
	  simply set the accountcode of an outgoing channel to the
	  accountcode of the channel initiating the call. It was done this
	  way a long time ago to allow the accountcode set on the SIP/100
	  channel to be propagated to a local channel so the dialplan
	  execution on the Local;2 channel would have the SIP/100
	  accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200
	  Propagating the SIP/100 accountcode to the local channels is very
	  useful. Without any dialplan manipulation, all channels in this
	  call would have the same accountcode. Using dialplan, you can set
	  a different accountcode on the SIP/200 channel either by setting
	  the accountcode on the Local;2 channel or by the Dial
	  application's b(pre-dial), M(macro) or U(gosub) options, or by
	  the FollowMe application's b(pre-dial) option, or by the Queue
	  application's macro or gosub options. Before Asterisk v12, the
	  altered accountcode on SIP/200 will remain until the local
	  channels optimize out and the accountcode would change to the
	  SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount
	  support but ultimately had to punt on the support. The
	  peeraccount support was rendered useless because of how the CDR
	  code needed to unconditionally force the caller's accountcode
	  onto the peer channel's accountcode. The CEL events were thus
	  intentionally made to always use the channel's accountcode as the
	  peeraccount value. With the arrival of Asterisk v12, the
	  situation has improved somewhat so peeraccount support can be
	  made to work. Using the indicated example, the the accountcode
	  values become as follows when the peeraccount is set on SIP/100
	  before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 --->
	  SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer:
	  200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already
	  has an accountcode it can only change by the following explicit
	  user actions: 1) A channel originate method that can specify an
	  accountcode to use. 2) The calling channel propagating its
	  non-empty peeraccount or its non-empty accountcode if the
	  peeraccount was empty to the outgoing channel's accountcode
	  before initiating the dial. e.g., Dial and FollowMe. The
	  exception to this propagation method is Queue. Queue will only
	  propagate peeraccounts this way only if the outgoing channel does
	  not have an accountcode. 3) Dialplan using CHANNEL(accountcode).
	  4) Dialplan using CHANNEL(peeraccount) on the other end of a
	  local channel pair. If a channel does not have an accountcode it
	  can get one from the following places: 1) The channel driver's
	  configuration at channel creation. 2) Explicit user action as
	  already indicated. 3) Entering a basic or stasis-mixing bridge
	  from a peer channel's peeraccount value. You can specify the
	  accountcode for an outgoing channel by setting the
	  CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue
	  applications. Queue adds the wrinkle that it will not overwrite
	  an existing accountcode on the outgoing channel with the calling
	  channels values. Accountcode and peeraccount values propagate to
	  an outgoing channel before dialing. Accountcodes also propagate
	  when channels enter or leave a basic or stasis-mixing bridge. The
	  peeraccount value only makes sense for mixing bridges with two
	  channels; it is meaningless otherwise. * Made peeraccount
	  functional by changing accountcode propagation as described
	  above. * Fixed CEL extracting the wrong ie value for the
	  peeraccount. This was done intentionally in Asterisk v1.8 when
	  that version had to punt on peeraccount. * Fixed a few places
	  dealing with accountcodes that were reading from channels without
	  the lock held. AFS-65 #close Review:
	  https://reviewboard.asterisk.org/r/3601/

2014-07-24 21:01 +0000 [r419504]  Michael L. Young <elgueromexicano@gmail.com>

	* main/db.c, include/asterisk/astdb.h: core/db: Revert Patch Added
	  In Attempt To Improve I/O Performance Reverting the patch since
	  it was causing a regression and after fixing the regression,
	  there were no performance gains. At least based on my method for
	  measurement. ASTERISK-24050 Review:
	  https://reviewboard.asterisk.org/r/3841/

2014-07-24 17:50 +0000 [r419438-419439]  Corey Farrell <git@cfware.com>

	* include/asterisk/astobj.h: Deprecate astobj.h This flags astobj.h
	  as deprecated, warns people to use astobj2.h instead. Only
	  netsock.c (also deprecated) still uses astobj.h. ASTERISK-24069
	  #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3818/

	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
	  complete upgrade to ao2 This change upgrades sip_registry and
	  sip_subscription_mwi to astobj2. ASTERISK-24067 #close Reported
	  by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3759/

2014-07-24 16:52 +0000 [r419377]  Jason Parker <jparker@digium.com>

	* addons/chan_ooh323.c, /: Don't cause Asterisk to exit if
	  ooh323.conf not found. (closes issue ASTERISK-23814) ........
	  Merged revisions 419374 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 419375 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 419376 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-24 15:20 +0000 [r419358]  Matthew Jordan <mjordan@digium.com>

	* main/devicestate.c, channels/chan_pjsip.c: device state: Update
	  the core to report ONHOLD if a channel is on hold In Asterisk, it
	  is possible for a device to have a status of ONHOLD. This is not
	  typically an easy thing to determine, as a channel being on hold
	  is not a direct channel state. Typically, this has to be
	  calculated outside of the core independently in channel drivers,
	  notably, chan_sip and chan_pjsip. Both of these channel drivers
	  already have to calculate device state in a fashion more complex
	  than the core can handle, as they aggregate all state of all
	  channels associated with a peer/endpoint; they also independently
	  track whether or not one of those channels is currently on hold
	  and mark the device state appropriately. In 12+, we now have the
	  ability to report an AST_DEVICE_ONHOLD state for all channels
	  that defer their device state to the core. This is due to channel
	  hold state actually now being tracked on the channel itself. If a
	  channel driver defers its device state to the core (which many,
	  such as DAHDI, IAX2, and others do in most situations), the
	  device state core already goes out to get a channel associated
	  with the device. As such, it can now also factor the channel hold
	  state in its calculation. This patch adds this logic to the
	  device state core. It also uses an existing mapping between
	  device state and channel state to handle more channel states.
	  chan_pjsip has been updated slightly as well to make use of this
	  (as it was, for some reason, reporting a channel state of BUSY as
	  a device state of INUSE, which feels slightly wrong). Review:
	  https://reviewboard.asterisk.org/r/3771/ ASTERISK-24038 #close

2014-07-24 13:00 +0000 [r419342]  Kinsey Moore <kmoore@digium.com>

	* include/asterisk/manager.h, doc/appdocsxml.dtd, main/xmldoc.c,
	  main/manager_bridges.c, main/manager.c,
	  include/asterisk/xmldoc.h, main/config_options.c: AMI: Allow for
	  command response documentation Allow for responses to AMI
	  actions/commands to be documented properly in XML and displayed
	  via the CLI. Response events are documented exactly as standard
	  AMI events are documented. Review:
	  https://reviewboard.asterisk.org/r/3812/

2014-07-23 16:46 +0000 [r419319]  Matthew Jordan <mjordan@digium.com>

	* main/endpoints.c, tests/test_stasis_endpoints.c, /: endpoints:
	  Fix failing unit tests from r419196 This patch does two things:
	  (1) It updates the unit tests to expect additional stasis
	  messages. More messages are now sent to the endpoint topic, due
	  to forwarding all channel messages and the forwarding
	  relationship set up between endpoints themselves. (2) Remove the
	  technology forwarding subscription during ast_endpoint_shutdown.
	  This prevents an improper double shutdown of an endpoint from
	  occurring. ........ Merged revisions 419318 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-23 14:00 +0000 [r419286]  Scott Griepentrog <sgriepentrog@digium.com>

	* apps/app_voicemail.c, /: app_voicemail: use a consistent
	  generator string When updating voicemail.conf when a user changes
	  their pin, change the generator string to be the same as the
	  module name when reading so that the same config_hook will be
	  called. Review: https://reviewboard.asterisk.org/r/3837/ ........
	  Merged revisions 419284 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 419285 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-23 01:28 +0000 [r419268]  Corey Farrell <git@cfware.com>

	* main/manager.c, res/res_fax.c: res_fax: unregister manager
	  actions on unload * Unregister manager actions FAXSessions,
	  FAXSession and FAXStats at unload. * Update ast_manager_register2
	  use ao2_t_alloc tagged with the action name. ASTERISK-24058
	  #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3831/

2014-07-22 20:22 +0000 [r419222-419252]  Michael L. Young <elgueromexicano@gmail.com>

	* CHANGES, main/bridge_channel.c: core/bridge_channel: Substitute
	  Variables In Features Application Map Say you wanted to include
	  variables in an application map and have those variables
	  substituted and passed along to the application being executed;
	  currently this does not happen. This patch adds this ability to
	  pass channel variable values to an application before being
	  executed. ASTERISK-22608 #close Reported by: Michael L. Young
	  patches: features_substitute_arguments_v2.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/3819/

	* CHANGES, apps/app_mixmonitor.c: apps/app_mixmonitor: Add Options
	  To Play Beep At Start Or Stop We have a new periodic beep feature
	  but sometimes a user needs some sort of feedback, without the
	  need to have a periodic beep during the recording, to let them
	  know that MixMonitor started recording or ended the recording.
	  The use case where this patch is being used is when using Dynamic
	  Features to start and end MixMonitor. This patch adds an option
	  to play a beep when MixMonitor starts and an option to play a
	  beep when MixMonitor ends. ASTERISK-24051 #close Reported by:
	  Michael L. Young patches: mixmonitor-play-beep-start-stop.diff
	  uploaded by Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/3820/

	* main/db.c, include/asterisk/astdb.h: core/db: Improve I/O When
	  Updating Rows When updating a row, we are currently doing an
	  INSERT OR REPLACE INTO. The downside to this is that the row is
	  deleted if it exists and then a new row is inserted. So, we are
	  hitting the disk twice. One for the deletion and one for the
	  insertion. This patch changes this statement to an INSERT INTO
	  and if the insert fails because a row with that key exists, we
	  will IGNORE the failure. Then we will attempt to perform an
	  UPDATE on the existing row if that row wasn't just INSERTed.
	  ASTERISK-24050 #close Reported by: Michael L. Young patches:
	  astdb-insert-update-io-help_trunk_v2.diff uploaded by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/3815/

2014-07-22 17:10 +0000 [r419206]  Richard Mudgett <rmudgett@digium.com>

	* codecs/codec_speex.c: codec_speex: Fix trashing normal static
	  frame for AST_FRAME_CNG. Made use a local static frame to
	  generate the AST_FRAME_CNG frame when silence starts. I don't
	  think the handling of the AST_FRAME_CNG has ever really worked
	  because there doesn't seem to be any consumers of it. Review:
	  https://reviewboard.asterisk.org/r/3813/

2014-07-22 16:20 +0000 [r419203]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/endpoints.h,
	  rest-api/api-docs/applications.json, include/asterisk/xmpp.h,
	  main/channel_internal_api.c, channels/chan_motif.c,
	  include/asterisk/channel.h, res/ari/resource_applications.h,
	  res/res_xmpp.c, channels/chan_iax2.c, main/endpoints.c,
	  channels/chan_pjsip.c, main/channel.c,
	  res/ari/resource_endpoints.c, /, channels/chan_sip.c: ARI: Fix
	  endpoint/channel subscription issues; allow for subscriptions to
	  tech This patch serves two purposes: (1) It fixes some bugs with
	  endpoint subscriptions not reporting all of the channel events
	  (2) It serves as the preliminary work needed for ASTERISK-23692,
	  which allows for sending/receiving arbitrary out of call text
	  messages through ARI in a technology agnostic fashion. The
	  messaging functionality described on ASTERISK-23692 requires two
	  things: (1) The ability to send/receive messages associated with
	  an endpoint. This is relatively straight forwards with the
	  endpoint core in Asterisk now. (2) The ability to send/receive
	  messages associated with a technology and an arbitrary technology
	  defined URI. This is less straight forward, as endpoints are
	  formed from a tech + resource pair. We don't have a mechanism to
	  note that a technology that *may* have endpoints exists. This
	  patch provides such a mechanism, and fixes a few bugs along the
	  way. The first major bug this patch fixes is the forwarding of
	  channel messages to their respective endpoints. Prior to this
	  patch, there were two problems: (1) Channel caching messages
	  weren't forwarded. Thus, the endpoints missed most of the
	  interesting bits (such as channel creation, destruction, state
	  changes, etc.) (2) Channels weren't associated with their
	  endpoint until after creation. This resulted in endpoints missing
	  the channel creation message, which limited the usefulness of the
	  subscription in the first place (a major use case being 'tell me
	  when this endpoint has a channel'). Unfortunately, this meant
	  another parameter to ast_channel_alloc. Since not all channel
	  technologies support an ast_endpoint, this patch makes such a
	  call optional and opts for a new function,
	  ast_channel_alloc_with_endpoint. When endpoints are created, they
	  will implicitly create a technology endpoint for their technology
	  (if one does not already exist). A technology endpoint is special
	  in that it has no state, cannot have channels created for it,
	  cannot be created explicitly, and cannot be destroyed except on
	  shutdown. It does, however, have all messages from other
	  endpoints in its technology forwarded to it. Combined with the
	  bug fixes, we now have Stasis messages being properly forwarded.
	  Consider the following scenario: two PJSIP endpoints (foo and
	  bar), where bar has a single channel associated with it and foo
	  has two channels associated with it. The messages would be
	  forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint
	  PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP /
	  channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the
	  applications resource, can: - subscribe to endpoint:PJSIP/foo and
	  get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and
	  endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get
	  notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar -
	  subscribe to endpoint:PJSIP and get notifications for channels
	  PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints
	  PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes,
	  it never has events itself. It merely provides an aggregation
	  point for all other endpoints in its technology (which in turn
	  aggregate all channel messages associated with that endpoint).
	  This patch also adds endpoints to res_xmpp and chan_motif,
	  because the actual messaging work will need it (messaging without
	  XMPP is just sad). Review:
	  https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........
	  Merged revisions 419196 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-22 14:36 +0000 [r419180]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: chan_iax2: Restore previous behavior of
	  iax2_best_codec. The iax2_best_codec function was changed to
	  convert the formats into a format compatibilities structure and
	  grab the first format from it. The resulting order differs from
	  the previous order of iax2_best_codec which causes unexpected
	  formats to get chosen (such as g723). This commit brings back the
	  old behavior of iax2_best_codec by having a specified preference
	  list. Review: https://reviewboard.asterisk.org/r/3835/

2014-07-22 14:22 +0000 [r419110-419175]  Kinsey Moore <kmoore@digium.com>

	* addons/ooh323c/src/printHandler.c, tests/test_sorcery_realtime.c,
	  tests/test_json.c, addons/ooh323c/src/ooq931.c,
	  tests/test_astobj2_thrash.c, addons/chan_ooh323.c, /,
	  tests/test_optional_api.c, tests/test_abstract_jb.c,
	  apps/app_meetme.c, tests/test_logger.c, tests/test_event.c,
	  tests/test_hashtab_thrash.c, res/res_mwi_external_ami.c,
	  tests/test_sorcery.c, res/res_corosync.c,
	  tests/test_voicemail_api.c, tests/test_aoc.c,
	  tests/test_astobj2.c, tests/test_config.c: Fix more dev-mode
	  build issues ........ Merged revisions 419129 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 419162 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 419163 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/dial.c: Dial API: Prevent crash on NULL cap This prevents a
	  crash in the Dial API triggered by use of the Page() application
	  where a format capability struct was used before checking whether
	  it was NULL. ASTERISK-24074 #close

	* channels/chan_skinny.c, tests/test_core_format.c: Fix build in
	  dev-mode

2014-07-21 16:26 +0000 [r419109]  Jonathan Rose <jrose@digium.com>

	* channels/chan_iax2.c: chan_iax2: Restore codec choice behavior
	  from media formats branch After merging the media formats branch,
	  chan_iax2 was discarding codec preferences for the purpose of
	  choosing which codec a channel would use once a call started.
	  This patch restores the Asterisk 1.8-12 codec choice behaviors.
	  ASTERISK-23958 #close Review:
	  https://reviewboard.asterisk.org/r/3800/

2014-07-21 16:09 +0000 [r419093]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: chan_iax2: Only send mini frames if the
	  underlying format has not changed, not if it has. ASTERISK-24072
	  #close Reported by: Matt Jordan

2014-07-21 14:49 +0000 [r419077]  Sean Bright <sean@malleable.com>

	* configure, configure.ac: Fix build when pjproject is installed in
	  a non-standard location. When configuring Asterisk to build
	  against a version of pjproject installed in a non-standard
	  location, the checks for "PJSIP Transaction Group Lock Support"
	  and "PJSIP Media Stream Replacement Support" fail. This is
	  because these secondary checks are not taking the CFLAGS and LIBS
	  returned by the pkg-config check into account. Review:
	  https://reviewboard.asterisk.org/r/3830

2014-07-21 08:41 +0000 [r419060]  Corey Farrell <git@cfware.com>

	* channels/sig_analog.c, res/res_smdi.c, channels/chan_motif.c,
	  include/asterisk/smdi.h, apps/app_voicemail.c,
	  channels/chan_dahdi.c: res_smdi: convert to astobj2 Remove
	  functions: ast_smdi_interface_unref ast_smdi_md_message_putback
	  ast_smdi_mwi_message_putback ast_smdi_md_message destructor
	  ast_smdi_mwi_message destructor Includes for astobj.h are removed
	  everywhere it's possible. ASTERISK-24066 #close Review:
	  https://reviewboard.asterisk.org/r/3758/

2014-07-20 22:06 +0000 [r419044]  Matthew Jordan <mjordan@digium.com>

	* apps/app_confbridge.c, res/ari/resource_channels.c,
	  include/asterisk/rtp_engine.h, include/asterisk/slinfactory.h,
	  res/res_calendar.c, codecs/codec_g722.c,
	  include/asterisk/res_pjsip_session.h, main/frame.c,
	  codecs/ex_lpc10.h, apps/app_dictate.c, res/res_fax.c,
	  apps/app_echo.c, include/asterisk/slin.h, codecs/codec_g726.c,
	  formats/format_ogg_vorbis.c, codecs/codec_gsm.c,
	  codecs/ex_alaw.h, formats/format_wav_gsm.c,
	  channels/iax2/provision.c, channels/chan_iax2.c,
	  res/res_format_attr_h264.c, main/data.c, main/manager.c,
	  include/asterisk/audiohook.h, formats/format_pcm.c,
	  main/config_options.c, res/res_format_attr_silk.c,
	  main/bridge_channel.c, res/res_speech.c, channels/chan_pjsip.c,
	  res/res_clioriginate.c, formats/format_g729.c,
	  channels/chan_unistim.c, res/res_rtp_asterisk.c,
	  include/asterisk/smoother.h (added), main/rtp_engine.c,
	  addons/format_mp3.c, formats/format_wav.c,
	  apps/confbridge/conf_chan_record.c, include/asterisk/speech.h,
	  codecs/ex_adpcm.h, channels/iax2/codec_pref.c (added),
	  include/asterisk/codec.h (added), formats/format_siren7.c,
	  include/asterisk/file.h, channels/chan_dahdi.c,
	  include/asterisk/image.h, funcs/func_channel.c,
	  main/abstract_jb.c, formats/format_h263.c, codecs/codec_dahdi.c,
	  main/dsp.c, apps/app_voicemail.c, apps/app_jack.c,
	  funcs/func_talkdetect.c, channels/chan_vpb.cc,
	  channels/chan_sip.c, formats/format_sln.c,
	  tests/test_abstract_jb.c, codecs/codec_alaw.c, UPGRADE.txt,
	  main/smoother.c (added), codecs/ex_speex.h,
	  channels/chan_console.c, apps/app_talkdetect.c,
	  main/format_pref.c (removed), main/indications.c,
	  include/asterisk/format_cap.h, main/media_index.c,
	  apps/app_agent_pool.c, res/res_pjsip_session.c, main/cli.c,
	  res/res_format_attr_celt.c, channels/chan_skinny.c,
	  tests/test_core_format.c (added), funcs/func_frame_trace.c,
	  res/res_pjsip/pjsip_configuration.c, main/file.c,
	  include/asterisk/frame.h, formats/format_g726.c,
	  apps/app_mixmonitor.c, channels/chan_mgcp.c, main/sorcery.c,
	  codecs/ex_ilbc.h, codecs/codec_lpc10.c, tests/test_format_cache.c
	  (added), apps/app_meetme.c, main/translate.c,
	  apps/app_originate.c, res/parking/parking_applications.c,
	  apps/app_ices.c, channels/iax2/parser.c, res/res_rtp_multicast.c,
	  pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_vox.c,
	  main/format_cap.c, tests/test_cel.c, include/asterisk/format.h,
	  formats/format_h264.c, apps/app_chanspy.c, apps/app_nbscat.c,
	  addons/chan_ooh323.c, bridges/bridge_holding.c,
	  channels/iax2/include/codec_pref.h (added), codecs/codec_adpcm.c,
	  apps/app_waitforsilence.c, res/res_pjsip_sdp_rtp.c,
	  addons/chan_ooh323.h, bridges/bridge_simple.c,
	  apps/app_alarmreceiver.c, bridges/bridge_softmix.c,
	  res/res_stasis_snoop.c, main/sounds_index.c, main/core_local.c,
	  main/codec_builtin.c (added), include/asterisk/format_cache.h
	  (added), apps/app_speech_utils.c, res/res_format_attr_opus.c,
	  include/asterisk/abstract_jb.h, main/channel.c,
	  include/asterisk/format_compatibility.h (added), apps/app_mp3.c,
	  tests/test_voicemail_api.c, channels/chan_alsa.c, main/app.c,
	  formats/format_g723.c, codecs/codec_ilbc.c, tests/test_config.c,
	  formats/format_gsm.c, apps/app_milliwatt.c, codecs/ex_ulaw.h,
	  main/asterisk.c, include/asterisk/res_pjsip.h, main/format.c,
	  main/ccss.c, main/bridge.c, codecs/codec_speex.c,
	  include/asterisk/format_pref.h (removed), apps/app_record.c,
	  main/slinfactory.c, res/res_adsi.c, main/core_unreal.c,
	  res/ari/resource_bridges.c, include/asterisk/callerid.h,
	  channels/pjsip/dialplan_functions.c, main/dial.c,
	  channels/dahdi/bridge_native_dahdi.c, main/format_cache.c
	  (added), include/asterisk/mod_format.h, apps/app_sms.c,
	  codecs/codec_resample.c, main/format_compatibility.c (added),
	  main/audiohook.c, formats/format_jpeg.c, res/res_stasis.c,
	  formats/format_g719.c, include/asterisk/translate.h,
	  funcs/func_speex.c, codecs/codec_a_mu.c,
	  channels/iax2/format_compatibility.c (added),
	  apps/app_festival.c, main/channel_internal_api.c,
	  tests/test_format_api.c (removed), codecs/ex_g722.h,
	  main/utils.c, res/ari/resource_sounds.c,
	  res/res_format_attr_h263.c, codecs/ex_g726.h,
	  include/asterisk/_private.h, channels/chan_oss.c,
	  channels/chan_misdn.c, main/codec.c (added), main/callerid.c,
	  addons/ooh323cDriver.c, apps/app_amd.c, codecs/codec_ulaw.c,
	  main/image.c, channels/chan_nbs.c, bridges/bridge_native_rtp.c,
	  channels/iax2/include/format_compatibility.h (added),
	  formats/format_siren14.c, res/res_fax_spandsp.c,
	  addons/chan_mobile.c, addons/ooh323cDriver.h,
	  channels/sip/include/sip.h, tests/test_format_cap.c (added),
	  channels/chan_multicast_rtp.c, include/asterisk/vector.h,
	  channels/chan_bridge_media.c, apps/app_fax.c,
	  main/bridge_basic.c, apps/app_test.c, include/asterisk/channel.h,
	  include/asterisk/data.h, tests/test_core_codec.c (added),
	  res/res_musiconhold.c, codecs/ex_gsm.h, formats/format_ilbc.c,
	  include/asterisk/config_options.h, channels/chan_phone.c,
	  include/asterisk/bridge_channel.h, apps/app_dumpchan.c,
	  channels/chan_motif.c, res/res_agi.c: media formats: re-architect
	  handling of media for performance improvements In the old times
	  media formats were represented using a bit field. This was fast
	  but had a few limitations. 1. Asterisk was limited in how many
	  formats it could handle. 2. Formats, being a bit field, could not
	  include any attribute information. A format was strictly its
	  type, e.g., "this is ulaw". This was changed in Asterisk 10 (see
	  https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
	  for notes on that work) which led to the creation of the
	  ast_format structure. This structure allowed Asterisk to handle
	  attributes and bundle information with a format. Additionally,
	  ast_format_cap was created to act as a container for multiple
	  formats that, together, formed the capability of some entity.
	  Another mechanism was added to allow logic to be registered which
	  performed format attribute negotiation. Everywhere throughout the
	  codebase Asterisk was changed to use this strategy.
	  Unfortunately, in software, there is no free lunch. These new
	  capabilities came at a cost. Performance analysis and profiling
	  showed that we spend an inordinate amount of time comparing,
	  copying, and generally manipulating formats and their related
	  structures. Basic prototyping has shown that a reasonably large
	  performance improvement could be made in this area. This patch is
	  the result of that project, which overhauled the media format
	  architecture and its usage in Asterisk to improve performance.
	  Generally, the new philosophy for handling formats is as follows:
	  * The ast_format structure is reference counted. This removed a
	  large amount of the memory allocations and copying that was done
	  in prior versions. * In order to prevent race conditions while
	  keeping things performant, the ast_format structure is immutable
	  by convention and lock-free. Violate this tenet at your peril! *
	  Because formats are reference counted, codecs are also reference
	  counted. The Asterisk core generally provides built-in codecs and
	  caches the ast_format structures created to represent them.
	  Generally, to prevent inordinate amounts of module reference
	  bumping, codecs and formats can be added at run-time but cannot
	  be removed. * All compatibility with the bit field representation
	  of codecs/formats has been moved to a compatibility API. The
	  primary user of this representation is chan_iax2, which must
	  continue to maintain its bit-field usage of formats for
	  interoperability concerns. * When a format is negotiated with
	  attributes, or when a format cannot be represented by one of the
	  cached formats, a new format object is created or cloned from an
	  existing format. That format may have the same codec underlying
	  it, but is a different format than a version of the format with
	  different attributes or without attributes. * While formats are
	  reference counted objects, the reference count maintained on the
	  format should be manipulated with care. Formats are generally
	  cached and will persist for the lifetime of Asterisk and do not
	  explicitly need to have their lifetime modified. An exception to
	  this is when the user of a format does not know where the format
	  came from *and* the user may outlive the provider of the format.
	  This occurs, for example, when a format is read from a channel:
	  the channel may have a format with attributes (hence, non-cached)
	  and the user of the format may last longer than the channel (if
	  the reference to the channel is released prior to the format's
	  reference). For more information on this work, see the API design
	  notes:
	  https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
	  Finally, this work was the culmination of a large number of
	  developer's efforts. Extra thanks goes to Corey Farrell, who took
	  on a large amount of the work in the Asterisk core, chan_sip, and
	  was an invaluable resource in peer reviews throughout this
	  project. There were a substantial number of patches contributed
	  during this work; the following issues/patch names simply reflect
	  some of the work (and will cause the release scripts to give
	  attribution to the individuals who work on them). Reviews:
	  https://reviewboard.asterisk.org/r/3814
	  https://reviewboard.asterisk.org/r/3808
	  https://reviewboard.asterisk.org/r/3805
	  https://reviewboard.asterisk.org/r/3803
	  https://reviewboard.asterisk.org/r/3801
	  https://reviewboard.asterisk.org/r/3798
	  https://reviewboard.asterisk.org/r/3800
	  https://reviewboard.asterisk.org/r/3794
	  https://reviewboard.asterisk.org/r/3793
	  https://reviewboard.asterisk.org/r/3792
	  https://reviewboard.asterisk.org/r/3791
	  https://reviewboard.asterisk.org/r/3790
	  https://reviewboard.asterisk.org/r/3789
	  https://reviewboard.asterisk.org/r/3788
	  https://reviewboard.asterisk.org/r/3787
	  https://reviewboard.asterisk.org/r/3786
	  https://reviewboard.asterisk.org/r/3784
	  https://reviewboard.asterisk.org/r/3783
	  https://reviewboard.asterisk.org/r/3778
	  https://reviewboard.asterisk.org/r/3774
	  https://reviewboard.asterisk.org/r/3775
	  https://reviewboard.asterisk.org/r/3772
	  https://reviewboard.asterisk.org/r/3761
	  https://reviewboard.asterisk.org/r/3754
	  https://reviewboard.asterisk.org/r/3753
	  https://reviewboard.asterisk.org/r/3751
	  https://reviewboard.asterisk.org/r/3750
	  https://reviewboard.asterisk.org/r/3748
	  https://reviewboard.asterisk.org/r/3747
	  https://reviewboard.asterisk.org/r/3746
	  https://reviewboard.asterisk.org/r/3742
	  https://reviewboard.asterisk.org/r/3740
	  https://reviewboard.asterisk.org/r/3739
	  https://reviewboard.asterisk.org/r/3738
	  https://reviewboard.asterisk.org/r/3737
	  https://reviewboard.asterisk.org/r/3736
	  https://reviewboard.asterisk.org/r/3734
	  https://reviewboard.asterisk.org/r/3722
	  https://reviewboard.asterisk.org/r/3713
	  https://reviewboard.asterisk.org/r/3703
	  https://reviewboard.asterisk.org/r/3689
	  https://reviewboard.asterisk.org/r/3687
	  https://reviewboard.asterisk.org/r/3674
	  https://reviewboard.asterisk.org/r/3671
	  https://reviewboard.asterisk.org/r/3667
	  https://reviewboard.asterisk.org/r/3665
	  https://reviewboard.asterisk.org/r/3625
	  https://reviewboard.asterisk.org/r/3602
	  https://reviewboard.asterisk.org/r/3519
	  https://reviewboard.asterisk.org/r/3518
	  https://reviewboard.asterisk.org/r/3516
	  https://reviewboard.asterisk.org/r/3515
	  https://reviewboard.asterisk.org/r/3512
	  https://reviewboard.asterisk.org/r/3506
	  https://reviewboard.asterisk.org/r/3413
	  https://reviewboard.asterisk.org/r/3410
	  https://reviewboard.asterisk.org/r/3387
	  https://reviewboard.asterisk.org/r/3388
	  https://reviewboard.asterisk.org/r/3389
	  https://reviewboard.asterisk.org/r/3390
	  https://reviewboard.asterisk.org/r/3321
	  https://reviewboard.asterisk.org/r/3320
	  https://reviewboard.asterisk.org/r/3319
	  https://reviewboard.asterisk.org/r/3318
	  https://reviewboard.asterisk.org/r/3266
	  https://reviewboard.asterisk.org/r/3265
	  https://reviewboard.asterisk.org/r/3234
	  https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close
	  Reported by: mjordan media_formats_translation_core.diff uploaded
	  by kharwell (License 6464) rb3506.diff uploaded by mjordan
	  (License 6283) media_format_app_file.diff uploaded by kharwell
	  (License 6464) misc-2.diff uploaded by file (License 5000)
	  chan_mild-3.diff uploaded by file (License 5000)
	  chan_obscure.diff uploaded by file (License 5000) jingle.diff
	  uploaded by file (License 5000) funcs.diff uploaded by file
	  (License 5000) formats.diff uploaded by file (License 5000)
	  core.diff uploaded by file (License 5000) bridges.diff uploaded
	  by file (License 5000) mf-codecs-2.diff uploaded by file (License
	  5000) mf-app_fax.diff uploaded by file (License 5000)
	  mf-apps-3.diff uploaded by file (License 5000)
	  media-formats-3.diff uploaded by file (License 5000)
	  ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License
	  5909) rb3689.patch uploaded by mjordan (License 6283)
	  ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283)
	  mf-attributes-3.diff uploaded by file (License 5000)
	  ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by
	  coreyfarrell (License 5909) rb3800.patch uploaded by jrose
	  (License 6182) chan_sip.diff uploaded by mjordan (License 6283)
	  rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959
	  #close Tested by: sgriepentrog, mjordan, coreyfarrell
	  sip_cleanup.diff uploaded by opticron (License 6273)
	  chan_sip_caps.diff uploaded by mjordan (License 6283)
	  rb3751.patch uploaded by coreyfarrell (License 5909)
	  chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960
	  #close Tested by: opticron direct_media.diff uploaded by opticron
	  (License 6273) pjsip-direct-media.diff uploaded by file (License
	  5000) format_cap_remove.diff uploaded by opticron (License 6273)
	  media_format_fixes.diff uploaded by opticron (License 6273)
	  chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966
	  #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti
	  (License 5621) chan_dahdi.diff uploaded by file (License 5000)
	  ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron,
	  file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by
	  rmudgett (License 5621) moh_cleanup.diff uploaded by opticron
	  (License 6273) bridge_leak.diff uploaded by opticron (License
	  6273) translate.diff uploaded by file (License 5000) rb3795.patch
	  uploaded by rmudgett (License 5621) tls_fix.diff uploaded by
	  mjordan (License 6283) fax-mf-fix-2.diff uploaded by file
	  (License 5000) rtp_transfer_stuff uploaded by mjordan (License
	  6283) rb3787.patch uploaded by rmudgett (License 5621)
	  media-formats-explicit-translate-format-3.diff uploaded by file
	  (License 5000) format_cache_case_fix.diff uploaded by opticron
	  (License 6273) rb3774.patch uploaded by rmudgett (License 5621)
	  rb3775.patch uploaded by rmudgett (License 5621)
	  rtp_engine_fix.diff uploaded by opticron (License 6273)
	  rtp_crash_fix.diff uploaded by opticron (License 6273)
	  rb3753.patch uploaded by mjordan (License 6283) rb3750.patch
	  uploaded by mjordan (License 6283) rb3748.patch uploaded by
	  rmudgett (License 5621) media_format_fixes.diff uploaded by
	  opticron (License 6273) rb3740.patch uploaded by mjordan (License
	  6283) rb3739.patch uploaded by mjordan (License 6283)
	  rb3734.patch uploaded by mjordan (License 6283) rb3689.patch
	  uploaded by mjordan (License 6283) rb3674.patch uploaded by
	  coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell
	  (License 5909) rb3667.patch uploaded by coreyfarrell (License
	  5909) rb3665.patch uploaded by mjordan (License 6283)
	  rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch
	  uploaded by coreyfarrell (License 5909)
	  format_compatibility-2.diff uploaded by file (License 5000)
	  core.diff uploaded by file (License 5000)

2014-07-18 21:48 +0000 [r419022]  Matthew Jordan <mjordan@digium.com>

	* rest-api/api-docs/recordings.json, res/ari/resource_recordings.c,
	  res/stasis_recording/stored.c, res/res_ari_recordings.c, /,
	  include/asterisk/stasis_app_recording.h,
	  res/ari/resource_recordings.h, CHANGES: ari: Add a copy operation
	  for stored recordings This patch adds a new operation for stored
	  recordings, copy. It takes an existing stored recording and makes
	  a copy of it in the same directory or a relative directory under
	  the stored recording directory.
	  /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}
	  This is particularly useful for voicemail-esque applications,
	  which may need to copy or move recordings around a directory
	  structure. Review: https://reviewboard.asterisk.org/r/3768/
	  ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam
	  Galarneau ........ Merged revisions 419021 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-18 21:25 +0000 [r418997-419020]  Corey Farrell <git@cfware.com>

	* main/stasis_message_router.c, /: stasis: fix call to ao2_t_alloc
	  for stasis_message_router_create This fixes a build failure
	  introduced by r3821. struct stasis_topic is opaque, so
	  topic->name is unavailable. Switch to using stasis_topic_name().
	  ........ Merged revisions 419019 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis.c, main/stasis_cache_pattern.c,
	  main/stasis_message.c, main/stasis_message_router.c, /: stasis:
	  use ao2_t_alloc for certain object allocators Add tags to stasis
	  objects using the name. This makes it easier to track the source
	  of certain stasis ref leaks. Review:
	  https://reviewboard.asterisk.org/r/3821/ ........ Merged
	  revisions 418996 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-18 19:07 +0000 [r418980]  Kinsey Moore <kmoore@digium.com>

	* res/res_fax_spandsp.c: Fix build in dev-mode

2014-07-18 17:55 +0000 [r418961-418963]  Scott Griepentrog <sgriepentrog@digium.com>

	* res/res_pjsip_pubsub.c, main/astobj2.c,
	  include/asterisk/astobj2.h, main/logger.c, main/utils.c: astobj2:
	  assert on invalid ref and backtrace cleanup If a reference count
	  goes negative, instead of just logging that fact, be more helpful
	  with a backtrace and an assert that will DO_CRASH. This patch
	  also removes the duplicate ao2_bt() function and cleans up
	  extraneous usage of the ast_log_backtrace() call. Review:
	  https://reviewboard.asterisk.org/r/3765/

	* /, channels/chan_sip.c: media formats: fix ref leak of peer for
	  mwi subscription Holding a reference to the peer during mwi
	  subscriptions resulted in a circular reference because the final
	  event message would not be sent until destruction of the peer.
	  Instead, pass the name of the peer to the event callback so that
	  it can fail gracefully after the peer has gone. ASTERISK-23959
	  Review: https://reviewboard.asterisk.org/r/3754/ ........ Merged
	  revisions 418636 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/features_config.c: feature_config: insure featuregroups
	  and applicationmaps are initialized If the features.conf is
	  missing, the cfg->featurgroups and cfg->applicationmaps is not
	  initialized, resulting in assert on ao2_find of a null container.
	  This patch changes the initialization call and adds asserts for a
	  safeguard. Review: https://reviewboard.asterisk.org/r/3809/
	  ........ Merged revisions 418886 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-18 16:47 +0000 [r418938]  Richard Mudgett <rmudgett@digium.com>

	* funcs/func_audiohookinherit.c, /: func_audiohookinherit.c: Fixup
	  some XML documentation wording. ........ Merged revisions 418937
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-18 16:28 +0000 [r418911-418936]  Jonathan Rose <jrose@digium.com>

	* main/channel.c, funcs/func_audiohookinherit.c, /,
	  include/asterisk/audiohook.h, main/framehook.c, res/res_fax.c,
	  main/bridge_basic.c, include/asterisk/res_fax.h,
	  bridges/bridge_native_rtp.c, main/audiohook.c, CHANGES,
	  include/asterisk/framehook.h, res/res_pjsip_refer.c: Channels:
	  Masquerades to automatically move frame/audio hooks Whenever
	  possible, audiohooks and framehooks will now be copied over to
	  the channel that the masquerading channel gets cloned into. This
	  should occur for all audiohooks and most framehooks. As a result,
	  in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
	  deprecated and its behavior is essentially the new default for
	  all audiohooks, plus some additional audiohooks/framehooks.
	  Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged
	  revisions 418914 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_fax.c, include/asterisk/res_fax.h, CHANGES,
	  res/res_fax.exports.in, res/res_fax_spandsp.c: res_fax: Provide
	  AMI equivalents for fax CLI commands Specifically the following
	  equivalents were created: fax show session -> FAXSession fax show
	  sessions -> FAXSessions fax show stats -> FAXStats Review:
	  https://reviewboard.asterisk.org/r/3666/

2014-07-18 00:11 +0000 [r418893-418895]  Sean Bright <sean@malleable.com>

	* config.sub, menuselect/config.guess, menuselect/config.sub,
	  config.guess: Update config.guess and config.sub

	* autoconf/ast_ext_tool_check.m4: Add missing file from previous
	  commit.

	* menuselect/aclocal.m4, menuselect/configure,
	  menuselect/acinclude.m4 (removed), menuselect/bootstrap.sh,
	  menuselect/autoconfig.h.in: Import Asterisk's autoconf magic
	  instead of using our own.

2014-07-17 21:17 +0000 [r418832-418870]  Matthew Jordan <mjordan@digium.com>

	* configs/samples/acl.conf.sample (added),
	  configs/samples/extensions.conf.sample (added),
	  configs/res_parking.conf.sample (removed),
	  configs/samples/cel_sqlite3_custom.conf.sample (added),
	  configs/cdr_sqlite3_custom.conf.sample (removed),
	  configs/modules.conf.sample (removed),
	  configs/samples/cli_aliases.conf.sample (added),
	  configs/meetme.conf.sample (removed),
	  configs/cdr_pgsql.conf.sample (removed),
	  configs/samples/extensions.ael.sample (added),
	  configs/samples/cdr_adaptive_odbc.conf.sample (added),
	  configs/samples/motif.conf.sample (added),
	  configs/samples/extensions_minivm.conf.sample (added),
	  configs/samples/res_curl.conf.sample (added),
	  configs/res_config_sqlite3.conf.sample (removed),
	  configs/mgcp.conf.sample (removed), configs/dsp.conf.sample
	  (removed), configs/udptl.conf.sample (removed),
	  configs/sip.conf.sample (removed), configs/dbsep.conf.sample
	  (removed), configs/queuerules.conf.sample (removed),
	  configs/samples/cdr_mysql.conf.sample (added),
	  configs/confbridge.conf.sample (removed),
	  configs/samples/cdr_odbc.conf.sample (added),
	  configs/samples/minivm.conf.sample (added),
	  configs/enum.conf.sample (removed),
	  configs/samples/codecs.conf.sample (added),
	  configs/samples/chan_dahdi.conf.sample (added),
	  configs/samples/cdr_custom.conf.sample (added),
	  configs/samples/res_config_mysql.conf.sample (added),
	  configs/samples/dundi.conf.sample (added),
	  configs/samples/oss.conf.sample (added),
	  configs/samples/app_mysql.conf.sample (added),
	  configs/samples/queues.conf.sample (added),
	  configs/samples/cdr.conf.sample (added),
	  configs/samples/cdr_syslog.conf.sample (added),
	  configs/festival.conf.sample (removed),
	  configs/samples/cel_pgsql.conf.sample (added),
	  configs/http.conf.sample (removed), configs/phoneprov.conf.sample
	  (removed), configs/alarmreceiver.conf.sample (removed),
	  configs/samples/features.conf.sample (added),
	  configs/cdr_tds.conf.sample (removed),
	  configs/func_odbc.conf.sample (removed),
	  configs/samples/logger.conf.sample (added),
	  configs/samples/res_odbc.conf.sample (added),
	  configs/samples/agents.conf.sample (added),
	  configs/res_fax.conf.sample (removed),
	  configs/samples/xmpp.conf.sample (added),
	  configs/iaxprov.conf.sample (removed),
	  configs/res_pgsql.conf.sample (removed),
	  configs/extensions.conf.sample (removed),
	  configs/chan_mobile.conf.sample (removed), configs/asterisk.adsi
	  (removed), configs/cel_sqlite3_custom.conf.sample (removed),
	  configs/users.conf.sample (removed),
	  configs/samples/res_pktccops.conf.sample (added),
	  configs/samples/amd.conf.sample (added), configs/rtp.conf.sample
	  (removed), configs/samples/res_parking.conf.sample (added),
	  configs/hep.conf.sample (removed),
	  configs/samples/modules.conf.sample (added),
	  configs/cel_tds.conf.sample (removed),
	  configs/res_curl.conf.sample (removed),
	  configs/samples/skinny.conf.sample (added),
	  configs/samples/cdr_pgsql.conf.sample (added),
	  configs/samples/sip_notify.conf.sample (added),
	  configs/samples/test_sorcery.conf.sample (added),
	  configs/samples/dsp.conf.sample (added),
	  configs/ss7.timers.sample (removed),
	  configs/samples/udptl.conf.sample (added),
	  configs/cdr_odbc.conf.sample (removed),
	  configs/samples/sip.conf.sample (added),
	  configs/minivm.conf.sample (removed),
	  configs/res_config_sqlite.conf.sample (removed),
	  configs/codecs.conf.sample (removed), configs/osp.conf.sample
	  (removed), configs/samples/cel_custom.conf.sample (added),
	  configs/samples/dbsep.conf.sample (added),
	  configs/samples/app_skel.conf.sample (added),
	  configs/console.conf.sample (removed),
	  configs/cdr_manager.conf.sample (removed),
	  configs/cdr_custom.conf.sample (removed),
	  configs/chan_dahdi.conf.sample (removed),
	  configs/res_config_mysql.conf.sample (removed),
	  configs/samples/statsd.conf.sample (added),
	  configs/cli.conf.sample (removed), configs/queues.conf.sample
	  (removed), configs/cdr_syslog.conf.sample (removed), UPGRADE.txt,
	  configs/manager.conf.sample (removed),
	  configs/samples/res_corosync.conf.sample (added),
	  configs/features.conf.sample (removed), configs/sla.conf.sample
	  (removed), configs/logger.conf.sample (removed),
	  configs/res_odbc.conf.sample (removed),
	  configs/agents.conf.sample (removed),
	  configs/samples/ooh323.conf.sample (added), Makefile,
	  configs/xmpp.conf.sample (removed),
	  configs/samples/phoneprov.conf.sample (added),
	  configs/samples/alarmreceiver.conf.sample (added),
	  configs/samples/cdr_tds.conf.sample (added),
	  configs/extconfig.conf.sample (removed),
	  configs/samples/func_odbc.conf.sample (added),
	  configs/samples/res_fax.conf.sample (added),
	  configs/samples/iaxprov.conf.sample (added),
	  configs/samples/res_ldap.conf.sample (added),
	  configs/samples/dnsmgr.conf.sample (added),
	  configs/res_pktccops.conf.sample (removed),
	  configs/cel.conf.sample (removed),
	  configs/samples/res_pgsql.conf.sample (added),
	  configs/samples/chan_mobile.conf.sample (added),
	  configs/samples/asterisk.adsi (added),
	  configs/samples/users.conf.sample (added),
	  configs/samples/rtp.conf.sample (added),
	  configs/phone.conf.sample (removed), configs/skinny.conf.sample
	  (removed), configs/muted.conf.sample (removed),
	  configs/samples/hep.conf.sample (added), configs/iax.conf.sample
	  (removed), configs/samples/cel_tds.conf.sample (added),
	  configs/sip_notify.conf.sample (removed),
	  configs/samples/telcordia-1.adsi (added),
	  configs/samples/alsa.conf.sample (added),
	  configs/samples/adsi.conf.sample (added),
	  configs/test_sorcery.conf.sample (removed),
	  configs/samples/followme.conf.sample (added),
	  configs/samples/asterisk.conf.sample (added),
	  configs/extensions.lua.sample (removed), configs/say.conf.sample
	  (removed), configs/cel_custom.conf.sample (removed),
	  configs/samples/ss7.timers.sample (added),
	  configs/samples/cel_odbc.conf.sample (added),
	  configs/app_skel.conf.sample (removed),
	  configs/samples/ccss.conf.sample (added),
	  configs/cli_permissions.conf.sample (removed),
	  configs/statsd.conf.sample (removed),
	  configs/samples/res_config_sqlite.conf.sample (added),
	  configs/config_test.conf.sample (removed),
	  configs/indications.conf.sample (removed),
	  configs/samples/osp.conf.sample (added),
	  configs/samples/cdr_manager.conf.sample (added),
	  configs/samples/console.conf.sample (added),
	  configs/voicemail.conf.sample (removed),
	  configs/res_corosync.conf.sample (removed),
	  configs/misdn.conf.sample (removed),
	  configs/samples/cli.conf.sample (added), configs/ari.conf.sample
	  (removed), configs/ooh323.conf.sample (removed),
	  configs/samples/calendar.conf.sample (added),
	  configs/samples/res_stun_monitor.conf.sample (added),
	  configs/samples/manager.conf.sample (added),
	  configs/samples/pjsip_notify.conf.sample (added),
	  configs/samples/sla.conf.sample (added),
	  configs/musiconhold.conf.sample (removed),
	  configs/pjsip.conf.sample (removed), configs/sorcery.conf.sample
	  (removed), configs/vpb.conf.sample (removed),
	  configs/unistim.conf.sample (removed),
	  configs/res_ldap.conf.sample (removed),
	  configs/dnsmgr.conf.sample (removed),
	  configs/samples/extconfig.conf.sample (added),
	  configs/samples/res_snmp.conf.sample (added),
	  configs/acl.conf.sample (removed),
	  configs/samples/smdi.conf.sample (added),
	  configs/samples/cel.conf.sample (added),
	  configs/cli_aliases.conf.sample (removed),
	  configs/samples/cdr_sqlite3_custom.conf.sample (added),
	  configs/extensions.ael.sample (removed),
	  configs/cdr_adaptive_odbc.conf.sample (removed),
	  configs/samples/phone.conf.sample (added),
	  configs/extensions_minivm.conf.sample (removed),
	  configs/motif.conf.sample (removed), configs/telcordia-1.adsi
	  (removed), configs/samples/meetme.conf.sample (added),
	  configs/adsi.conf.sample (removed), configs/alsa.conf.sample
	  (removed), configs/samples/muted.conf.sample (added),
	  configs/followme.conf.sample (removed),
	  configs/asterisk.conf.sample (removed),
	  configs/samples/iax.conf.sample (added),
	  configs/samples/res_config_sqlite3.conf.sample (added),
	  configs/samples/mgcp.conf.sample (added),
	  configs/cel_odbc.conf.sample (removed), configs/ccss.conf.sample
	  (removed), configs/cdr_mysql.conf.sample (removed),
	  configs/samples/extensions.lua.sample (added),
	  configs/samples/say.conf.sample (added),
	  configs/dundi.conf.sample (removed),
	  configs/samples/queuerules.conf.sample (added),
	  configs/oss.conf.sample (removed), configs/app_mysql.conf.sample
	  (removed), configs/samples/confbridge.conf.sample (added),
	  configs/samples/cli_permissions.conf.sample (added),
	  configs/samples/enum.conf.sample (added),
	  configs/samples/config_test.conf.sample (added),
	  configs/cdr.conf.sample (removed),
	  configs/samples/indications.conf.sample (added),
	  configs/cel_pgsql.conf.sample (removed),
	  configs/res_stun_monitor.conf.sample (removed),
	  configs/calendar.conf.sample (removed),
	  configs/samples/voicemail.conf.sample (added),
	  configs/pjsip_notify.conf.sample (removed),
	  configs/samples/misdn.conf.sample (added),
	  configs/samples/ari.conf.sample (added),
	  configs/samples/festival.conf.sample (added),
	  configs/samples/http.conf.sample (added),
	  configs/res_snmp.conf.sample (removed),
	  configs/samples/musiconhold.conf.sample (added),
	  configs/samples/pjsip.conf.sample (added),
	  configs/samples/sorcery.conf.sample (added),
	  configs/samples/vpb.conf.sample (added), configs/smdi.conf.sample
	  (removed), configs/samples/unistim.conf.sample (added),
	  configs/samples (added), configs/amd.conf.sample (removed):
	  configs: Move sample config files into a subdirectory of configs
	  This moves all samples configs from configs/ to configs/samples.
	  This allows for additional sets of sample configuration files to
	  be added in the future. Review:
	  https://reviewboard.asterisk.org/r/3804/

	* channels/chan_sip.c, UPGRADE.txt: chan_sip: Make
	  progressinband=never really mean 'never' progressinband=never in
	  sip.conf is easily defeated if an onward trunk sends a progress
	  indication of its own. This is almost certain to happen if the
	  onward trunk is ISDN or IAX as these technologies send a progress
	  indication even if early media is not required. This progress
	  message is passed to the caller, and causes the "never" option to
	  be rather badly named. This patch changes the behaviour of this
	  setting in the following ways: 1) In sip_write(), do not pass the
	  media unless we have either progressed beyond INV_EARLY_MEDIA, or
	  we are in INV_EARLY_MEDIA state, and early media is both set-up
	  and wanted. This helps resolve double-ringing on some buggy
	  handsets. 2) In sip_indicate(), if we see AST_CONTROL_PROGRESS,
	  but SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to
	  avoid implicitly enabling early media. Avoid sending double ring
	  indications. NOTE: the meaning of the SIP_PROGRESS_SENT flag
	  changes slightly in this patch to also encapsulate the fact that
	  a channel has *sent or received* a 183 Progress indication. This
	  makes the updated code in sip_write() much more simple. Review:
	  https://reviewboard.asterisk.org/r/3700 ASTERISK-23972 #close
	  Reported by: Steve Davies patches:
	  inband_never_present_early_media2 uploaded by Steve Davies
	  (License 5012)

	* menuselect: Add svn:ignore property

	* UPGRADE.txt, menuselect/configure, menuselect/configure.ac,
	  configure, configure.ac: configure: Fix libxml2 development
	  library dependency checking The commit that added libxml2 support
	  didn't fully check for the libxml2 development script in the
	  Asterisk configure file. As a result, Asterisk could be
	  configured, then fail on menuselect. This patch fixes it so that
	  Asterisk should detect the libxml2 dependency failure first.

	* menuselect/makeopts.in, menuselect/autoconfig.h.in,
	  menuselect/menuselect.h, menuselect/example_menuselect-tree,
	  configure, include/asterisk/autoconfig.h.in, menuselect/Makefile,
	  menuselect/README, menuselect/aclocal.m4, configure.ac,
	  UPGRADE.txt, menuselect/configure, menuselect/configure.ac,
	  menuselect/menuselect.c, menuselect/acinclude.m4: menuselect: Add
	  libxml2 support (Patch 3) This is the final patch in adding
	  menuselect to Asterisk. - The first patch (r418832) added
	  menuselect along with mxml - The second patch (r418833) removed
	  mxml from menuselect This patch adds support for libxml2 to
	  menuselect, and makes libxml2 a required library for Asterisk.
	  Note that the libxml2 portion of this patch was written by Sean
	  Bright, and was made available on a team branch:
	  http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/
	  Review: https://reviewboard.asterisk.org/r/3773/ ASTERISK-20703
	  #close patches: some_mysterious_team_branch uploaded by
	  seanbright (License 5060)

	* menuselect/mxml (removed): menuselect: Remove mxml from
	  menuselect (Patch 2) This is the second patch that adds
	  menuselect to Asterisk trunk. The previous commit (r418832) added
	  menuselect along with mxml; this patch removes mxml completely
	  from Menuselect. A subsequent patch will switch menuselect over
	  to using libxml2, and make libxml2 a required dependency for
	  Asterisk. ASTERISK-20703

	* menuselect/mxml/configure.in (added), menuselect/acinclude.m4
	  (added), menuselect/mxml/mxml.list.in (added),
	  menuselect/mxml/README (added), menuselect/linkedlists.h (added),
	  menuselect/mxml (added), menuselect/mxml/config.h.in (added),
	  menuselect/aclocal.m4 (added), menuselect/install-sh (added),
	  menuselect/mxml/mxml-string.c (added),
	  menuselect/menuselect_stub.c (added), menuselect/make_version
	  (added), menuselect/mxml/mxml-entity.c (added),
	  menuselect/bootstrap.sh (added), menuselect/makeopts.in (added),
	  menuselect/autoconfig.h.in (added), menuselect/config.guess
	  (added), menuselect/mxml/install-sh (added),
	  menuselect/test/build_tools/menuselect-deps (added), /,
	  menuselect/contrib/menuselect-dummy (added),
	  menuselect/config.sub (added), menuselect/mxml/configure (added),
	  menuselect/mxml/Makefile.in (added), menuselect (added),
	  menuselect/contrib (added), menuselect/mxml/mxml.pc.in (added),
	  menuselect/configure.ac (added), menuselect/mxml/mxml-set.c
	  (added), menuselect/contrib/Makefile-dummy (added),
	  menuselect/mxml/ANNOUNCEMENT (added), menuselect/missing (added),
	  menuselect/menuselect_curses.c (added),
	  menuselect/example_menuselect-tree (added), menuselect/Makefile
	  (added), menuselect/mxml/mxml-search.c (added), menuselect/test
	  (added), menuselect/test/menuselect-tree (added),
	  menuselect/mxml/mxml.h (added), menuselect/mxml/mxml-index.c
	  (added), menuselect/configure (added),
	  menuselect/menuselect_newt.c (added), menuselect/mxml/mxml-attr.c
	  (added), menuselect/mxml/mxml-private.c (added),
	  menuselect/menuselect.c (added), menuselect/mxml/CHANGES (added),
	  menuselect/mxml/COPYING (added), menuselect/mxml/mxml-file.c
	  (added), menuselect/menuselect.h (added),
	  menuselect/menuselect_gtk.c (added), menuselect/README (added),
	  menuselect/strcompat.c (added), menuselect/mxml/mxml-node.c
	  (added), menuselect/test/build_tools (added): menuselect: Add
	  menuselect to Asterisk trunk (Patch 1) This is the first patch
	  that adds menuselect to Asterisk trunk, and removes the
	  svn:externals property. This is being done for two reasons: (1)
	  The removal of external repositories eases a future migration to
	  git (2) Asterisk is now the only thing that uses menuselect; as a
	  result, there's little need to keep it in an external repository
	  Subsequent patches will remove the mxml dependency from
	  menuselect and tidy up the build system. ASTERISK-20703

2014-07-17 14:28 +0000 [r418811]  Kinsey Moore <kmoore@digium.com>

	* /, main/bridge_channel.c: TEST_FRAMEWORK: Fix threewaytransfer
	  reporting Ensure that three-way transfers can be reported even if
	  featuremap is non-NULL. ........ Merged revisions 418810 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-16 23:08 +0000 [r418788]  Corey Farrell <git@cfware.com>

	* /, channels/dahdi/bridge_native_dahdi.c: Remove include of
	  astobj.h from channels/dahdi/bridge_native_dahdi.c. The include
	  was unneeded, this is split off from r3758 as it applies to 12.
	  ........ Merged revisions 418787 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-16 14:03 +0000 [r418717-418757]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip/pjsip_configuration.c, CHANGES, res/res_pjsip.c,
	  channels/chan_pjsip.c, include/asterisk/res_pjsip.h,
	  contrib/ast-db-manage/config/versions/1d50859ed02e_create_accountcode.py
	  (added), /, configs/pjsip.conf.sample: res_pjsip: Support setting
	  a default accountcode on endpoints Most channel drivers let you
	  specify a default accountcode to be set on channels associated
	  with a particular peer/endpoint/object. Prior to this patch,
	  chan_pjsip/res_pjsip did not support such a setting. This patch
	  adds a new setting to the res_pjsip endpoint object,
	  'accountcode'. When a channel is created that is associated with
	  an endpoint with this value set, the channel will automatically
	  have its accountcode property set to the value configured for the
	  endpoint. Review: https://reviewboard.asterisk.org/r/3724/
	  ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged
	  revisions 418756 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* cdr/cdr_pgsql.c, CHANGES, configs/cdr_pgsql.conf.sample,
	  configs/res_pgsql.conf.sample, cel/cel_pgsql.c,
	  res/res_config_pgsql.c, configs/cel_pgsql.conf.sample: cel_pgsql,
	  cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name
	  support This patch adds support for the PostgreSQL
	  application_name connection setting. When the appropriate
	  PostgreSQL module's configuration is set with an application
	  name, the name will be passed to PostgreSQL on connection and
	  displayed in the database's pg_stat_activity view, as well as in
	  CSV logs. This aids in managing which applications/servers are
	  connected to a PostgreSQL database, as well as tracing the
	  activity of those connections. Review:
	  https://reviewboard.asterisk.org/r/3591 ASTERISK-23737 #close
	  Reported by: Gergely Domodi patches: pgsql_application_name.patch
	  uploaded by Gergely Domodi (License 6610)

	* codecs/codec_adpcm.c, main/format.c: codec_adpcm: Change
	  description of codec "ADPCM" to "Dialogic ADPCM" Technically,
	  ADPCM is a method that can be applied to several codecs.
	  Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec. See
	  http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information
	  about said codec. Review: https://reviewboard.asterisk.org/r/3744
	  patches: rb3744.patch uploaded by dennis.guse (License 6513)

	* UPGRADE.txt, main/manager.c, /: manager: Return ActionID on
	  nominal responses to PresenceState action When the PresenceState
	  action is executed, the nominal path fails to include the
	  ActionID in the successful response. This patch adds a call to
	  astman_start_ack, which guarantees that an ActionID (if provided)
	  will be sent back to the AMI client. Unlike the Asterisk 11 and
	  12 patches, this patch also deprecates the duplicate Message key
	  in the response to the action, replacing it with the key
	  'PresenceMessage'. Review:
	  https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close
	  ........ Merged revisions 418713 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 418714 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-15 23:03 +0000 [r418716]  Kinsey Moore <kmoore@digium.com>

	* /, main/bridge_channel.c: TEST_FRAMEWORK: Fix ref leak in feature
	  activation This fixes two reference leaks that would occur when
	  TEST_FRAMEWORK was enabled and features were successfully
	  executed. ........ Merged revisions 418715 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-15 17:57 +0000 [r418654]  Jonathan Rose <jrose@digium.com>

	* funcs/func_uri.c, /: func_uri: URIENCODE/URIDECODE - allow empty
	  strings as argument Previously these two dialplan functions would
	  issue warnings and return failure when an empty string is used as
	  the argument. Now they will not issue a warning and will
	  successfully return an empty string. ASTERISK-23911 #close
	  Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3745/ ........ Merged
	  revisions 418641 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 418649 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 418650 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-15 12:11 +0000 [r418616]  Sean Bright <sean@malleable.com>

	* main/asterisk.c: Update Asterisk copyright year in
	  main/asterisk.c It's been 2014 for like... 6 months.

2014-07-14 14:55 +0000 [r418566-418587]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/logger.h, /: logger.h: Extract DEBUG_ATLEAST()
	  to complement VERBOSITY_ATLEAST(). ........ Merged revisions
	  418586 from http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/jabber.h (removed), include/asterisk/jingle.h
	  (removed), include/asterisk/frame_defs.h (removed),
	  configs/h323.conf.sample (removed): Actually delete the removed
	  files.

2014-07-13 21:57 +0000 [r418507]  Corey Farrell <git@cfware.com>

	* /, main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work
	  around REF_DEBUG race which causes out of order log entries *
	  Update refcounter.py to use delta's to track the current
	  reference count. * Use result from internal_ao2_ref to write
	  old_refcount to refs_log. Review:
	  https://reviewboard.asterisk.org/r/3756/ ........ Merged
	  revisions 418504 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 418505 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 418506 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-13 20:08 +0000 [r418488]  Scott Griepentrog <sgriepentrog@digium.com>

	* include/asterisk/astobj2.h: astobj2: correct define for
	  ao2_t_cleanup This change maps the ao2_t_cleanup() function to
	  the correct debug function so that it can be used. Review:
	  https://reviewboard.asterisk.org/r/3764/

2014-07-13 16:48 +0000 [r418448-418467]  Corey Farrell <git@cfware.com>

	* main/manager.c, /, apps/app_skel.c: Fix minor reference leaks in
	  app_skel and TEST_FRAMEWORK * Cleanup games object in app_skel. *
	  Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+).
	  Review: https://reviewboard.asterisk.org/r/3757/ ........ Merged
	  revisions 418465 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 418466 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/jabber.h, include/asterisk/jingle.h,
	  configs/h323.conf.sample: Remove files left behind on removal of
	  h323, jingle and jabber. This change removes h323.conf.sample,
	  jingle.h, jabber.h left behind by r3698. Review:
	  https://reviewboard.asterisk.org/r/3755/

2014-07-11 23:00 +0000 [r418419]  Matthew Jordan <mjordan@digium.com>

	* main/astobj2.c, include/asterisk/astobj2.h: astobj2: Add tag
	  variants for ao2_bump, ao2_cleanup, and ao2_replace Tags are
	  useful in hunting down ref imbalances; this patch adds tag
	  variants for these commonly used macros/functions. Review:
	  https://reviewboard.asterisk.org/r/3750/

2014-07-11 21:10 +0000 [r418397]  Corey Farrell <git@cfware.com>

	* /, include/asterisk/astobj2.h: astobj2: tweak ao2_replace to do
	  nothing when it would be a NoOp This change causes ao2_replace to
	  do nothing when src == dst. This avoids REF_DEBUG logging when
	  we're not actually doing anything. Review:
	  https://reviewboard.asterisk.org/r/3743/ ........ Merged
	  revisions 418396 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-11 16:42 +0000 [r418370]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, main/config.c: config: inform config hook of change when
	  writing file When updated configuration is written back to the
	  conf file - for example when a user changes their voicemail pin,
	  make sure that any config hook that wants to know of changes is
	  informed. Review: https://reviewboard.asterisk.org/r/3708/
	  ........ Merged revisions 418366 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 418369 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-10 15:36 +0000 [r418325]  Matthew Jordan <mjordan@digium.com>

	* /, include/asterisk/xmpp.h: include/asterisk/xmpp.h: Convert
	  indentation to tabs This is a whitespace only change. ........
	  Merged revisions 418323 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 418324 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-10 01:59 +0000 [r418226-418264]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, /: chan_dahdi/sig_pri: Fix type mismatch in
	  the idledial feature's channel creation. Square pegs in round
	  holes don't work very well. ........ Merged revisions 418261 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 418262 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 418263 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/stasis/stasis_bridge.h (added), main/bridge_channel.c,
	  res/res_stasis.c, /, res/stasis/stasis_bridge.c (added),
	  include/asterisk/bridge_channel.h, main/bridge_basic.c: ARI: Make
	  mixing bridges propagate linkedids and accountcodes. * Create a
	  Stasis bridge sub-class to propagate linkedids and accountcodes.
	  * Fixed the basic bridge sub-class to update peeraccount codes
	  when the number of channels in the bridge drops back down to two
	  parties. * Refactored ast_bridge_channel_update_accountcodes() to
	  handle channels joining/leaving the bridge. * Fixed the basic
	  bridge sub-class to not call the base bridge class pull method
	  twice. AFS-105 #close ASTERISK-23852 #close Reported by: Richard
	  Mudgett Review: https://reviewboard.asterisk.org/r/3720/ ........
	  Merged revisions 418225 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-08 14:48 +0000 [r418174-418183]  Matthew Jordan <mjordan@digium.com>

	* rest-api/api-docs/deviceStates.json,
	  rest-api/api-docs/endpoints.json,
	  rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
	  /, rest-api/api-docs/asterisk.json,
	  rest-api/api-docs/applications.json,
	  rest-api/api-docs/playbacks.json,
	  rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
	  rest-api/resources.json, include/asterisk/manager.h,
	  rest-api/api-docs/bridges.json,
	  rest-api/api-docs/recordings.json: manager/ARI: Update version to
	  2.4.0/1.4.0; Update UPGRADE.txt ........ Merged revisions 418182
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix undefined
	  function when PJPROJECT is not installed The
	  dtls_perform_handshake function was mistakenly placed under the
	  guards for USE_PJPROJECT. If PJPROJECT was not installed, the
	  function would not be defined, while other functions would
	  attempt to still use it. This prevented res_rtp_asterisk from
	  being loaded. ASTERISK-24001 #close Reported by: Don Fanning
	  ........ Merged revisions 418172 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-07 16:08 +0000 [r418117]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/res_pjsip_body_generator_types.h,
	  res/res_pjsip_dialog_info_body_generator.c (added),
	  res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c, /,
	  include/asterisk/res_pjsip_presence_xml.h:
	  res_pjsip_dialog_info_body_generator: Add dialog-info+xml support
	  for presence. This module implements dialog-info+xml for the
	  purposes of presence. This means that phones such as Grandstreams
	  can now subscribe to receive presence information for an
	  extension. ASTERISK-21443 #close Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3705/ ........ Merged
	  revisions 418116 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-07 02:15 +0000 [r418090]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/stasis_app.h, res/ari/resource_channels.c,
	  res/res_stasis.c, /, res/stasis/app.c: ARI/res_stasis: Subscribe
	  to both Local channel halves when originating to app This patch
	  fixes two bugs: 1. When originating a channel into a Stasis
	  application, we already create a subscription for the channel
	  that is going into our Stasis app. Unfortunately, when you create
	  a Local channel and pass it off to a Stasis app, you really
	  aren't creating just one channel: you're creating two. This patch
	  snags the second half of the Local channel pair (assuming it is a
	  Local channel pair, but luckily core_local is kind about such
	  assumptions) and subscribes to it as well. 2. Subscriptions are a
	  bit sticky right now. If a subscription is made, the 'interest'
	  count gets bumped on the Stasis subscription - but unless
	  something explicitly unsubscribes the channel, said subscription
	  sticks around. This is not much of a problem is a user is
	  creating the subscription - if they made it, they must want it.
	  However, when we are creating implicit subscriptions, we need to
	  make sure something clears them out. This patch takes a
	  pessimistic approach: it watches the cache updates coming from
	  Stasis and, if we notice that the cache just cleared out an
	  object, we delete our subscription object. This keeps our ao2
	  container of Stasis forwards in an application from growing out
	  of hand; it also is a bit more forgiving for end users who may
	  not realize they were supposed to unsubscribe from that channel
	  that just hung up. Review:
	  https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close
	  ........ Merged revisions 418089 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-07 01:22 +0000 [r418067-418084]  Kinsey Moore <kmoore@digium.com>

	* tests/test_cel.c, main/cel.c, channels/chan_pjsip.c,
	  res/res_pjsip_session.c, /: CEL: Fix incorrect/missing extra
	  field information This corrects two issues with the extra field
	  information in Asterisk 12+ in channel event logs. It is possible
	  to inject custom values into the dialstatus provided by
	  ast_channel_dial_type() Stasis messages that fall outside the
	  enumeration allowed for the DIALSTATUS channel variable. CEL now
	  filters for the allowed values and ignores other values. The
	  "hangupsource" extra field key is always blank if the far end
	  channel is a chan_pjsip channel. This is because the hangupsource
	  is never set for the pjsip channel driver. This change sets the
	  hangupsource whenever a hangup is queued for chan_pjsip channels.
	  This corrects an issue with the pjsip channel driver where the
	  hangupcause information was not being set properly. Review:
	  https://reviewboard.asterisk.org/r/3690/ ........ Merged
	  revisions 418071 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/http.c: HTTP: Fix build for gcc 4.10 ........ Merged
	  revisions 418066 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-04 15:26 +0000 [r418019-418050]  Matthew Jordan <mjordan@digium.com>

	* main/Makefile: main/Makefile: fix compilation error of buildinfo
	  occurring on 'make install' Egads. Another bad deletion of too
	  much when attempting to remove h323 stuff.

	* configure.ac, build_tools/menuselect-deps.in, configure,
	  main/Makefile: configure: Remove last vestiges of h323; DO create
	  menuselect-deps The previous patch (r418034) fixed the 'glitch'
	  that the channels/h323 Makefile no longer existed. Unfortunately,
	  removing the entire line was a bit of a blunder, as it meant that
	  build_tools/menuselect-deps was never generated. Hilarity ensued
	  when actually trying to compile. But hey! At least configure
	  worked. This patch fixes *that* glitch, and removes some more of
	  the vestiges of h323. (It had tendrils in the main Makefile?
	  Crazy.)

	* configure.ac, configure: configure: Update script to pass if
	  channels/h323/Makefile.in does not exist This simply removes that
	  check from the configure script, as r418019 removed chan_h323.

	* apps/app_dahdibarge.c (removed), configs/gtalk.conf.sample
	  (removed), main/pbx.c, apps/app_readfile.c (removed),
	  channels/chan_sip.c, configs/jingle.conf.sample (removed),
	  UPGRADE.txt, res/res_musiconhold.c, channels/chan_gtalk.c
	  (removed), channels/Makefile, CHANGES, res/res_jabber.c
	  (removed), channels/h323 (removed), utils/conf2ael.c,
	  channels/chan_jingle.c (removed), res/ael/pval.c,
	  configs/jabber.conf.sample (removed),
	  configs/asterisk.conf.sample, res/res_agi.c, channels/chan_h323.c
	  (removed), addons/Makefile, pbx/pbx_realtime.c, utils/ael_main.c,
	  include/asterisk/options.h, main/asterisk.c,
	  addons/app_saycountpl.c (removed): Remove many deprecated modules
	  Billing records are fair, To get paid is quite bright, You should
	  really use ODBC; Good-bye cdr_sqlite. Microsoft did once push
	  H.323, Hell, we all remember NetMeeting. But try to compile
	  chan_h323 now And you will take quite a beating. The XMPP and SIP
	  war was fierce, And in the distant fray Was birthed
	  res_jabber/chan_jingle; But neither to stay. For everyone did
	  care and chase what Google professed. "Free Internet Calling" was
	  what devotees cried, But Google did change the specs so often
	  That the developers were happy the day chan_gtalk died. And then
	  there was that odd application Dedicated to the Polish tongue.
	  app_saycountpl was subsumed by Say; One could say its bell was
	  rung. To read and parse a file from the dialplan You could (I
	  guess) use an application. app_readfile did fill that purpose,
	  but I think A function is perhaps better in its creation. Barging
	  is rude, I'm not sure why we do it. Inwardly, the caller will
	  probably sigh. But if you really must do it, Don't use
	  app_dahdibarge, use ChanSpy. We all despise the sound of tinny
	  robots It makes our queues so cold. To control such an
	  abomination It's better to not use Wait/SetMusicOnHold. It's
	  often nice to know properties of a channel It makes our calls
	  right We have a nice function called CHANNEL And so SIPCHANINFO
	  is sent off into the night. And now things get odd; Apparently
	  one could delimit with a colon Properties from the SIPPEER
	  function! Commas are in; all others are done. Finally, a word on
	  pipes and commas. We're sorry. We can't say it enough. But those
	  compatibility options in asterisk.conf; To maintain them forever
	  was just too tough. This patch removes: * cdr_sqlite * chan_gtalk
	  * chan_jingle * chan_h323 * res_jabber * app_saycountpl *
	  app_readfile * app_dahdibarge It removes the following
	  applications/functions: * WaitMusicOnHold * SetMusicOnHold *
	  SIPCHANINFO It removes the colon delimiter from the SIPPEER
	  function. Finally, it also removes all compatibility options that
	  were configurable from asterisk.conf, as these all applied to
	  compatibility with Asterisk 1.4 systems. Review:
	  https://reviewboard.asterisk.org/r/3698/

2014-07-03 22:22 +0000 [r417933-417976]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, /, UPGRADE.txt,
	  channels/sig_pri.c: chan_dahdi: Add inband_on_setup_ack
	  compatibility option. The new inband_on_setup_ack option causes
	  Asterisk to assume inband audio may be present when a
	  SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says
	  that in scenarios with overlap dialing, when a dialtone is sent
	  from the network side, progress indicator 8 "Inband info now
	  available" MAY be sent to the CPE if no digits were received with
	  the SETUP. It is thus implied that the ie is mandatory if digits
	  came with the SETUP and dialtone is needed. This option should be
	  enabled, when the network sends dialtone and you want to hear it,
	  but the network doesn't send the progress indicator when needed.
	  NOTE: For Q.SIG setups this option should be enabled when
	  outgoing overlap dialing is also enabled because Q.SIG does not
	  send the progress indicator with the SETUP ACK. The commit
	  -r413714 (AST-1338) which causes this issue was dealing with a
	  SIP-to-ISDN interoperability issue. This commit is a merge of the
	  two patches indicated below. ASTERISK-23897 #close Reported by:
	  Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded
	  by Pavel Troller jira_asterisk_23897_v11.patch (license #5621)
	  patch uploaded by rmudgett Review:
	  https://reviewboard.asterisk.org/r/3633/ ........ Merged
	  revisions 417956 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 417957 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417958 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/ari/resource_channels.c, res/res_ari.c, main/manager.c, /:
	  res_ari: Fix some off-nominal paths just dropping the HTTP
	  connection. * Removed some incorrect newlines on ast_http_error()
	  messages in manager.c. * Removed an incorrect newline in
	  res_ari_channels.c. Addendum to ASTERISK-23552 ........ Merged
	  revisions 417932 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-03 17:34 +0000 [r417910-417916]  Jonathan Rose <jrose@digium.com>

	* CHANGES, channels/chan_dahdi.c: chan_dahdi: Add AMI commands for
	  controlling PRI debugging output Adds the following AMI commands:
	  PRIDebugSet - Set PRI debug levels for a specific span
	  PRIDebugFileSet - Set the file used for PRI debug message output
	  PRIDebugFileUnset - Disables file output for PRI debug messages
	  Review: https://reviewboard.asterisk.org/r/3681/

	* CHANGES, pbx/pbx_config.c, main/pbx.c: pbx_config: Add manager
	  actions to add/remove extensions Adds two new manager commands to
	  pbx_config - DialplanExtensionAdd and DialplanExtensionRemove
	  which allow manager users to create and delete extensions
	  respectively. Review: https://reviewboard.asterisk.org/r/3650/

2014-07-03 17:16 +0000 [r417901]  Richard Mudgett <rmudgett@digium.com>

	* res/res_phoneprov.c, main/http.c, UPGRADE.txt,
	  include/asterisk/tcptls.h, res/res_http_post.c,
	  res/res_http_websocket.c, configs/http.conf.sample,
	  include/asterisk/http.h, main/tcptls.c, res/res_ari.c,
	  main/manager.c, /: HTTP: Add persistent connection support.
	  Persistent HTTP connection support is needed due to the increased
	  usage of the Asterisk core HTTP transport and the frequency at
	  which REST API calls are going to be issued. * Add http.conf
	  session_keep_alive option to enable persistent connections. *
	  Parse and discard optional chunked body extension information and
	  trailing request headers. * Increased the maximum
	  application/json and application/x-www-form-urlencoded body size
	  allowed to 4k. The previous 1k was kind of small. * Removed a
	  couple inlined versions of ast_http_manid_from_vars() by calling
	  the function. manager.c:generic_http_callback() and
	  res_http_post.c:http_post_callback() * Add missing va_end() in
	  ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use
	  in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott
	  Griepentrog Review: https://reviewboard.asterisk.org/r/3691/
	  ........ Merged revisions 417880 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-03 16:55 +0000 [r417900]  Matthew Jordan <mjordan@digium.com>

	* main/tcptls.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: main/tcptls: Add checks for OpenSSL Elliptic Curve
	  support The patch for ASTERISK-23905 that added PFS support in
	  Asterisk depends on the elliptic curve library support being
	  present in OpenSSL. As it turns out, some versions of OpenSSL
	  don't have this library - notably the version running on our
	  build agents. This patch fixes the build by providing a configure
	  check for the specific library calls that the PFS patch relies
	  on. Review: https://reviewboard.asterisk.org/r/3709/

2014-07-03 16:14 +0000 [r417877-417879]  sgalarneau <sgalarneau@localhost>:

	* res/ari/resource_events.h, rest-api/api-docs/channels.json,
	  res/ari/resource_channels.h, rest-api/api-docs/events.json, /:
	  ARI: Improvements to body parameters documentation The variables
	  body parameter under the originate and originate with id
	  operations of the channel resource showed invalid JSON in its
	  description. The variables body parameter under the userEvent
	  operation of the event resource made no mention that the custom
	  key/value pairs should be wrapped in a variables key in order to
	  be added to the custom user event. ASTERISK-23975 #close Review:
	  https://reviewboard.asterisk.org/r/3692/ ........ Merged
	  revisions 417878 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* rest-api-templates/api.wiki.mustache,
	  rest-api-templates/swagger_model.py, /: api.wiki.mustache: Update
	  wiki template to support body parameters This patch updates the
	  api.wiki.mustache template and the swagger_model python script to
	  understand if an operation has a body parameter. If an operation
	  does have a body parameter, it will now be displayed in the
	  corresponding wiki entry. ........ Merged revisions 407389 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-03 14:08 +0000 [r417863]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* Makefile, contrib/scripts/dahdi_span_config_hook (added):
	  dahdi_span_config_hook: automatically register new dahdi channels
	  Install a hook script for DAHDI to register new spans with
	  Asterisk automatically by running: asterisk -rx 'dahdi create
	  channel FIRST LAST' Review:
	  https://reviewboard.asterisk.org/r/3157/

2014-07-03 12:10 +0000 [r417800-417803]  Matthew Jordan <mjordan@digium.com>

	* main/tcptls.c, CHANGES: main/tcptls: Add support for Perfect
	  Forward Secrecy This patch enables Perfect Forward Secrecy (PFS)
	  in Asterisk's core TLS API. Modules that wish to enable PFS
	  should consider the following: - Ephemeral ECDH (ECDHE) is
	  enabled by default. To disable it, do not specify a ECDHE cipher
	  suite in a module's configuration, for example:
	  tlscipher=AES128-SHA:DES-CBC3-SHA - Ephemeral DH (DHE) is
	  disabled by default. To enable it, add DH parameters into the
	  private key file, i.e., tlsprivatekey. For an example, see the
	  default dh2048.pem at
	  http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
	  - Because clients expect the server to prefer PFS, and because
	  OpenSSL sorts its cipher suites by bit strength, (see "openssl
	  ciphers -v DEFAULT") consider re-ordering your cipher suites in
	  the conf file. For example:
	  tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
	  will use PFS when offered by the client. Clients which do not
	  offer PFS fall-back to AES-128 (or even 3DES as recommend by RFC
	  3261). Review: https://reviewboard.asterisk.org/r/3647/
	  ASTERISK-23905 #close Reported by: Alexander Traud patches:
	  tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520)
	  tlsPFS.patch uploaded by Alexander Traud (License 6520)

	* /, main/utils.c: main/untils: Prevent potential infinite loop in
	  ast_careful_fwrite A loop in ast_careful_fwrite exists that will
	  continually attempt to write to a file stream, even in the
	  presence of EAGAIN/EINTR errors. However, if a connection that
	  uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
	  call to fflush may return EAGAIN/EINTER along with EOF. A
	  subsequent call to fflush will return EOF but not clear errno,
	  resulting in an infinite loop. This patch clears errno after it
	  is detected and handled the loop, such that any subsequent call
	  to fflush will not get erroneously stuck. Review:
	  https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close
	  Reported by: Steve Davies patches: fflush_loop_fix uploaded by
	  one47 (License 5012) ........ Merged revisions 417797 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 417798 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417799 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-02 21:13 +0000 [r417770]  Jonathan Rose <jrose@digium.com>

	* res/ari/resource_events.h, res/ari/resource_asterisk.h,
	  res/ari/resource_applications.h, res/ari/resource_playbacks.h,
	  res/ari/resource_channels.h, res/ari/resource_sounds.h, /,
	  res/ari/resource_bridges.h, res/ari/resource_recordings.h,
	  rest-api-templates/ari_resource.h.mustache,
	  res/ari/resource_device_states.h, res/ari/resource_endpoints.h,
	  res/ari/resource_mailboxes.h: ARI: Remove unnecessary \briefs
	  from automatically generated documentation Review:
	  https://reviewboard.asterisk.org/r/3440/ ........ Merged
	  revisions 412653 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-07-01 14:42 +0000 [r417679-417706]  Joshua Colp <jcolp@digium.com>

	* /, res/res_rtp_asterisk.c: res_rtp_asterisk: Don't leak memory or
	  reset state if DTLS configuration is set multiple times. ........
	  Merged revisions 417705 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_rtp_asterisk.c,
	  contrib/ast-db-manage/config/versions/51f8cb66540e_add_further_dtls_options.py
	  (added), include/asterisk/res_pjsip_session.h, main/rtp_engine.c,
	  /, channels/chan_sip.c, main/sdp_srtp.c, res/res_pjsip_sdp_rtp.c,
	  res/res_pjsip/pjsip_configuration.c, configs/sip.conf.sample,
	  include/asterisk/rtp_engine.h, res/res_pjsip.c,
	  channels/sip/include/sip.h, include/asterisk/res_pjsip.h,
	  include/asterisk/sdp_srtp.h: Recorded merge of revisions 417677
	  from http://svn.asterisk.org/svn/asterisk/branches/11 ........
	  res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS
	  negotiation on RTCP. This change fixes up DTLS support in
	  res_rtp_asterisk so it can accept and provide a SHA-256
	  fingerprint, so it occurs on RTCP, and so it occurs after ICE
	  negotiation completes. Configuration options to chan_sip and
	  chan_pjsip have also been added to allow behavior to be tweaked
	  (such as forcing the AVP type media transports in SDP).
	  ASTERISK-22961 #close Reported by: Jay Jideliov Review:
	  https://reviewboard.asterisk.org/r/3679/ Review:
	  https://reviewboard.asterisk.org/r/3686/ ........ Merged
	  revisions 417678 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-30 18:39 +0000 [r417663]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pubsub.c: Reverse logic during subscription
	  persistence recreation. In the abstraction effort, this bit of
	  logic got messed up. We want to recreate the persistence if
	  things go well, not if things fail.

2014-06-30 13:02 +0000 [r417590-417649]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: apps/app_voicemail: Fix compilation error
	  introduced in r417591 Not sure why that change to
	  ast_channel_alloc was made but ... okay.

	* apps/app_voicemail.c, main/say.c, CHANGES: app_voicemail, say:
	  Add support for Japanese Language This patch adds support for the
	  Japanese language to both the say family of applications, as well
	  as for VoiceMail and VoiceMailMain. A new pack of language sounds
	  will be released at the same time as the next major version of
	  Asterisk to support the new language features. The language
	  features can be enabled using a language code of 'ja'. Review:
	  https://reviewboard.asterisk.org/r/3477 ASTERISK-23324 #close
	  Reported by: Kevin McCoy patches:
	  app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy
	  (License 6586) say.c.20140226.jb.patch uploaded by Kevin McCoy
	  (License 6586)

	* /, channels/chan_sip.c: chan_sip: be more tolerant of whitespace
	  between attributes in SDP fmtp line This patch is essentially a
	  backport of a small portion of r397526 from ASTERISK-21981. In
	  that patch, pass through support and format attribute negotiation
	  was added for Opus. Part of that included being more tolerant to
	  whitespace in the fmtp line of an SDP; that part of the patch is
	  being applied here. As the author of the backport pointed out, in
	  SDP, the fmtp line is allowed to include whitespace between
	  attributes. RFC 3267 chapter 8.3 (from 2001) includes an example
	  for this. This was not removed in the updated RFC 4867 in 2007.
	  Review: https://reviewboard.asterisk.org/r/3658 #ASTERISK-23916
	  #close Reported by: Alexander Traud patches:
	  sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud
	  (License 6520) ........ Merged revisions 417587 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 417588 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417589 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-27 23:21 +0000 [r417571]  Richard Mudgett <rmudgett@digium.com>

	* /, main/event.c: event.c: Fix type mismatch errors in ie_maps[].
	  In v12+ the type values from the table are only used by the CEL
	  unit tests. Since the unit tests were only comparing a generated
	  expected event with a real event to see if the ie contents
	  matched and using the same table IE_PLTYPE values to read the
	  event contents, the type mismatches were not detected. ........
	  Merged revisions 417565 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-27 19:27 +0000 [r417485-417511]  Corey Farrell <git@cfware.com>

	* /, main/astobj2.c: Ensure REF_DEBUG records entrys for attempts
	  to ao2_ref an invalid object This change ensures that
	  __ao2_ref_debug writes to ref_log when given a non-NULL pointer
	  to an invalid ao2 object. This is to ensure that we record any
	  attempt manipulate references of already freed objects.
	  ASTERISK-23948 #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3677/ ........ Merged
	  revisions 417500 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 417505 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417509 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, contrib/scripts/refcounter.py: refcounter.py: prevent use of
	  excessive RAM with large refs logs When processing a 212MB refs
	  file, refcounter.py used over 3GB of RAM. This change greatly
	  reduces memory usage in two ways: * Saving object history in
	  whole lines instead of separated values. * Not saving
	  normal/skewed/leaked object lists unless they are requested.
	  ASTERISK-23921 #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3668/ ........ Merged
	  revisions 417480 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 417481 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417483 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-27 13:50 +0000 [r417461]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip/pjsip_configuration.c, res/res_pjsip_pubsub.c,
	  res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h, /,
	  res/res_pjsip_outbound_registration.c: res_pjsip: Add ActionID to
	  events created as a result of PJSIP AMI actions A number of
	  various PJSIP AMI actions were failing to parse out and place the
	  ActionID into their responses. This patch updates the various
	  PJSIP actions such that the passed in ActionID is emitted on any
	  event list complete events, as well as any intermediate events
	  created as a result of the action. #ASTERISK-23947 #close
	  Reported by: Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/3675/ ........ Merged
	  revisions 417460 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-27 02:04 +0000 [r417423-417447]  Kinsey Moore <kmoore@digium.com>

	* tests/test_cel.c: CEL: Update unit tests for bridge tech field
	  Update the CEL unit tests that handle BRIDGE_ENTER and
	  BRIDGE_EXIT events to expect the "bridge_technology" extra field
	  key.

	* CHANGES: CHANGES: Add missing changes Add missing CHANGES changes
	  from r417361 and r417383.

2014-06-26 18:27 +0000 [r417400-417421]  Matthew Jordan <mjordan@digium.com>

	* res/res_http_websocket.exports.in, /: res_http_websocket: Export
	  symbol for ast_websocket_set_timeout Thanks to Sean Bright for
	  pointing out that this was missed in #asterisk-dev. ........
	  Merged revisions 417419 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417420 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_pjsip.c, /: chan_pjsip: Add a test event for fast
	  picture updates This will drive the test on review r3419. Note
	  that the patch for this was done by Ben Ford, although it was
	  slightly modified for this commit. ASTERISK-23562 Reported by:
	  Matt Jordan ........ Merged revisions 417399 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-26 14:48 +0000 [r417361-417383]  Kinsey Moore <kmoore@digium.com>

	* main/cel.c: CEL: Add bridge tech to relevant CEL records Add the
	  "bridge_technology" extra field key to BRIDGE_ENTER and
	  BRIDGE_EXIT CEL events to convey the bridge technology in use at
	  the time the record was generated.

	* main/bridge.c, include/asterisk/channel.h,
	  include/asterisk/bridge_features.h,
	  tests/test_channel_feature_hooks.c (added),
	  main/bridge_channel.c, main/channel.c: Bridging: Allow channels
	  to define bridging hooks This patch allows the current owner of a
	  channel to define various feature hooks to be made available once
	  the channel has entered a bridge. This includes any hooks that
	  are setup on the ast_bridge_features struct such as DTMF hooks,
	  bridge event hooks (join, leave, etc.), and interval hooks.
	  Review: https://reviewboard.asterisk.org/r/3649/

2014-06-26 12:43 +0000 [r417317-417360]  Matthew Jordan <mjordan@digium.com>

	* CHANGES, apps/app_jack.c: app_jack: Support audio with a sampling
	  rate higher than 8kHz This patch enables the jack-audiohook to
	  cope with dynamic sampling rates from and to Asterisk.
	  Information from the channel is taken to derive the channel's
	  sampling rate, suiting SLINxx format and frame->datalen. There
	  are stil a few limitations after this patch: * Required
	  information is taken from the channel during initialization as
	  the audiohook does not provide this information.
	  Audiohook.internal_sampl_rate(...) is set later, but no callback
	  is available to inform app_jack. * Frame.datalen is computed
	  using "rate / 50" assuming a ptime of 20ms. There is no internal
	  API available to determine datalen for a SLINxx. * Ringbuffer
	  size is now dynamic depending on the value of frame.datalen (see
	  above) and the number of frames, which are in
	  RINGBUFFER_FRAME_CAPACITY, that need to fit. Review:
	  https://reviewboard.asterisk.org/r/3618 Note that the patch being
	  committed here is based on the patch posted on ASTERISK-23836.
	  However, Matthis Schmieder also provided a patch to enable this
	  functionality, and that patch is noted below. ASTERISK-20696
	  #close Reported by: Matthis Schmieder patches: app_jack.patch
	  uploaded by Matthis Schmieder (License 6445) ASTERISK-23836
	  #close Reported by: Dennis Guse patches: patch-app_jack.c
	  uploaded by Dennis Guse (License 6513)

	* main/udptl.c, /: udptl: Correct FEC to not consider negative
	  sequence numbers as missing When using FEC, with span=3 and
	  entries=4 Asterisk will attempt to repair the packet with
	  sequence number 5, as it will see that packet -4 is missing. The
	  result is Asterisk sending garbage packets that can kill a fax.
	  This patch adds a check to see if the sequence number is valid
	  before checking if the packet is missing. Review:
	  https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close
	  Reported by: Torrey Searle patches: udptl_fec.patch uploaded by
	  Torrey Searle (License 5334) ........ Merged revisions 417318
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 417320 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417324 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/ari/internal.h, configs/ari.conf.sample,
	  res/res_http_websocket.c, res/res_pjsip.c,
	  configs/pjsip.conf.sample, include/asterisk/http_websocket.h,
	  configs/sip.conf.sample, res/res_pjsip/config_transport.c,
	  res/ari/ari_websockets.c, res/res_pjsip_transport_websocket.c,
	  res/ari/config.c, channels/sip/include/sip.h,
	  include/asterisk/res_pjsip.h, res/res_ari.c, /,
	  channels/chan_sip.c, UPGRADE.txt: res_http_websocket: Close
	  websocket correctly and use careful fwrite When a client takes a
	  long time to process information received from Asterisk, a write
	  operation using fwrite may fail to write all information. This
	  causes the underlying file stream to be in an unknown state, such
	  that the socket must be disconnected. Unfortunately, there are
	  two problems with this in Asterisk's existing websocket code: 1.
	  Periodically, during the read loop, Asterisk must write to the
	  connected websocket to respond to pings. As such, Asterisk
	  maintains a reference to the session during the loop. When
	  ast_http_websocket_write fails, it may cause the session to
	  decrement its ref count, but this in and of itself does not break
	  the read loop. The read loop's write, on the other hand, does not
	  break the loop if it fails. This causes the socket to get in a
	  'stuck' state, preventing the client from reconnecting to the
	  server. 2. More importantly, however, is that the fwrite in
	  ast_http_websocket_write fails with a large volume of data when
	  the client takes awhile to process the information. When it does
	  fail, it fails writing only a portion of the bytes. With some
	  debugging, it was shown that this was failing in a similar
	  fashion to ASTERISK-12767. Switching this over to
	  ast_careful_fwrite with a long enough timeout solved the problem.
	  Note that this version of the patch, unlike r417310 in Asterisk
	  11, exposes configuration options beyond just chan_sip's
	  sip.conf. Configuration options to configure the write timeout
	  have also been added to pjsip.conf and ari.conf. #ASTERISK-23917
	  #close Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3624/ ........ Merged
	  revisions 417310 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417311 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-26 10:06 +0000 [r417251]  Corey Farrell <git@cfware.com>

	* /, channels/chan_sip.c: chan_sip: Fix handling of "From" headers
	  longer than 256 characters From headers were processed using a
	  256 character buffer on the stack. This change replaces that with
	  a heap allocation by ast_strdup. ASTERISK-23790 #close Reported
	  by: uniken1 Tested by: uniken1 Review:
	  https://reviewboard.asterisk.org/r/3669/ Patches:
	  chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes
	  (license 5674) ........ Merged revisions 417248 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 417249 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 417250 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-25 20:57 +0000 [r417233]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c,
	  include/asterisk/res_pjsip_pubsub.h,
	  res/res_pjsip_pidf_body_generator.c,
	  res/res_pjsip_pubsub.exports.in, res/res_pjsip_mwi.c,
	  res/res_pjsip_xpidf_body_generator.c: Abstract PJSIP-specific
	  elements from the pubsub API. This helps to pave the way for RLS
	  work that is to come. Since this is a self-contained change and
	  subscription tests still pass, this work is being committed
	  directly to trunk instead of a working branch. ASTERISK-23865
	  #close Review: https://reviewboard.asterisk.org/r/3628

2014-06-25 18:57 +0000 [r417213]  Corey Farrell <git@cfware.com>

	* main/astobj2_container.c, /: ao2_container node object ignores
	  REF_DEBUG in all places except one Almost every reference
	  operation against container node's uses __ao2_alloc or __ao2_ref,
	  thereby preventing ref logging for the nodes. One node reference
	  is released with ao2_t_ref, causing refcounter.py to falsely
	  report skews and leaks for many nodes. ASTERISK-23922 #close
	  Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3670/ ........ Merged
	  revisions 417212 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-25 00:45 +0000 [r417193]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Skinny: cleanup some log messages around
	  sessions.

2014-06-24 02:50 +0000 [r417167]  Corey Farrell <git@cfware.com>

	* include/asterisk/netsock.h, main/utils.c, main/netsock.c,
	  include/asterisk/res_pjsip_session.h: Move eid functions to
	  utils.c, mark netsock.h deprecated Move eid functions from
	  netsock.c to utils.c. These functions were already published by
	  utils.h. Flag netsock.h as deprecated and switch
	  res_pjsip_session.h to use netsock2.h. The only code that still
	  uses netsock.h is chan_iax2. ASTERISK-23920 #close Reported by:
	  Corey Farrell Review: https://reviewboard.asterisk.org/r/3661/

2014-06-23 18:50 +0000 [r417143]  Joshua Colp <jcolp@digium.com>

	* /, res/res_rtp_asterisk.c: res_rtp_asterisk: Return the length of
	  data written when sending via ICE instead of 0. ASTERISK-23834
	  #close Reported by: Richard Kenner ........ Merged revisions
	  417141 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........ Merged revisions 417142 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-23 16:04 +0000 [r417120]  Richard Mudgett <rmudgett@digium.com>

	* /, main/core_unreal.c: core_unreal: Fix off by one buffer
	  overwrite error. Appending the ;2 to the user supplied ;1
	  uniqueid to create the ;2 version if the user did not also supply
	  an extra uniqueid for the ;2 channel resulted in allocating a
	  buffer that was one byte too small. * Fix off by one error in
	  ast_unreal_new_channels() when generating the ;2 uniqueid from
	  the user suppled ;1 version. * Pulled some long assignment lines
	  from if tests to improve line break readability in
	  ast_unreal_new_channels(). ........ Merged revisions 417119 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-23 07:44 +0000 [r417059]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
	  suspended destructions of pri spans on events If a DAHDI span
	  disappears, we wish for its representation in Asterisk to be
	  destroyed as well. The information about the span's removal may
	  come from several paths: 1. DAHDI sends DAHDI_EVENT_REMOVE on
	  every channel. 2. An extra DAHDI_EVENT_REMOVED is sent on every
	  subsequent call to DAHDI_GET_EVENT. 3. Every read (including the
	  internal one by libpri on the D-channel) returns -ENODEV.
	  Asterisk responsds to DAHDI_EVENT_REMOVE on a channel by
	  destroying it. Destroying a channel requires holding the channel
	  list lock (iflock). Destroying a channel that is part of a span
	  requires holding the span's lock. Destroying a channel from a
	  context that holds the span lock, while at the same time another
	  channel is destroyed directly, leads to a deadlock. Solution:
	  don't destroy span while holding the channels list lock. Thus
	  changes in this patch: * Deferring removal of PRI spans in
	  response to events: doomed spans are collected on a list. *
	  Doomed spans are removed periodically by the monitor thread. *
	  ENODEV reads from the D-channel will warant the same deferred
	  removal. Review: https://reviewboard.asterisk.org/r/3548/

2014-06-22 18:53 +0000 [r416996]  George Joseph <george.joseph@fairview5.com>

	* include/asterisk/astobj2.h, Makefile.rules, Makefile, /: astobj2:
	  Add an ao2_replace macro to astobj2.h This macro replaces one
	  object reference with another cleaning up the original. param dst
	  Pointer to the object that will be cleaned up. param src Pointer
	  to the object replacing it. src's ref count is bumped if it's
	  non-NULL. dst's ref count is decremented if it's non-NULL. src is
	  assigned to dst, This patch was reviewed on IRC by coreyfarrell
	  and mjordan. Tested by: George Joseph ........ Merged revisions
	  416995 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-20 23:18 +0000 [r416872-416935]  George Joseph <george.joseph@fairview5.com>

	* /, configure, include/asterisk/autoconfig.h.in: build: Allow
	  autoconf/ast_ext_tool_check to handle cross-compiling better.
	  ast_ext_tool_check.m4 isn't handling cases where a path to a
	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
	  the package has a config tool (E.G. mysql_config) and the package
	  has its own subdirectories in include or lib. For example,
	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
	  fail and there are others in the same boat. The problem is caused
	  by logic in ast_ext_tool_check that overrides the result of the
	  config tool's --cflags and --libs options if package_DIR is set.
	  This patch prepends package_DIR (if specified) to the -L and -I
	  results from the package's config tool instead of overriding
	  them. A regenerated ./configure and
	  include/asterisk/autoconfig.h.in are included but can be
	  regenerated by running ./bootstrap.sh at any time. Tested by:
	  George Joseph Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3550/ ........ Merged
	  revisions 416929 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 416930 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416931 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* autoconf/ast_ext_tool_check.m4, /: build: Allow
	  autoconf/ast_ext_tool_check to handle cross-compiling better.
	  ast_ext_tool_check.m4 isn't handling cases where a path to a
	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
	  the package has a config tool (E.G. mysql_config) and the package
	  has its own subdirectories in include or lib. For example,
	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
	  fail and there are others in the same boat. The problem is caused
	  by logic in ast_ext_tool_check that overrides the result of the
	  config tool's --cflags and --libs options if package_DIR is set.
	  This patch prepends package_DIR (if specified) to the -L and -I
	  results from the package's config tool instead of overriding
	  them. Tested by: George Joseph Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3550/ ........ Merged
	  revisions 416870 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416871 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-20 20:57 +0000 [r416848-416850]  Jonathan Rose <jrose@digium.com>

	* res/parking/parking_manager.c, /: res_parking: Make manager
	  commands register with module information Previously module
	  information was not included due to an oversight. Review:
	  https://reviewboard.asterisk.org/r/3626/ ........ Merged
	  revisions 416849 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/logger.c, CHANGES, include/asterisk/logger.h,
	  main/manager.c: Logger: Add manager command 'LoggerRotate' to
	  rotate logger Part of a series of AMI command equivalents to
	  existing CLI commands Review:
	  https://reviewboard.asterisk.org/r/3651/

2014-06-20 17:06 +0000 [r416830]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_voicemail.c, include/asterisk/app.h, main/app.c,
	  apps/app_directory.c, apps/app_chanspy.c: voicemail API
	  callbacks: Extract the sayname API call to its own registerd
	  callback. * Extract the sayname API call to its own registerd
	  callback. This allows the app_directory and app_chanspy
	  applications to say a mailbox owner's name using an alternate
	  provider when app_voicemail is not available because you are
	  using res_mwi_external. app_directory still uses the
	  voicemail.conf file. AFS-64 #close Reported by: Mark Michelson

2014-06-20 15:27 +0000 [r416738-416807]  George Joseph <george.joseph@fairview5.com>

	* main/astobj2_private.h, main/astobj2_container_private.h,
	  main/astobj2_container.c, main/astobj2_hash.c,
	  main/astobj2_rbtree.c, build_tools/cflags.xml, /,
	  tests/test_astobj2.c: astobj2: Additional refactoring to push
	  impl specific code down into the impls. Move some implementation
	  specific code from astobj2_container.c into astobj2_hash.c and
	  astobj2_rbtree.c. This completely removes the need for
	  astobj2_container to switch on RTTI and it no longer has any
	  knowledge of the implementation details. Also adds AO2_DEBUG as a
	  new compile option in menuselect which controls astobj2 debugging
	  independently of AST_DEVMODE and REF_DEBUG. Tested by: George
	  Joseph Review: https://reviewboard.asterisk.org/r/3593/ ........
	  Merged revisions 416806 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_endpoint_identifier_ip.c, main/acl.c,
	  include/asterisk/netsock2.h, include/asterisk/acl.h,
	  main/netsock2.c: pjsip cli: Change Identify to show CIDR notation
	  instead of netmasks. * Added ast_sockaddr_cidr_bits() to count
	  the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which
	  uses ast_sockaddr_cidr_bits() for the netmask instead of
	  ast_sockaddr_stringify_addr. * Changed
	  res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr()
	  instead of ast_ha_join() for the CLI output. This is a CLI change
	  only. AMI was not affected. Tested by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3652/ ........ Merged
	  revisions 416737 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-19 19:40 +0000 [r416736]  Kinsey Moore <kmoore@digium.com>

	* /, main/bridge.c, res/parking/parking_tests.c,
	  channels/sip/reqresp_parser.c, main/logger.c, main/test.c: Fix
	  build warnings with TEST_FRAMEWORK enabled ........ Merged
	  revisions 416732 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 416733 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416734 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-19 16:04 +0000 [r416589-416670]  George Joseph <george.joseph@fairview5.com>

	* pbx/pbx_lua.c, /: Remove the problematic and unneeded
	  AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c
	  AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be
	  incorrectly loaded before pbx_config. pbx_config was therefore
	  blowing away contexts that were created by pbx_lua. With
	  AST_MODFLAG_DEFAULT the load order is now correct and contexs are
	  being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed
	  anyway since no other modules needed its global symbols that
	  early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by:
	  Dennis Guse Tested by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3629/ ........ Merged
	  revisions 416668 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416669 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* configs/extensions.lua.sample, /: Update extensions.lua.sample
	  with naming conflict guidance. The sample extensions.lua was
	  causing pbx_lua to fail to load when parsing 'app.goto("default",
	  "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This
	  patch adds guidance to extensions.lua.sample and changed
	  'app.goto("default", "s", 1)' to 'app.['goto']("default", "s",
	  1)'. ASTERISK-23844 #close Reported by: rnewton Tested by:
	  gtjoseph Review: https://reviewboard.asterisk.org/r/3627/
	  ........ Merged revisions 416581 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416582 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-18 04:22 +0000 [r416561]  Matthew Jordan <mjordan@digium.com>

	* /, main/stasis_channels.c: stasis_channels: Update the stasis
	  cache if manager variables are needed In r416211, the publishing
	  of variable changes was modified such that a cached channel
	  snapshot was used if manager variables were not requested with
	  each AMI event. This was done to reduce the amount of channel
	  snapshots created. However, an assumption was made that
	  generating a channel snapshot and publishing the snapshot to the
	  channel topic was sufficient to ensure that the cache would be
	  updated; this is not the case. The channel snapshot type must be
	  used to force a snapshot update. This patch updates the
	  publication of channel variables such that the cache is updated
	  prior to publication of the channel variable message if manager
	  variables are in use. This ensures that all AMI events receive
	  the variable update when they are supposed to. Note that this
	  issue was caught by the Asterisk Test Suite (go go testing)
	  ........ Merged revisions 416557 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-17 18:45 +0000 [r416444-416503]  Mark Michelson <mmichelson@digium.com>

	* /, funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to
	  set inheritable channel variables. ........ Merged revisions
	  416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 416501 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416502 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_pidf_body_generator.c, /,
	  res/res_pjsip_xpidf_body_generator.c: Fix string growth algorithm
	  for XML presence bodies. pjpidf_print() does not return < 0 if
	  there is not enough room for the document to be printed. Rather,
	  it returns 39, the length of the XML prolog. The algorithm also
	  had a bug in that it would return if it attempted to grow the
	  string larger. ........ Merged revisions 416442 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-17 16:33 +0000 [r416443]  Kinsey Moore <kmoore@digium.com>

	* res/res_musiconhold.c, /: MoH: Don't restart stream on repeated
	  start calls Currently, music on hold will stop and then start
	  again from the beginning if ast_moh_start() is called multiple
	  times. This can happen if a call is put on hold repeatedly (the
	  channel receives multiple HOLD control frames) and can be
	  triggered from ARI by starting MoH on a channel multiple times.
	  This is fairly jarring/annoying to users. This change prevents
	  MoH from being restarted if the requested music class is the same
	  as the one currently playing. This includes an extra check to
	  prevent the errors previously experienced in the testsuite and
	  has 100+ test runs behind it. Review:
	  https://reviewboard.asterisk.org/r/3615/ ........ Merged
	  revisions 416439 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 416440 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416441 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-16 18:27 +0000 [r416416]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
	  channels/sig_ss7.h, configure, channels/chan_dahdi.h,
	  configure.ac, UPGRADE.txt, configs/ss7.timers.sample (added),
	  CHANGES, channels/sig_ss7.c: chan_dahdi: Adds support for major
	  update to libss7. * SS7 support now requires libss7 v2.0 or
	  later. The new libss7 is not backwards compatible. * Added SS7
	  support for connected line and redirecting. * Most SS7 CLI
	  commands are reworked as well as new SS7 commands added. See
	  online CLI help. * Added several SS7 config option parameters
	  described in chan_dahdi.conf.sample. * ISUP timer support
	  reworked and now requires explicit configuration. See
	  ss7.timers.sample. Special thanks to Kaloyan Kovachev for his
	  support and persistence in getting the original patch by adomjan
	  updated and ready for release. SS7-27 #close Reported by: adomjan

2014-06-16 16:22 +0000 [r416394]  Kevin Harwell <kharwell@digium.com>

	* include/asterisk/http_websocket.h, tests/test_websocket_client.c,
	  res/res_http_websocket.c: res_http_websocket: read/write string
	  fixup There was a problem when reading a string from the
	  websocket. It assumed the received data had a null terminator and
	  tried to write the data to an ast_str. This of course could/would
	  read past the end of the given buffer while writing the data to
	  the internal buffer of ast_str. Modified the the code to
	  correctly place a null terminator on the result string.

2014-06-16 09:04 +0000 [r416339]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* cel/cel_sqlite3_custom.c, main/db.c, res/res_config_sqlite3.c,
	  cdr/cdr_sqlite3_custom.c, /: We have faced situation when using
	  CDR and CEL by sqlite3 modules. With system having high load
	  (~100 concurrent calls created by sipp) we found many cdr and cel
	  records missed. There is special finction in sqlite3, that make
	  able to fix this situation - sqlite3_wait_timeout, that also can
	  replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this
	  function can be used for aastdb and res_config_sqlite3 to avoid
	  missed writes to sqlite db. #ASTERISK-23766 #close Reported by:
	  Igor Goncharovsky Review:
	  https://reviewboard.asterisk.org/r/3559/ ........ Merged
	  revisions 416336 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 416337 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416338 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-16 02:40 +0000 [r416267-416319]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: channels/chan_sip: Forbid remote bridging
	  if T.38 is negotiated When a framehook is removed - such as the
	  fax gateway framehook - the bridge framework will re-evaluate the
	  bridge mixing technologies to see if it can improve the bridging.
	  When this occurs, get_rtp_info will be called to determine if
	  local or remote bridging can be used. Using remote bridging will
	  cause a fax to fail, as direct media negotiation will cause some
	  small number of packets to not arrive at the remote endpoint.
	  This patch forces local native bridging if T.38 negotiation is in
	  progress or has been established. ........ Merged revisions
	  416318 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/channel_internal_api.c: channel_internal_api: Publish a
	  snapshot change when linkedids change Snapshots are now not
	  published *quite* as much as they used to. One instance where
	  they are not published any longer is during bridge enter and exit
	  - the state of the channel doesn't change, the bridge does.
	  However, channels are changed when a linkedid is propagated;
	  previously, the channel's state would be updated and published
	  during the bridge enter event. Now this must be explicitly done.
	  ........ Merged revisions 416300 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, tests/test_stasis_endpoints.c: test_stasis_endpoints: Remove
	  expected channel snapshot We no longer publish a channel snapshot
	  when it is associated with an endpoint; after all, the channel
	  itself hasn't changed - the endpoint state has changed. This
	  updates the channel_messages unit test accordingly. ........
	  Merged revisions 416298 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_musiconhold.c: MoH: Undo commit r416150 (1.8) This
	  patch reverts r416150. When the comparison between mohclass->name
	  and state->class->name is made, you are not guaranteed that (a)
	  state->class is non-NULL or that state or state->class are in a
	  safe state. Crashes caught by the bridges/transfer_capabilities
	  test. ........ Merged revisions 416251 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 416252 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416255 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-14 19:26 +0000 [r416237]  Corey Farrell <git@cfware.com>

	* res/res_manager_devicestate.c, res/res_manager_presencestate.c:
	  res_manager_devicestate and res_manager_presencestate missing
	  support level Add MODULEINFO comment block to define support
	  level core for these new modules. Review:
	  https://reviewboard.asterisk.org/r/3620/

2014-06-13 18:24 +0000 [r416216]  Matthew Jordan <mjordan@digium.com>

	* res/res_agi.c, res/res_pjsip/pjsip_configuration.c,
	  main/stasis_channels.c, res/ari/resource_channels.c,
	  main/bridge_channel.c, main/pbx.c, main/stasis_cache.c, /,
	  apps/app_meetme.c, main/pickup.c, main/channel_internal_api.c,
	  include/asterisk/channel.h, main/core_local.c, main/aoc.c,
	  main/endpoints.c, main/cel.c, apps/app_queue.c,
	  main/stasis_bridges.c, apps/app_agent_pool.c, main/cli.c,
	  main/channel.c, main/dial.c, main/manager.c,
	  include/asterisk/stasis_channels.h: stasis: Reduce creation of
	  channel snapshots to improve performance During some performance
	  testing of Asterisk with AGI, ARI, and lots of Local channels, we
	  noticed that there's quite a hit in performance during channel
	  creation and releasing to the dialplan (ARI continue). After
	  investigating the performance spike that occurs during channel
	  creation, we discovered that we create a lot of channel snapshots
	  that are technically unnecessary. This includes creating
	  snapshots during: * AGI execution * Returning objects for ARI
	  commands * During some Local channel operations * During some
	  dialling operations * During variable setting * During some
	  bridging operations And more. This patch does the following: - It
	  removes a number of fields from channel snapshots. These fields
	  were rarely used, were expensive to have on the snapshot, and
	  hurt performance. This included formats, translation paths, Log
	  Call ID, callgroup, pickup group, and all channel variables. As a
	  result, AMI Status, "core show channel", "core show channelvar",
	  and "pjsip show channel" were modified to either hit the live
	  channel or not show certain pieces of data. While this is
	  unfortunate, the performance gain from this patch is worth the
	  loss in behaviour. - It adds a mechanism to publish a cached
	  snapshot + blob. A large number of publications were changed to
	  use this, including: - During Dial begin - During Variable
	  assignment (if no AMI variables are emitted - if AMI variables
	  are set, we have to make snapshots when a variable is changed) -
	  During channel pickup - When a channel is put on hold/unhold -
	  When a DTMF digit is begun/ended - When creating a bridge
	  snapshot - When an AOC event is raised - During Local channel
	  optimization/Local bridging - When endpoint snapshots are
	  generated - All AGI events - All ARI responses that return a
	  channel - Events in the AgentPool, MeetMe, and some in Queue -
	  Additionally, some extraneous channel snapshots were being made
	  that were unnecessary. These were removed. - The result of
	  ast_hashtab_hash_string is now cached in stasis_cache. This
	  reduces a large number of calls to ast_hashtab_hash_string, which
	  reduced the amount of time spent in this function in gprof by
	  around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged
	  revisions 416211 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-13 13:11 +0000 [r416149-416153]  Kinsey Moore <kmoore@digium.com>

	* res/res_musiconhold.c, /: MoH: Don't restart stream on repeated
	  start calls Currently, music on hold will stop and then start
	  again from the beginning if ast_moh_start() is called multiple
	  times. This can happen if a call is put on hold repeatedly (the
	  channel receives multiple HOLD control frames) and can be
	  triggered from ARI by starting MoH on a channel multiple times.
	  This is fairly jarring/annoying to users. This change prevents
	  MoH from being restarted if the requested music class is the same
	  as the one currently playing. Review:
	  https://reviewboard.asterisk.org/r/3615/ ........ Merged
	  revisions 416150 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 416151 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416152 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cel.c, /: CEL: Expose parking retreiver in extra field This
	  exposes the retreiver of a parked call under the "retreiver" key
	  of the extra field when this information is available. Review:
	  https://reviewboard.asterisk.org/r/3608/ ........ Merged
	  revisions 416148 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-13 05:16 +0000 [r416071]  Richard Mudgett <rmudgett@digium.com>

	* main/http.c, include/asterisk/tcptls.h, main/tcptls.c,
	  main/manager.c, /, channels/chan_sip.c: AST-2014-007: Fix of fix
	  to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close
	  Reported by: Richard Mudgett Review:
	  https://reviewboard.asterisk.org/r/3617/ ........ Merged
	  revisions 416066 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 416067 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 416070 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 21:27 +0000 [r416024]  Rusty Newton <rnewton@digium.com>

	* main/pbx.c: main/pbx - documentation - enhance 'core show hints'
	  and 'core show hint' help text Adds descriptive help text to
	  'core show hints' and 'core show hint'. The text describes the
	  various columns for the sake of clarity. It takes into account
	  recent changes to the content displayed by the commands
	  https://reviewboard.asterisk.org/r/3604/ and
	  https://reviewboard.asterisk.org/r/3611/. ASTERISK-23764 Review:
	  https://reviewboard.asterisk.org/r/3610/

2014-06-12 20:17 +0000 [r415982]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip_pubsub.c, /: Fix build in devmode for GCC 4.10
	  ........ Merged revisions 415980 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 17:00 +0000 [r415907]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/utils.h, main/tcptls.c, main/manager.c, /,
	  channels/chan_sip.c, main/http.c, UPGRADE.txt, main/utils.c,
	  include/asterisk/tcptls.h, res/res_http_websocket.c,
	  configs/http.conf.sample: AST-2014-007: Fix DOS by consuming the
	  number of allowed HTTP connections. Simply establishing a TCP
	  connection and never sending anything to the configured HTTP port
	  in http.conf will tie up a HTTP connection. Since there is a
	  maximum number of open HTTP sessions allowed at a time you can
	  block legitimate connections. A similar problem exists if a HTTP
	  request is started but never finished. * Added http.conf
	  session_inactivity timer option to close HTTP connections that
	  aren't doing anything. Defaults to 30000 ms. * Removed the
	  undocumented manager.conf block-sockets option. It interferes
	  with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections
	  now have better authentication timeout protection. Though I
	  didn't remove the bizzare TLS timeout polling code from chan_sip.
	  * chan_sip can now handle SSL certificate renegotiations in the
	  middle of a session. It couldn't do that before because the
	  socket was non-blocking and the SSL calls were not restarted as
	  documented by the OpenSSL documentation. * Fixed an off nominal
	  leak of the ssl struct in handle_tcptls_connection() if the FILE
	  stream failed to open and the SSL certificate negotiations
	  failed. The patch creates a custom FILE stream handler to give
	  the created FILE streams inactivity timeout and timeout after a
	  specific moment in time capability. This approach eliminates the
	  need for code using the FILE stream to be redesigned to deal with
	  the timeouts. This patch indirectly fixes most of ASTERISK-18345
	  by fixing the usage of the SSL_read/SSL_write operations.
	  ASTERISK-23673 #close Reported by: Richard Mudgett ........
	  Merged revisions 415841 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415854 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415896 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 15:50 +0000 [r415839]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, apps/app_queue.c: app_queue: delayed state can cause early
	  leavewhenempty ringing In app_queue, device state changes arrive
	  in event messages and update the queue member status value. That
	  value is checked in get_member_status() to decide that the caller
	  should leave when there are no available members. Although event
	  messages can be delayed by other activity, there is no adverse
	  affect by lagged status except in one specific case: there is
	  only one available member, it was just rung, and leavewhenempty
	  is enabled set for ringing members. This change adds a direct
	  check of the device state only under this condition where the
	  caller may be dropped incorrectly, resolving this issue without
	  affecting performance of app_queue normally. AST-1248 #close
	  Review: https://reviewboard.asterisk.org/r/3595/ Reported by:
	  Thomas Arimont ........ Merged revisions 415833 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415835 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415836 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 15:39 +0000 [r415834]  Jonathan Rose <jrose@digium.com>

	* apps/app_mixmonitor.c, /, UPGRADE.txt: MixMontior: Add class
	  authorization requirements to MixMonitor AMI commands MixMonitor
	  AMI commands StartMixMonitor and StopMixMonitor lacked class
	  authorization. StopMixMonitor now requires that the manager user
	  either have the call or system class authorization.
	  StartMixMonitor is a slightly larger issue since it can execute
	  shell commands if the right arguments are passed into it, and we
	  consider this a permission escalation. A security release will be
	  issued for problem this shortly. ASTERISK-23609 #close Reported
	  by: Corey Farrell ........ Merged revisions 415825 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415832 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 14:39 +0000 [r415813]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: unauthenticated
	  remote crash in PJSIP pub/sub framework A remotely exploitable
	  crash vulnerability exists in the PJSIP channel driver's pub/sub
	  framework. If an attempt is made to unsubscribe when not
	  currently subscribed and the endpoint's "sub_min_expiry" is set
	  to zero, Asterisk tries to create an expiration timer with zero
	  seconds, which is not allowed, so an assertion raised. The fix
	  was to reject a subscription that is attempting to unsubscribe
	  when not being already subscribed. Asterisk now checks for this
	  situation appropriately and responds with a 400 instead of
	  crashing. AST-2014-005 ASTERISK-23489 #close ........ Merged
	  revisions 415812 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 14:15 +0000 [r415795]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip.c, /: Fix potential deadlock situation in
	  res_pjsip. SIP transaction timeouts are handled in the PJSIP
	  monitor thread. When this happens on a subscription, and the
	  subscription is destroyed, the subscription destruction is
	  dispatched synchronously to the threadpool. The issue is that the
	  PJSIP dialog is locked by the monitor thread, and then the
	  dispatched task attempts to lock the dialog. This leads to a
	  deadlock that causes SIP traffic to no longer be accepted on the
	  Asterisk server. The fix here is to treat the monitor thread as
	  if it were a threadpool thread when it attempts to dispatch
	  synchronous tasks. This way, the dispatched task turns into a
	  simple function call within the same thread, and the locking
	  issue is averted. AST-2014-008 ASTERISK-23802 #close ........
	  Merged revisions 415794 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 11:34 +0000 [r415767]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip.c, res/res_pjsip_pubsub.c,
	  res/res_pjsip_exten_state.c, include/asterisk/res_pjsip.h,
	  include/asterisk/res_pjsip_pubsub.h,
	  res/res_pjsip_pubsub.exports.in, /,
	  contrib/ast-db-manage/config/versions/c6d929b23a8_create_pjsip_subscription_persistence_.py
	  (added), res/res_pjsip_mwi.c: res_pjsip_pubsub: Persist
	  subscriptions in sorcery so they are recreated on startup. This
	  change makes res_pjsip_pubsub persist inbound subscriptions in
	  sorcery. By default this uses the local astdb but it can also be
	  configured to store within an outside database. When Asterisk is
	  started these subscriptions are recreated if they have not
	  expired. Notifications are sent to the devices which have
	  subscribed and they are none the wiser that the system has
	  restarted. Review: https://reviewboard.asterisk.org/r/3598/
	  ........ Merged revisions 415766 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-12 07:52 +0000 [r415749]  Walter Doekes <walter+asterisk@wjd.nu>

	* UPGRADE.txt, contrib/scripts/safe_asterisk, Makefile, /:
	  safe_asterisk: Overwrite old safe_asterisk on make install. From
	  now on, make install will overwrite safe_asterisk with the latest
	  version. You need to move any local modifications to files inside
	  /etc/asterisk/startup.d, if you have any. See also commits
	  r394939 and r397938. ASTERISK-21965 #close Patches:
	  safe_asterisk.patch uploaded by jkister (License 6232, modified
	  by me) ........ Merged revisions 415748 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-11 23:01 +0000 [r415730]  Richard Mudgett <rmudgett@digium.com>

	* main/format.c, /: format.c: Fix misuse of hash container
	  function. The supplied hash function to a container must be
	  idempotent given the object's key value to figure out which
	  container bucket the object belongs in. Returning a random number
	  or the current container count is not idempotent. The "computed
	  hash" value doesn't help find the object later in those cases. *
	  Fixed the format_list container to actually be a list since that
	  is how the container is used. Conceptually, if more than 283
	  formats were added to the format_list then odd things may have
	  happened before the fix. ........ Merged revisions 415728 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415729 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-11 20:22 +0000 [r415698-415715]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/pbx.c: CLI: correct presence information on core show hints
	  Adds presence to core show hint and changes presence string
	  conversion to use the correct function. ASTERISK-23858 #close
	  Review: https://reviewboard.asterisk.org/r/3611/

	* main/pbx.c: CLI: add presence information to core show hints Adds
	  presence state value to output of core show hints. Also reformats
	  the output slightly so it doesn't use as much space as it would
	  otherwise. Was: 1000@demo : SIP/1000 State:Unavailable Watchers 0
	  Now: 1000@demo : SIP/1000 State:Unavailable Presence:Idle
	  Watchers 0 AFS-53 #close Review:
	  https://reviewboard.asterisk.org/r/3604/

2014-06-10 18:32 +0000 [r415679]  Kinsey Moore <kmoore@digium.com>

	* main/channel.c, /: Fix build in dev mode due to signed/unsigned
	  mismatch ........ Merged revisions 415678 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-10 16:06 +0000 [r415659]  Jonathan Rose <jrose@digium.com>

	* main/message.c, /, res/res_pjsip_notify.c: PJSIP: PJSIPNotify -
	  Strip content-length headers and add documentation Documentation
	  for how to add custom headers/content to notifies created with
	  the PJSIPNotify manager action was a little sparse and it also
	  wasn't vetting application of Content-length headers like its
	  chan_sip equivalent was (so two Content-length headers could be
	  applied... and PJSIP determines the content length anyway, so it
	  just opens people up for error). This patch also flips the
	  variable order so that the variables are interpreted in the same
	  order as they are put in the AMI action. Review:
	  https://reviewboard.asterisk.org/r/3587/ ........ Merged
	  revisions 415658 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-10 09:28 +0000 [r415630]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: chan_ooh323: fix loading module failure
	  if there no accessible h323_log or ooh323 config file change
	  return 1 to return AST_MODULE_LOAD_FAILURE on module load routine
	  few cosmetic changes ASTERISK-23814 #close (closes issue
	  ASTERISK-23814) Reported by: Igor Goncharovsky Patches:
	  ASTERISK-23814-ast11.patch ........ Merged revisions 415599 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415602 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-09 20:21 +0000 [r415580]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_header_funcs.c, /: chan_pjsip: Fix bug where custom
	  SIP headers could be duplicated on outgoing INVITEs. When using
	  PJSIP_HEADER() to add custom headers to outgoing INVITE requests,
	  certain situations could result in the headers being duplicated.
	  For instance, if the request were retransmitted, or if the INVITE
	  were re-sent with authentication credentials, the custom headers
	  would be re-added to the request. The fix here is to, after
	  adding the custom headers to the outbound INVITE, remove the
	  datastore that holds the custom headers to add. This way, there
	  is no risk in accidentally adding them if the session supplement
	  is called into a second or third time. ........ Merged revisions
	  415579 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-09 12:12 +0000 [r415524]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, UPGRADE.txt, contrib/scripts/safe_asterisk: safe_asterisk:
	  Cleanup additions to r415132. * Replaced a stray echo that
	  should've been a message call in safe_asterisk. This replaces a
	  conditional log message by a slightly different message. Please
	  update your log parsing scripts. * Made the $NOTIFY mail Subject
	  more verbose by adding the machine name and exitstatus. (Note
	  that a 'make install' still won't overwrite your old
	  safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492
	  #close ........ Merged revisions 415521 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415522 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415523 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-09 03:50 +0000 [r415466]  Corey Farrell <git@cfware.com>

	* /, main/autoservice.c: autoservice: stop thread on graceful
	  shutdown This change adds thread shutdown to autoservice for
	  graceful shutdowns only. ast_register_cleanup is backported to
	  1.8 to allow this. The logger callid is also released on shutdown
	  in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3594/ ........ Merged
	  revisions 415463 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415464 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415465 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-08 18:12 +0000 [r415444]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
	  main/bridge_channel.c, main/channel.c, main/pbx.c, /,
	  main/framehook.c, main/bridge_after.c: bridges/bridge_native_rtp:
	  Reconfigure bridge on removal of framehook This patch is a re-do
	  of r414122. When r414122 was merged, a major problem with it was
	  uncovered. UNBRIDGE soft hangup flags have a catastrophic effect
	  on the pbx core if they leak out from the bridge layer: the
	  channel gets hung up. With the number of threads involved in a
	  blind transfer, and with the initial patch, it was likely that
	  this would occur. This caused a large number of test failures
	  This patch is nearly identical with the one proposed in r414122,
	  save for the following changes: - We explicitly clear the
	  UNBRIDGE flag when setting an after goto on a channel in a bridge
	  - Defensively, if we encounter an UNBRIDGE flag in the pbx core,
	  we handle it https://reviewboard.asterisk.org/r/3585/ ........
	  Merged revisions 415443 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-07 00:42 +0000 [r415428]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/bridge.h, /: bridge.h: Remove redundant struct
	  ast_bridge_channel forward declaration. ........ Merged revisions
	  415427 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-06 21:44 +0000 [r415411]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/manager.h, main/config.c, main/manager.c, /,
	  channels/chan_sip.c, include/asterisk/config.h: chan_sip: Fix
	  order of variables specified in SIPNotify action Prior to this
	  patch, sequential variables would be ordered in reverse from the
	  order specified in the manager action. Review:
	  https://reviewboard.asterisk.org/r/3588/ ........ Merged
	  revisions 415359 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415390 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415410 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-06 20:45 +0000 [r415358]  Kevin Harwell <kharwell@digium.com>

	* main/uri.c, tests/test_websocket_client.c: core uri: Custom uri
	  parsing error when no query parameters If using the custom URI
	  parsing code (not external uriparser lib) and there was no query
	  parameters the resulting pointer would be NULL and then an
	  attempt was made to subtract from it. The pointer is now set to a
	  valid value if there is no query parameter(s). Also, in the
	  'ast_uri_make_host_with_port' function when setting the
	  terminator on the resulting string it was writing it one past the
	  end of allocated memory. It now writes the string terminator
	  appropriately.

2014-06-06 19:13 +0000 [r415343]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_pjsip_sdp_rtp.c: PJSIP: Remove premature write of raw
	  formats Currently, there are situations that can occur when using
	  chan_pjsip and certain dialplan applications (notably ChanSpy())
	  that can cause the channel to get no audio with scrolling
	  warnings about format mismatches. This is caused by a failure to
	  update translation paths on a mid-call native format update since
	  the raw formats have already been updated by res_pjsip_sdp_rtp.c
	  in set_caps(). Removing the premature raw format updates allows
	  the translation paths to be setup correctly and the raw read and
	  write formats with them. AFS-63 #close ........ Merged revisions
	  415342 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-06 14:12 +0000 [r415319]  George Joseph <george.joseph@fairview5.com>

	* tests/test_astobj2.c, main/astobj2_private.h (added),
	  main/astobj2.c, main/astobj2_container_private.h (added),
	  main/astobj2_container.c (added), main/astobj2_hash.c (added),
	  main/astobj2_rbtree.c (added), /, include/asterisk/astobj2.h:
	  Split astobj2.c into more maintainable components. Split
	  astobj2.c into the following files to improve maintainability.
	  astobj2.c - object primitives, object primitive misc and
	  initialization code. astobj2_private.h - internal object
	  declarations needed by the containers. astobj2_container.c -
	  generic conainer and container misc code.
	  astobj2_container_hash.c - hash container specific code.
	  astobj2_container_rbtree.c - rbtree container specific code.
	  astobj2_container_private.h - generic container definitions and
	  rtti prototypes. https://reviewboard.asterisk.org/r/3576/
	  ........ Merged revisions 415317 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-06 12:49 +0000 [r415302]  Rusty Newton <rnewton@digium.com>

	* /, configs/cli_aliases.conf.sample: configs/cli_aliases.conf: Two
	  new aliases, plus enhancements for context names. Changed naming
	  of included alias templates to avoid confusion between version
	  names. For example, asterisk12 was for asterisk 1.2, so I changed
	  it to asterisk_1dot2, so that later we can use asterisk_12 for
	  Asterisk 12. Added alias for "features reload" to the template
	  for Asterisk 11 style syntax template, as features reload was
	  removed in 12, but you can still do "module reload features"
	  Added alias for "pjsip reload" to the friendly template. It is
	  shorter than "module reload res_pjsip.so" and if some are like
	  me; I constantly forget that reloading chan_pjsip doesn't parse
	  config. Remembering "pjsip reload" is just easier. ASTERISK-23654
	  #close ASTERISK-23654 #comment Fixed by adding two new aliases
	  and enhancements for context names. Review:
	  https://reviewboard.asterisk.org/r/3572/ ........ Merged
	  revisions 415301 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-05 19:04 +0000 [r415231-415288]  Richard Mudgett <rmudgett@digium.com>

	* main/config.c: config: Fix indentation and missing curlies in
	  config_text_file_load().

	* main/config.c, /: config: Fix config files not reloading when
	  only an included file changes. The twisted logic determining if a
	  config file should be reloaded was mostly broken and disabled.
	  The incorrect test that ASTERISK-23383 fixed actually reenabled
	  the broken logic. The incorrect test was causing the timestamp to
	  always be cleared which caused config files with includes to
	  always be reloaded. * Made wildcard includes always cause a
	  reload. Determining if a file was deleted cannot be determined
	  without restructuring the cache to determine if any files are
	  missing from the last files actually loaded. Also without
	  refactoring config_text_file_load(), the glob loop couldn't check
	  more than one file for changes anyway. * Made remove the cache
	  entry if the file no longer exists when trying to get its
	  timestamp or it is no longer a regular file. This fixes the
	  corner case where the file was loaded, then deleted, then the
	  config reloaded, then the file restored with the same timestamp,
	  and then the config reloaded again. * Made remove the cache entry
	  include list when actually loading the file. This gets rid of any
	  stale includes the file had from the last time the file was
	  loaded. ASTERISK-23683 #close Reported by: tootai Review:
	  https://reviewboard.asterisk.org/r/3575/ ........ Merged
	  revisions 415225 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415229 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415230 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-05 17:22 +0000 [r415223]  Kevin Harwell <kharwell@digium.com>

	* tests/test_uri.c (added), include/asterisk/http_websocket.h,
	  main/http.c, main/uri.c (added), tests/test_websocket_client.c
	  (added), res/res_http_websocket.c, include/asterisk/http.h,
	  include/asterisk/uri.h (added),
	  res/res_http_websocket.exports.in: res_http_websocket: Create a
	  websocket client Added a websocket server client in Asterisk.
	  Asterisk has a websocket server, but not a client. The ability to
	  have Asterisk be able to connect to a websocket server can
	  potentially be useful for future work (for instance this could
	  allow ARI to connect back to some external system, although more
	  work would be needed in order to incorporate that). Also a couple
	  of things to note - proxy connection support has not been
	  implemented and there is limited http response code handling
	  (basically, it is connect or not). Also added an initial new URI
	  handling mechanism to core. Internet type URI's are parsed into a
	  data structure that contains pointers to the various parts of the
	  URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell
	  Review: https://reviewboard.asterisk.org/r/3541/

2014-06-05 14:49 +0000 [r415208]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_confbridge.c: app_confbridge: Allow muting of users
	  waiting to enter a ConfBridge Prior to this patch, users waiting
	  to enter a ConfBridge were not considered when muted via the CLI
	  or via AMI. Instead, a confusing message would be emitted stating
	  that the channel did not exist. This patch allows a user to be
	  muted when waiting to enter a ConfBridge conference. This is
	  equivalent to start when muted, only toggled via the CLI or AMI.
	  Review: https://reviewboard.asterisk.org/r/3582 #ASTERISK-23824
	  #close patches: rb3582.patch uploaded by tm1000 (License 6524)
	  ........ Merged revisions 415206 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415207 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-05 11:59 +0000 [r415192]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_pjsip.c: PJSIP: Send initial connected line
	  information This makes chan_pjsip send connected line information
	  when it is called so that connected line information is available
	  on the connected channel. (closes issue DPMA-442) Reported by:
	  John Bigelow Review: https://reviewboard.asterisk.org/r/3584/
	  ........ Merged revisions 415191 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-04 20:16 +0000 [r415173]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, contrib/scripts/safe_asterisk: safe_asterisk: Cleanup and
	  debian compatibility. Cleans up the safe_asterisk script and adds
	  the ASTSAFE_FOREGROUND option that allows the debian asterisk
	  init script to capture the right pid. * Drop the vim #modeline
	  which wasn't used. Use test consistently without the odd
	  configure xno syntax. Double quote all paths. General cleanup. *
	  Don't output message()s to the console but only to TTY if set. *
	  Allow TTY to be "no" as well as empty (debian compatibility with
	  debian/patches/safe_asterisk-config). * Add option to export
	  ASTSAFE_FOREGROUND=1 from the init script that calls this to
	  disable backgrounding. Debian uses a similar method in
	  debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review:
	  https://reviewboard.asterisk.org/r/3574/ ........ Merged
	  revisions 415132 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415171 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415172 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-04 14:13 +0000 [r415116-415118]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_pjsip.c: chan_pjsip: Add debug in RTP Engine
	  glue callback This patch adds some debug statements that aid with
	  determining why a direct media request may or may not be
	  initiated. ........ Merged revisions 415117 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_session.c, /: res_pjsip_session: Add debug
	  statement for session refreshes This small patch adds a debug
	  level 3 statement indicating how a session refresh is being sent
	  - either as a re-INVITE or as an UPDATE - and where the session
	  refresh is going. ........ Merged revisions 415115 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-04 07:27 +0000 [r415080]  Corey Farrell <git@cfware.com>

	* /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
	  app_confbridge: Correct verification of conference name length
	  Conference names were not checked for maximum length, allowing
	  unexpected behaviour. This change adds checking to ensure the
	  maximum length is not exceeded. The maximum length is also
	  changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close
	  Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches:
	  confbridge-enforce_max-1.8.patch uploaded by coreyfarrell
	  (license 5909) confbridge-enforce_max-11up.patch uploaded by
	  coreyfarrell (license 5909) ........ Merged revisions 415060 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 415066 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 415078 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-03 07:36 +0000 [r415000]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, funcs/func_odbc.c: func_odbc: Fix fixed size buffers fix
	  (r414968). The change that removed the fixed size buffers in
	  odbc-related code -- removing arbitrary column width limits --
	  was incomplete. This change adds: no segfault on writesql without
	  insertsql and return value checks after strdup. While I was in
	  the vicinity I cleaned up the linefeeds in the odbc function
	  descriptions, moved some code for clarity, removed some blobs and
	  noted (but didn't fix) that the 'odbc write ... exec' CLI command
	  doesn't behave as the dialplan equivalent when insertsql= is
	  used. ASTERISK-23582 #close Review:
	  https://reviewboard.asterisk.org/r/3579/ ........ Merged
	  revisions 414997 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414998 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414999 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-06-01 15:32 +0000 [r414976]  Joshua Colp <jcolp@digium.com>

	* /, bridges/bridge_native_rtp.c: bridge_native_rtp: Take the
	  bridge type choice of both channels into account. The
	  bridge_native_rtp module currently uses the bridge result of the
	  first channel that joins a bridge as the ultimate result. This
	  means that if the first channel has direct media enabled but the
	  second does not a direct media bridge will still occur. This
	  change makes it so that both sides are taken into account. If
	  either side forbids the bridge or responds with a local bridge
	  result then either a generic or local bridge occurs.
	  ASTERISK-23541 #close Reported by: Justin E Review:
	  https://reviewboard.asterisk.org/r/3577/ ........ Merged
	  revisions 414975 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-30 14:53 +0000 [r414949]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip_refer.c, /: PJSIP: Prevent crash on blind transfer
	  Blind transfers don't go too well with NULL channels which can
	  occur if the channel has already been transferred away. (closes
	  issue ASTERISK-23718) Reported by: Jonathan Rose ........ Merged
	  revisions 414948 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-30 12:42 +0000 [r414883-414935]  Matthew Jordan <mjordan@digium.com>

	* main/audiohook.c, CHANGES, res/ari/ari_model_validators.c,
	  res/ari/ari_model_validators.h, funcs/func_talkdetect.c (added),
	  include/asterisk/stasis_channels.h,
	  rest-api/api-docs/events.json, /, main/stasis_channels.c:
	  TALK_DETECT: A channel function that raises events when talking
	  is detected This patch adds a new channel function TALK_DETECT
	  that, when set on a channel, causes events indicating the
	  start/stop of talking on a channel to be emitted to both AMI and
	  ARI clients. The function allows setting both the silence
	  threshold (the length of silence after which we decide no one is
	  talking) as well as the talking threshold (the amount of energy
	  that counts as talking). Parameters can be updated on a channel
	  after talk detection has been enabled, and talk detection can be
	  removed at any time. The events raised by the function use a
	  nomenclature similar to existing AMI/ARI events. For AMI:
	  ChannelTalkingStart/ChannelTalkingStop For ARI:
	  ChannelTalkingStarted/ChannelTalkingFinished Review:
	  https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close
	  Reported by: Matt Jordan ........ Merged revisions 414934 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/config.c, /: main/config.c: AMI action UpdateConfig EmptyCat
	  clears all categories When invoking UpdateConfig AMI action with
	  Action set to EmptyCat, Asterisk will make all categories empty
	  in the config but the one requested with a Cat variable. This is
	  due to a bug in ast_category_empty (main/config.c) that makes an
	  incorrect comparison for a category name. This patch corrects the
	  comparison such that only the requested category is cleared.
	  Review: https://reviewboard.asterisk.org/r/3573/ #ASTERISK-23803
	  #close Reported by: zvision patches: manager.c.diff uploaded by
	  zvision (License 5755) ........ Merged revisions 414880 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414881 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414882 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-29 18:51 +0000 [r414861]  Kinsey Moore <kmoore@digium.com>

	* main/pbx.c, /: PBX: Prevent incorrect hint parsing Dynamic and
	  pattern matching hints should not be checked for their last known
	  state until they are instantiated by subscribers. (closes issue
	  AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted
	  by Matt Jordan (license 6283) ........ Merged revisions 414813
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 414859 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414860 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-28 22:54 +0000 [r414798]  Matthew Jordan <mjordan@digium.com>

	* main/loader.c, include/asterisk/logger.h, res/res_config_curl.c,
	  cel/cel_odbc.c, res/res_config_odbc.c,
	  bridges/bridge_builtin_features.c, main/optional_api.c,
	  main/logger.c, main/config_options.c, cdr/cdr_odbc.c,
	  apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c,
	  main/xmldoc.c, apps/app_voicemail.c, cel/cel_pgsql.c,
	  channels/chan_unistim.c, res/res_config_pgsql.c, main/pbx.c,
	  cdr/cdr_sqlite3_custom.c, res/res_fax.c, main/bridge.c,
	  apps/app_waitforsilence.c, cdr/cdr_adaptive_odbc.c,
	  res/parking/parking_applications.c, cdr/cdr_pgsql.c,
	  res/res_jabber.c: Logger/CLI/etc.: Fix some aesthetic issues;
	  reduce chatty verbose messages This patch addresses some
	  aesthetic issues in Asterisk. These are all just minor tweaks to
	  improve the look of the CLI when used in a variety of settings.
	  Specifically: * A number of chatty verbose messages were removed
	  or demoted to DEBUG messages. Verbose messages with a verbosity
	  level of 5 or higher were - if kept as verbose messages - demoted
	  to level 4. Several messages that were emitted at verbose level 3
	  were demoted to 4, as announcement of dialplan applications being
	  executed occur at level 3 (and so the effects of those
	  applications should generally be less). * Some verbose messages
	  that only appear when their respective 'debug' options are
	  enabled were bumped up to always be displayed. *
	  Prefix/timestamping of verbose messages were moved to the
	  verboser handlers. This was done to prevent duplication of
	  prefixes when the timestamp option (-T) is used with the CLI. *
	  Verbose magic is removed from messages before being emitted to
	  non-verboser handlers. This prevents the magic in multi-line
	  verbose messages (such as SIP debug traces or the output of
	  DumpChan) from being written to files. * _Slightly_ better
	  support for the "light background" option (-W) was added. This
	  includes using ast_term_quit in the output of XML documentation
	  help, as well as changing the "Asterisk Ready" prompt to bright
	  green on the default background (which stands a better chance of
	  being displayed properly than bright white). Review:
	  https://reviewboard.asterisk.org/r/3547/

2014-05-28 20:53 +0000 [r414781]  Rusty Newton <rnewton@digium.com>

	* /, configs/pjsip.conf.sample: pjsip.conf: privkey_file should be
	  priv_key_file, mediaencryption=yes should be mediaencryption=sdes
	  privkey_file was missed in the snake case update. An example
	  included an invalid value for the mediaencryption option.
	  ........ Merged revisions 414780 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-28 17:46 +0000 [r414764-414766]  Matthew Jordan <mjordan@digium.com>

	* rest-api/api-docs/deviceStates.json,
	  rest-api/api-docs/endpoints.json,
	  rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
	  /, rest-api/api-docs/asterisk.json,
	  rest-api/api-docs/applications.json,
	  rest-api/api-docs/playbacks.json,
	  rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
	  rest-api/resources.json, include/asterisk/manager.h,
	  rest-api/api-docs/bridges.json,
	  rest-api/api-docs/recordings.json: AMI/ARI: Update version
	  numbers Update the semantic versioning of ARI to 1.3.0 and AMI to
	  2.3.0 to account for backwards compatible changes going from
	  12.2.0 to 12.3.0. ........ Merged revisions 414765 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* contrib/ast-db-manage/cdr/env.py, /: ast-db-manage/cdr/env.py:
	  Don't fail if a config file can't be loaded When generating SQL
	  files via the repotools alembic_creator.py script, a
	  configuration object is used programatically with SQLAlechemy, as
	  opposed to a configuration file. This patch ignores failures to
	  interpret a config file, as ... there isn't one in this case.
	  ........ Merged revisions 414763 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-28 16:56 +0000 [r414748-414750]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h, /,
	  res/res_pjsip_t38.c: res_pjsip_session: Fix leaked video RTP
	  ports. Simply enabling PJSIP to negotiage a video codec (e.g.,
	  h264) would leak video RTP ports if the codec were not negotiated
	  by an incoming call. * Made add_sdp_streams() associate the
	  handler with the media stream if the handler handled the media
	  stream. Otherwise, when the ast_sip_session_media object was
	  destroyed it didn't know how to clean up the RTP resources. *
	  Fixed sdp_requires_deferral() associating the handler with the
	  media stream when deciding if the SDP processing needs to be
	  deferred for T.38. Like the leaked video RTP ports, the T.38
	  handler needs to clean up allocated resources from deciding if
	  SDP processing needs to be deffered. * Cleaned up some dead code
	  in handle_incoming_sdp() and sdp_requires_deferral().
	  ASTERISK-23721 #close Reported by: cervajs Review:
	  https://reviewboard.asterisk.org/r/3571/ ........ Merged
	  revisions 414749 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, CHANGES, apps/app_agent_pool.c: app_agent_pool: Return to
	  dialplan if the agent fails to ack the call. Improvements to the
	  agent pool functionality. * AgentRequest no longer hangs up the
	  caller if the agent fails to connect with the caller. It now
	  continues in the dialplan. * AgentRequest returns AGENT_STATUS
	  set to NOT_CONNECTED if the agent failed to connect with the
	  call. Most likely because the agent did not acknowledge the call
	  in time or got disconnected. * The agent alerting play file
	  configured by the agent.conf custom_beep option can now be
	  disabled by setting the option to an empty string. The agent is
	  effectively alerted to a call presence when MOH stops. * Fixed
	  bridge reference leak when the agent connects with a caller.
	  ASTERISK-23499 #close Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3551/ ........ Merged
	  revisions 414747 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-28 11:37 +0000 [r414696]  Joshua Colp <jcolp@digium.com>

	* res/res_config_odbc.c, /, funcs/func_odbc.c: res_config_odbc: Use
	  dynamically sized buffers to store row data so values do not get
	  truncated. ASTERISK-23582 #close ASTERISk-23582 #comment Reported
	  by: Walter Doekes Review:
	  https://reviewboard.asterisk.org/r/3557/ ........ Merged
	  revisions 414693 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414694 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414695 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-28 09:43 +0000 [r414567-414679]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_unistim.c: chan_unistim: Unlock mutex in rare
	  OOM condition. #ASTERISK-23792 #close Reported by: Peter Whisker
	  Review: https://reviewboard.asterisk.org/r/3567/ ........ Merged
	  revisions 414677 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414678 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: chan_sip: Start session timer at 200, not
	  at INVITE. Asterisk started counting the session timer at INVITE
	  while the other end correctly started at 200. This meant that for
	  short session-expiries (90 seconds) combined with long ringing
	  times (e.g. 30 seconds), asterisk would wrongly assume that the
	  timer was hit before the other end thought it was time to send a
	  session refresh. This resulted in prematurely ended calls. This
	  changes the session timer to start counting first at 200 like RFC
	  says it should. (Also removed a few excess NULL checks that would
	  never hit, because if they did, asterisk would have crashed
	  already.) ASTERISK-22551 #close Reported by: i2045 Review:
	  https://reviewboard.asterisk.org/r/3562/ ........ Merged
	  revisions 414620 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414628 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414636 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_config_odbc.c, /: res_config_odbc: Fix old and new
	  ast_string_field memory leaks. The ODBC realtime driver uses ^NN
	  parameter encoding to cope with the special meaning of the
	  semi-colon. A semi-colon in a field is interpreted as if the key
	  was supplied twice, something which isn't otherwise possible with
	  fixed database columns. E.g. allow=alaw;ulaw is parsed as
	  allow=alaw and allow=ulaw. A literal semi-colon is rewritten to
	  ^3B when stored in the database. The module uses a stringfield to
	  efficiently store the encoded parameters. However, this
	  stringfield wasn't always freed in some off-nominal cases. Commit
	  r413241 fixed initialization so the encoding for INSERT and
	  DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked
	  apparently.) But that commit forgot the frees. This change cleans
	  that up. Review: https://reviewboard.asterisk.org/r/3555/
	  ........ Merged revisions 414564 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414565 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414566 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-25 02:37 +0000 [r414543]  Matthew Jordan <mjordan@digium.com>

	* /, main/core_unreal.c: core_unreal: Prevent double free of
	  core_unreal pvt When a channel is destroyed (such as via
	  ast_channel_release in off nominal paths in core_unreal), it will
	  attempt to free (via ast_free) the channel tech pvt. This is
	  problematic for a few reasons: 1. The channel tech pvt is an ao2
	  object in core_unreal. Free'ing the pvt directly is no good. 2.
	  The channel tech pvt's reference count is dropped just prior to
	  calling ast_channel_release, resulting in the pvt's destruction.
	  Hence, the channel destructor is free'ing an invalid pointer.
	  This patch keeps the dropping of the reference count, but sets
	  the pvt to NULL on the channel prior to releasing it. This models
	  what would occur if the channel was hung up directly. ........
	  Merged revisions 414542 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-23 17:36 +0000 [r414529]  Matthew Jordan <mjordan@digium.com>

	* tests/test_cel.c, /: test_cel: Fix unit tests broken due to event
	  def changes from res_corosync This patch instructs test_cel to
	  skip any IE types it doesn't care about. The addition of the raw
	  and bitfield types caused the tests to fail. ........ Merged
	  revisions 414528 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-23 14:36 +0000 [r414475]  Kinsey Moore <kmoore@digium.com>

	* main/event.c, /: Fix signed/unsigned build warnings ........
	  Merged revisions 414474 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-22 16:19 +0000 [r414417]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_meetme.c: app_meetme: Don't interrupt MOH for
	  waitmarked users. Occasionally, when the last marked user leaves
	  the conference, waitmarked users don't get MOH if MOH is supposed
	  to be played while a waitmarked user is waiting for another
	  marked user. * Made not interrupt MOH when the user is a
	  waitmarked user. The waitmarked user doesn't need to hear any
	  leave announcements from the conference as the user would have
	  already heard different leave announcements if they were enabled.
	  Apparently DAHDI occasionally sends unending non-silent streams
	  to these users or a normal user still in the conference has
	  continuous high background noise. These non-silent streams cause
	  MOH to be suspended while the never ending "announcement" is
	  played. Issue caused by ASTERISK-13680. AST-1349 #close Reported
	  by: Tyler Stewart Review:
	  https://reviewboard.asterisk.org/r/3543/ ........ Merged
	  revisions 414401 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414402 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414404 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-22 16:09 +0000 [r414406]  Scott Griepentrog <sgriepentrog@digium.com>

	* rest-api/api-docs/events.json, /, res/stasis/app.c,
	  res/ari/resource_events.c, include/asterisk/stasis_app.h,
	  include/asterisk/stasis.h, apps/app_userevent.c,
	  res/ari/resource_events.h, res/ari/ari_model_validators.c,
	  CHANGES, main/stasis.c, res/ari/ari_model_validators.h,
	  include/asterisk/stasis_channels.h, res/res_ari_events.c,
	  main/stasis_channels.c, res/res_stasis.c,
	  main/manager_channels.c, main/stasis_endpoints.c: ARI: Add
	  ability to raise arbitrary User Events User events can now be
	  generated from ARI. Events can be signalled with arbitrary json
	  variables, and include one or more of channel, bridge, or
	  endpoint snapshots. An application must be specified which will
	  receive the event message (other applications can subscribe to
	  it). The message will also be delivered via AMI provided a
	  channel is attached. Dialplan generated user event messages are
	  still transmitted via the channel, and will only be received by a
	  stasis application they are attached to or if the channel is
	  subscribed to. This change also introduces the multi object blob
	  mechanism used to send multiple snapshot types in a single
	  message. The dialplan app UserEvent was also changed to use multi
	  object blob, and a new stasis message type created to handle
	  them. ASTERISK-22697 #close Review:
	  https://reviewboard.asterisk.org/r/3494/ ........ Merged
	  revisions 414405 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-22 15:52 +0000 [r414403]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/bridge.h, res/parking/parking_bridge_features.c,
	  channels/chan_mgcp.c, res/res_pjsip_refer.c,
	  channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/chan_sip.c, main/parking.c, main/bridge.c,
	  main/bridge_basic.c, res/parking/parking_applications.c,
	  include/asterisk/parking.h: res_pjsip_refer: Fix bugs involving
	  Parking/PJSIP/transfers PJSIP would never send the final 200
	  Notify for a blind transfer when transferring to parking. This
	  patch fixes that. In addition, it fixes a reference leak when
	  performing blind transfers to non-bridging extensions. Review:
	  https://reviewboard.asterisk.org/r/3485/ ........ Merged
	  revisions 414400 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-22 14:02 +0000 [r414331-414348]  Matthew Jordan <mjordan@digium.com>

	* /, UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag ........
	  Merged revisions 414345 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414346 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414347 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_corosync.c, include/asterisk/stasis.h, main/app.c,
	  main/devicestate.c, main/event.c, main/stasis.c,
	  include/asterisk/devicestate.h, include/asterisk/event.h,
	  main/stasis_message.c, /, include/asterisk/event_defs.h:
	  res_corosync: Update module to work with Stasis (and compile)
	  This patch fixes res_corosync such that it works with Asterisk
	  12. This restores the functionality that was present in previous
	  versions of Asterisk, and ensures compatibility with those
	  versions by restoring the binary message format needed to pass
	  information from/to them. The following changes were made in the
	  core to support this: * The event system has been partially
	  restored. All event definition and event types in this patch were
	  pulled from Asterisk 11. Previously, we had hoped that this
	  information would live in res_corosync; however, the approach in
	  this patch seems to be better for a few reasons: (1)
	  Theoretically, ast_events can be used by any module as a binary
	  representation of a Stasis message. Given the structure of an
	  ast_event object, that information has to live in the core to be
	  used universally. For example, defining the payload of a device
	  state ast_event in res_corosync could result in an incompatible
	  device state representation in another module. (2) Much of this
	  representation already lived in the core, and was not easily
	  extensible. (3) The code already existed. :-) * Stasis message
	  types now have a message formatter that converts their payload to
	  an ast_event object. * Stasis message forwarders now handle
	  forwarding to themselves. Previously this would result in an
	  infinite recursive call. Now, this simply creates a new
	  forwarding object with no forwards set up (as it is the thing it
	  is forwarding to). This is advantageous for res_corosync, as
	  returning NULL would also imply an unrecoverable error. Returning
	  a subscription in this case allows for easier handling of message
	  types that are published directly to an aggregate topic that has
	  forwarders. Review: https://reviewboard.asterisk.org/r/3486/
	  ASTERISK-22912 #close ASTERISK-22372 #close ........ Merged
	  revisions 414330 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-21 22:24 +0000 [r414297]  Richard Mudgett <rmudgett@digium.com>

	* /, main/core_unreal.c: core_unreal: Only block media frames when
	  a generator is on both ends of an unreal channel. The fix for
	  ASTERISK-12292 was a bit too aggressive. You could have
	  generators pointed at each other on local channels but need to
	  get other kinds of frames such as DTMF or CONNECTED_LINE frames
	  accross. ........ Merged revisions 414269 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414270 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414272 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-21 19:08 +0000 [r414217]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, funcs/func_strings.c: pbx.c: prevent potential crash from
	  recursive replace() Recurisve usage of replace() resulted in
	  corruption of the temporary string storage and potential crash.
	  By changing the string to be allocated separtely per instance,
	  this is eliminated. ASTERISK-23650 #comment Reported by: Roel van
	  Meer ASTERISK-23650 #close Review:
	  https://reviewboard.asterisk.org/r/3539/ ........ Merged
	  revisions 414214 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414215 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414216 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-19 19:52 +0000 [r414196]  Paul Belanger <paul.belanger@polybeacon.com>

	* res/res_stasis_answer.c, /: Replace __ast_answer with
	  ast_raw_answer in app_control_answer While load testing an ARI
	  application, I noticed asterisk was returning HTTP 500 internal
	  server errors on channels/:id/answer. After talking to
	  #asterisk-dev, the issue appeared to be a lack of media flowing
	  after __ast_answer() was called. So now, we call ast_raw_answer
	  instead and no longer wait for media. ASTERISK-23758 #close
	  Review: https://reviewboard.asterisk.org/r/3549/ ........ Merged
	  revisions 414195 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-19 01:10 +0000 [r414123-414138]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
	  main/bridge_channel.c, res/res_pjsip_refer.c,
	  res/res_pjsip_session.c, main/channel.c, /, main/framehook.c:
	  Undo r414123 The Test Suite caught a few problems, undoing until
	  those are resolved

	* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
	  main/bridge_channel.c, res/res_pjsip_session.c, main/channel.c,
	  /, main/framehook.c: bridge_native_rtp/bridge_channel: Fix direct
	  media issues due to frame hook This patch fixes issues with
	  direct media bridges that occur after a blind transfer. These
	  issues were caught by the (currently failing)
	  pjsip/transfers/blind_transfer/caller_direct_media test. The test
	  currently fails primarily for two reasons: (1) When Bob and
	  Charlie (the transfer target and the transfer destination) enter
	  a bridge together, the framehook remains on the transfer target
	  channel until both channels are in the bridge. As it consumes
	  voice frames, the initial bridge type is a simple bridge. The
	  framehook is removed when both channels are in the bridge;
	  however, this does not currently cause the bridging framework to
	  re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE
	  poke to the transfer target channel when a framehook is removed
	  so the bridge can re-evaluate itself. (2) When a channel leaves a
	  native RTP bridge, it may be leaving due to being hung up.
	  Sending a re-INVITE to a channel that is about to be hung up is
	  not nice - in fact, there's a good chance we'll send the BYE
	  request before the channel has had a chance to send back a 200
	  OK. To be somewhat nicer, this patch adds a function to channel.h
	  that allows the bridging framework to query for exactly why a
	  channel is leaving a bridge via the channel's soft hangup flags.
	  This allows it to only send the re-INVITE if there's a chance the
	  channel will survive the native bridging experience. Review:
	  https://reviewboard.asterisk.org/r/3535/ ........ Merged
	  revisions 414122 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-16 20:06 +0000 [r413994-414070]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone
	  detection. * Check if waitingfordt (waitfordialtone) is enabled
	  in dahdi_read() to allow the DSP to operate early enough to
	  detect dialtone. * Made use the correct variable in
	  my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve
	  Davies Patches: dialtone_detect_fix (license #5012) patch
	  uploaded by Steve Davies Review:
	  https://reviewboard.asterisk.org/r/3534/ ........ Merged
	  revisions 414067 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 414068 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414069 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/sig_pri.c, /: sig_pri.c: Pull the pri_dchannel()
	  PRI_EVENT_RING case into its own function. * Populate the
	  CALLERID(ani2) value (and the special CALLINGANI2 channel
	  variable) with the ANI2 value in addition to the PRI specific
	  ANI2 channel variable. * Made complete snapshot staging with the
	  channel lock held. All channel snapshots need to be done while
	  the channel lock is held. ........ Merged revisions 414050 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 414051 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI
	  conference data structure. Starting a conference recording using
	  the admin menu overwrites the DAHDI conference data structure
	  used to modify the admin user's conference mute mode. * Made no
	  longer pass the user's DAHDI conference data structure into the
	  menu functions. The menu now uses its own DAHDI conference data
	  structure to start the recording channel. * Moved the unlock
	  conf->playlock to before playing the conf-full message. No sense
	  keeping the lock while that prompt is playing. The user is never
	  going to get into the conference at that point. ........ Merged
	  revisions 413991 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413992 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413993 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-14 15:41 +0000 [r413897]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, res/res_musiconhold.c: res_musiconhold: Minor cleanup. Fix a
	  few free()'s that should be ast_free()'s. Reverted an old
	  workaround that isn't necessary. Reorder a tiny bit of code.
	  Remove a bit of commented-out code. Review:
	  https://reviewboard.asterisk.org/r/3536/ ........ Merged
	  revisions 413894 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413895 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413896 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-13 18:09 +0000 [r413878]  Jonathan Rose <jrose@digium.com>

	* main/netsock2.c, /, channels/chan_sip.c,
	  include/asterisk/netsock2.h: chan_sip: Add TLS and SRTP status to
	  CLI command 'sip show channel' ASTERISK-23564 #close Reported by:
	  Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/
	  ........ Merged revisions 413876 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413877 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-13 13:53 +0000 [r413790-413793]  Walter Doekes <walter+asterisk@wjd.nu>

	* res/res_format_attr_h264.c, /: h264: Fix H264 SDP payload format.
	  https://tools.ietf.org/html/rfc3984#section-8.1 says
	  profile-level-id takes 3 bytes in base16 (6 hex digits). This
	  fixes video setup in certain cases. ASTERISK-23664 #close
	  ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume
	  Maudoux. Review: https://reviewboard.asterisk.org/r/3530/
	  ........ Merged revisions 413791 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413792 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/rtp_engine.c: rtp: Fix case typo in H263+ mime.
	  http://tools.ietf.org/html/rfc3555#section-4.2.6 says the
	  canonical mime subtype is "H263-1998", not "h263-1998". Original
	  code was added in r183101 on 2009-03-19 02:26:50 +0100. This
	  fixes issues with Polycom phones. ASTERISK-23665 #close
	  ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume
	  Maudoux, backported by me. Review:
	  https://reviewboard.asterisk.org/r/3529/ ........ Merged
	  revisions 413787 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413788 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413789 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-13 00:35 +0000 [r413770-413772]  Richard Mudgett <rmudgett@digium.com>

	* configure.ac, channels/sig_pri.c, /, configure,
	  include/asterisk/autoconfig.h.in: chan_dahdi/sig_pri: Prevent
	  unnecessary PROGRESS events when overlap dialing is enabled. When
	  overlap dialing is enabled, the lack of inband audio available
	  information in the SETUP_ACKNOWLEDGE events causes an
	  interoperability problem with SIP. sig_pri doesn't know if there
	  is dialtone present when a SETUP_ACKNOWLEDGE is received so it
	  assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
	  SIP channel driver then sends out a 183 Session Progress and
	  blocks the desired 180 Ringing message when the ALERTING message
	  comes in. * Made the configure script detect if the installed
	  version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
	  Using the new API, made generate an AST_CONTROL_PROGRESS frame on
	  an incoming SETUP_ACKNOWLEDGE message when the message indicates
	  inband audio is present instead of assuming that dialtone is
	  present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
	  inband audio available indication only if dialtone is expected.
	  The change also makes the fallback behaviour of sending the
	  PROGRESS message better by sending it only if dialtone is
	  expected. * Changed receiving a PROCEEDING message to not
	  generate an AST_CONTROL_PROGRESS frame if the progress indication
	  ie indicates non-end-to-end-ISDN. This helps interoperability
	  with SIP. * Changed sending a PROCEEDING message in response to
	  an AST_CONTROL_PROCEEDING frame to not indicate inband audio
	  available. It was silly to do so anyway because the channel
	  driver doesn't know if inband audio is even available. This helps
	  interoperability with SIP. This patch and a corresponding change
	  in libpri work together to allow Asterisk to control the inband
	  audio available progress indication ie on the SETUP_ACKNOWLEDGE
	  message when dialtone is present. AST-1338 #close Reported by:
	  Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
	  ........ Merged revisions 413714 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413765 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413771 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/sig_pri.c: Fix compiler warning from GCC 4.10 fixup.
	  ........ Merged revisions 413766 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-12 22:33 +0000 [r413713]  Jonathan Rose <jrose@digium.com>

	* apps/app_chanspy.c, /: app_chanspy: Fix a test that was failing
	  on account of r413551 ASTERISK-23381 #close ASTERISK-23381
	  #comment Reported by: Robert Moss Review:
	  https://reviewboard.asterisk.org/r/3505/ ........ Merged
	  revisions 413710 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413712 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-11 02:09 +0000 [r413651-413682]  Joshua Colp <jcolp@digium.com>

	* main/bridge_basic.c, include/asterisk/channel.h,
	  bridges/bridge_native_rtp.c, include/asterisk/framehook.h,
	  main/channel.c, /, main/framehook.c: framehooks: Add callback for
	  determining if a hook is consuming frames of a specific type. In
	  the past framehooks have had no capability to determine what
	  frame types a hook is actually interested in consuming. This has
	  meant that code has had to assume they want all frames, thus
	  preventing native bridging. This change adds a callback which
	  allows a framehook to be queried for whether it is consuming a
	  frame of a specific type. The native RTP bridging module has also
	  been updated to take advantange of this, allowing native bridging
	  to occur when previously it would not. ASTERISK-23497 #comment
	  Reported by: Etienne Lessard ASTERISK-23497 #close Review:
	  https://reviewboard.asterisk.org/r/3522/ ........ Merged
	  revisions 413681 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
	  include/asterisk/framehook.h, main/channel.c, /,
	  main/framehook.c, main/bridge_basic.c: Undoing framehook support.
	  Issues were uncovered by Bamboo.

	* /, main/framehook.c, main/bridge_basic.c,
	  include/asterisk/channel.h, bridges/bridge_native_rtp.c,
	  include/asterisk/framehook.h, main/channel.c: framehooks: Add
	  callback for determining if a hook is consuming frames of a
	  specific type. In the past framehooks have had no capability to
	  determine what frame types a hook is actually interested in
	  consuming. This has meant that code has had to assume they want
	  all frames, thus preventing native bridging. This change adds a
	  callback which allows a framehook to be queried for whether it is
	  consuming a frame of a specific type. The native RTP bridging
	  module has also been updated to take advantange of this, allowing
	  native bridging to occur when previously it would not.
	  ASTERISK-23497 #comment Reported by: Etienne Lessard
	  ASTERISK-23497 #close Review:
	  https://reviewboard.asterisk.org/r/3522/ ........ Merged
	  revisions 413650 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-09 23:18 +0000 [r413589-413599]  Kinsey Moore <kmoore@digium.com>

	* /, funcs/func_env.c: Fix 32bit build for func_env ........ Merged
	  revisions 413592 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413595 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413597 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_festival.c, pbx/dundi-parser.c, apps/app_getcpeid.c,
	  main/netsock.c, funcs/func_channel.c, main/audiohook.c,
	  pbx/pbx_config.c, res/res_pjsip_registrar.c, main/xmldoc.c,
	  channels/iax2/firmware.c, apps/app_voicemail.c, main/format.c,
	  cel/cel_pgsql.c, main/rtp_engine.c, main/parking.c,
	  main/bridge.c, res/res_jabber.c, res/res_http_websocket.c,
	  main/config.c, res/res_format_attr_opus.c, main/loader.c,
	  res/parking/parking_bridge.c, main/cdr.c, main/manager.c,
	  include/asterisk/astobj.h, main/bucket.c, apps/app_dumpchan.c,
	  main/app.c, res/res_pjsip/config_transport.c,
	  res/res_pjsip_refer.c, channels/chan_mgcp.c,
	  res/res_rtp_asterisk.c, main/slinfactory.c, main/core_unreal.c,
	  res/res_pjsip_sdp_rtp.c, res/res_crypto.c, main/acl.c,
	  channels/sig_pri.c, res/res_monitor.c, res/res_srtp.c,
	  main/data.c, res/res_corosync.c, channels/sip/config_parser.c,
	  res/res_fax_spandsp.c, apps/app_stack.c, main/asterisk.c,
	  main/udptl.c, res/res_sorcery_config.c, main/security_events.c,
	  res/res_timing_dahdi.c, res/res_pjsip_t38.c,
	  res/res_musiconhold.c, main/taskprocessor.c,
	  res/res_format_attr_h263.c, res/res_xmpp.c, res/res_pktccops.c,
	  funcs/func_hangupcause.c, channels/chan_phone.c,
	  main/manager_bridges.c, cel/cel_odbc.c, channels/chan_skinny.c,
	  channels/chan_motif.c, res/res_agi.c, main/logger.c,
	  funcs/func_srv.c, channels/chan_alsa.c, apps/app_confbridge.c,
	  res/res_pjsip_pubsub.c, channels/sip/include/sip.h, main/sched.c,
	  apps/app_adsiprog.c, main/pbx.c, channels/chan_sip.c,
	  res/res_fax.c, main/aoc.c, res/res_calendar_ews.c,
	  res/parking/parking_bridge_features.c, channels/iax2/parser.c,
	  main/callerid.c, main/file.c,
	  res/res_pjsip/pjsip_configuration.c, main/adsi.c,
	  main/config_options.c, pbx/pbx_dundi.c, funcs/func_iconv.c,
	  main/bridge_channel.c, res/res_odbc.c, channels/chan_pjsip.c,
	  res/parking/parking_manager.c, res/res_calendar.c, /,
	  funcs/func_sysinfo.c, main/utils.c, cdr/cdr_adaptive_odbc.c,
	  res/res_calendar_caldav.c, res/res_stasis_snoop.c,
	  res/res_format_attr_h264.c, main/channel.c, res/ael/pval.c,
	  res/res_ari_model.c, channels/chan_dahdi.c,
	  channels/sig_analog.c, funcs/func_frame_trace.c,
	  res/res_format_attr_silk.c, main/manager_channels.c,
	  apps/app_dial.c, res/res_calendar_icalendar.c, main/translate.c,
	  apps/app_queue.c, channels/chan_jingle.c, res/res_stun_monitor.c,
	  main/abstract_jb.c, res/res_stasis_recording.c, apps/app_sms.c,
	  main/event.c, apps/app_verbose.c, main/dsp.c,
	  channels/chan_unistim.c, main/frame.c, res/res_stasis_playback.c,
	  main/ccss.c, funcs/func_env.c, main/devicestate.c,
	  bridges/bridge_softmix.c, channels/chan_gtalk.c,
	  channels/chan_iax2.c, main/enum.c, main/cli.c,
	  res/res_format_attr_celt.c, apps/confbridge/conf_config_parser.c,
	  main/io.c, channels/pjsip/dialplan_functions.c,
	  res/res_config_odbc.c, res/res_pjsip/location.c,
	  res/res_pjsip_outbound_registration.c, formats/format_pcm.c,
	  apps/app_minivm.c, main/stdtime/localtime.c, main/stun.c: Allow
	  Asterisk to compile under GCC 4.10 This resolves a large number
	  of compiler warnings from GCC 4.10 which cause the build to fail
	  under dev mode. The vast majority are signed/unsigned mismatches
	  in printf-style format strings. ........ Merged revisions 413586
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 413587 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413588 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-09 18:15 +0000 [r413572]  Richard Mudgett <rmudgett@digium.com>

	* main/http.c: http.c: Remove dead code.

2014-05-09 17:03 +0000 [r413557]  Jonathan Rose <jrose@digium.com>

	* apps/app_chanspy.c, /: app_chanspy: Fix a bug where Barge mode
	  could fail If the barge audiohook was attached prior to the spyee
	  and its peer actually being bridged, the audiohook would not be
	  applied and the connected peer would not be able to hear audio
	  from the spy when the spy is in barge mode. (closes issue
	  ASTERISK-23381) Reported by: Robert Moss Review:
	  https://reviewboard.asterisk.org/r/3505/ ........ Merged
	  revisions 413551 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413556 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-08 00:36 +0000 [r413488]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c, main/manager.c, /: app_queue: Extend
	  documentation for various Manager actions and events. ........
	  Merged revisions 413485 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413486 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413487 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-07 21:58 +0000 [r413469]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_presencestate.c: Ensure that presence state is decoded
	  properly on Asterisk startup. The CustomPresence provider
	  callback will automatically base64 decode stored data if the 'e'
	  option was present when the state was set. However, since the
	  provider callback was bypassed on Asterisk startup, encoded
	  presence subtypes and messages were being sent instead. This fix
	  makes it so the provider callback is always used when providing
	  presence state updates.

2014-05-07 20:59 +0000 [r413453-413455]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_confbridge.c, /: app_confbridge: Fixed "CBAnn" channels
	  not going away. Fixed a ref leak in conf_handle_talker_cb()
	  everytime the conference bridge was found to report a channel's
	  talker status change. The resulting leak caused the "CBAnn"
	  channels and the conference bridge to never be destroyed. Thanks
	  to Richard Kenner on the asterisk-user's list for locating the
	  problem. Reported by: Richard Kenner ........ Merged revisions
	  413454 from http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_confbridge.c, /: app_confbridge: Fix ref leak in CLI
	  "confbridge kick" command. Fixed ref leak in the CLI "confbridge
	  kick" command when the channel to be kicked was not in the
	  conference. ........ Merged revisions 413451 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413452 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-07 17:56 +0000 [r413307-413399]  Mark Michelson <mmichelson@digium.com>

	* res/res_config_odbc.c, /: Fix encoding of custom prepare extra
	  data. Patches: res_config_odbc-take2.patch by John Hardin
	  (License #6512) ........ Merged revisions 413396 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413397 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413398 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip/presence_xml.c, /,
	  res/res_pjsip_pidf_digium_body_supplement.c: Improve XML
	  sanitization in NOTIFYs, especially for presence subtypes and
	  messages. Embedded carriage return line feed combinations may
	  appear in presence subtypes and messages since they may be
	  derived from user input in an instant messenger client. As such,
	  they need to be properly escaped so that XML parsers do not vomit
	  when the messages are received. ........ Merged revisions 413372
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_registrar.c, /: Check for an act on failures to
	  update contacts during registration. There was an underlying
	  issue in a realtime backend where database updates would fail.
	  Since we were not checking for failure, we would end up in a
	  strange state where the old database entry was still present but
	  Asterisk thought that it had been updated. Now when an entry
	  fails to update, we print a warning and delete the old contact
	  from sorcery so there is no mismatch between foreground and
	  backend state. Patches: res_pjsip_registrar.patch by John Hardin
	  (License #6512) ........ Merged revisions 413358 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_config_odbc.c, /: Ensure that all parts of SQL UPDATEs
	  and DELETEs are encoded. Patches: res_config_odbc.patch by John
	  Hardin (License #6512) ........ Merged revisions 413304 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413305 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413306 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-02 20:28 +0000 [r413227-413263]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_config_odbc.c: Prevent crashes in res_config_odbc due
	  to uninitialized string fields. Patches: odbc-crash.patch by John
	  Hardin (License #6512) ........ Merged revisions 413241 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413251 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413258 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_config_pgsql.c, /: Return the number of rows affected by
	  a SQL insert, rather than an object ID. The realtime API
	  specifies that the store callback is supposed to return the
	  number of rows affected. res_config_pgsql was instead returning
	  an Oid cast as an int, which during any nominal execution would
	  be cast to 0. Returning 0 when more than 0 rows were inserted
	  causes problems to the function's callers. To give an idea of how
	  strange code can be, this is the necessary code change to fix a
	  device state issue reported against chan_pjsip in Asterisk 12+.
	  The issue was that the registrar would attempt to insert contacts
	  into the database. Because of the 0 return from res_config_pgsql,
	  the registrar would think that the contact was not successfully
	  inserted, even though it actually was. As such, even though the
	  contact was query-able and it was possible to call the endpoint,
	  Asterisk would "think" the endpoint was unregistered, meaning it
	  would report the device state as UNAVAILABLE instead of
	  NOT_INUSE. The necessary fix applies to all versions of Asterisk,
	  so even though the bug reported only applies to Asterisk 12+, the
	  code correction is being inserted into 1.8+. Closes issue
	  ASTERISK-23707 Reported by Mark Michelson ........ Merged
	  revisions 413224 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 413225 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413226 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-02 16:39 +0000 [r413211]  Richard Mudgett <rmudgett@digium.com>

	* UPGRADE.txt, res/res_pjsip_refer.c, /, channels/chan_sip.c:
	  res_pjsip_refer: Add Referred-By header on INVITE for blind
	  transfers. Per rfc3892, the Referred-By header in a REFER must be
	  copied into the referenced request (IE. The outgoing INVITE to
	  the transfer target). * Automatically put the Referred-By header
	  in the outgoing INVITE message if the SIPREFERREDBYHDR channel
	  variable is defined with a value. * Made
	  chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance
	  so chan_pjsip has a better chance to interoperate. * Fixed
	  refer_blind_callback() and refer_incoming_refer_request() to not
	  modify the data in the pointer returned by
	  pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data
	  since the calling routine doesn't own the buffer. ASTERISK-23501
	  #close Reported by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/3514/ ........ Merged
	  revisions 413210 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-02 16:06 +0000 [r413197]  Jonathan Rose <jrose@digium.com>

	* res/parking/res_parking.h, /, CHANGES,
	  res/parking/parking_bridge_features.c,
	  res/parking/parking_manager.c: Parking: Add 'AnnounceChannel'
	  argument to manager action 'Park' (closes ASTERISK-23397)
	  Reported by: Denis Review:
	  https://reviewboard.asterisk.org/r/3446/ ........ Merged
	  revisions 413196 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-01 16:21 +0000 [r413174-413183]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_presencestate.c: Make behavior of the PRESENCE_STATE
	  'e' option more consistent. When writing presence state, if 'e'
	  is specified, then the presence state will be stored in the astdb
	  encoded. However, consumers of presence state events or those
	  that query for the presence state will be given decoded
	  information. If base64 encoding is desired for consumers, then
	  the information can be base64-encoded manually and the 'e' option
	  can be omitted. closes issue ASTERISK-23671 Reported by Mark
	  Michelson Review: https://reviewboard.asterisk.org/r/3482

	* res/res_pjsip_exten_state.c, /: Remove unnecessary repetition
	  checks from res_pjsip_exten_state The PBX core already takes care
	  of ensuring that repeated state changes are not communicated to
	  exten state consumers. Because the check in res_pjsip_exten_state
	  was incomplete, it was causing valid presence state changes not
	  to be sent out. For instance, if the presence state did not
	  change but the message or subtype did, then no presence-related
	  NOTIFY request would be sent out. closes issue ASTERISK-23672
	  Reported by Mark Michelson ........ Merged revisions 413173 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-05-01 12:31 +0000 [r413160]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip/config_transport.c, /: res_pjsip: Add the ability
	  to configure ciphers based on name. Previously this code would
	  only accept the OpenSSL identifier instead of the documented
	  name. ASTERISK-23498 #close ASTERISK-23498 #comment Reported by:
	  Anthony Messina Review: https://reviewboard.asterisk.org/r/3491/
	  ........ Merged revisions 413159 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-30 21:03 +0000 [r413144]  Richard Mudgett <rmudgett@digium.com>

	* main/message.c, /, channels/chan_sip.c,
	  include/asterisk/message.h, res/res_pjsip_messaging.c:
	  chan_sip.c: Fixed off-nominal message iterator ref count and
	  alloc fail issues. * Fixed early exit in sip_msg_send() not
	  destroying the message iterator. * Made
	  ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
	  tolerant of a NULL iter parameter in case
	  ast_msg_var_iterator_init() fails. * Made
	  ast_msg_var_iterator_destroy() clean up any current message data
	  ref. * Made struct ast_msg_var_iterator,
	  ast_msg_var_iterator_init(), ast_msg_var_iterator_next(),
	  ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy()
	  use iter instead of i. * Eliminated RAII_VAR usage in
	  res_pjsip_messaging.c:vars_to_headers(). ........ Merged
	  revisions 413139 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413142 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-30 20:39 +0000 [r413141]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_pjsip.c: chan_pjsip: Fix deadlock when
	  retrieving call-id of channel. If a task was in-flight which
	  required the channel or bridge lock it was possible for the
	  synchronous task retrieving the call-id to deadlock as it holds
	  those locks. After discussing with Mark Michelson the synchronous
	  task was removed and the call-id accessed directly. This should
	  be safe as each object involved is guaranteed to exist and the
	  call-id will never change. ........ Merged revisions 413140 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-30 13:08 +0000 [r413125]  Kinsey Moore <kmoore@digium.com>

	* res/res_http_websocket.c, /: Websocket: Add session locking and
	  delay close This resolves a race condition where data could be
	  written to a NULL FILE pointer causing a crash as a websocket
	  connection was in the process of shutting down by adding locking
	  to websocket session writes and by deferring session teardown
	  until session destruction. (closes issue ASTERISK-23605) Review:
	  https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan
	  ........ Merged revisions 413123 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413124 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-30 12:42 +0000 [r413118-413122]  Joshua Colp <jcolp@digium.com>

	* /, res/stasis/control.c: res_stasis: Add progress indications to
	  operations which perform media. This change fixes operations
	  which did not account for the fact that they may be executed on
	  channels which have not been answered. These operations will now
	  indicate progress when invoked. ASTERISK-23560 #close
	  ASTERISk-23560 #comment Reported by: Jan Svoboda Review:
	  https://reviewboard.asterisk.org/r/3495/ ........ Merged
	  revisions 413121 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix issue where
	  sending a hold SDP twice could cause an unhold. This change fixes
	  a bug where if an SDP with media address and sendonly was
	  received twice the underlying call would go off hold, instead of
	  remaining on hold. This occured because the code did not properly
	  take into account that the SDP may contain both a valid media
	  address and the sendonly attribute. The code now examines the
	  sendonly attribute and media address first, so if the SDP is
	  received again no change will occur. ASTERISK-23558 #comment
	  Reported by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/3472/ ........ Merged
	  revisions 413119 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip:
	  Add support for picking up calls in the configured pickup group.
	  AST-1363 Review: https://reviewboard.asterisk.org/r/3478/
	  ........ Merged revisions 413117 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-29 15:10 +0000 [r413103]  George Joseph <george.joseph@fairview5.com>

	* /, include/asterisk/spinlock.h: Add "destroy" implementation for
	  spinlock. The original commit for spinlock was missing "destroy"
	  implementations. Most of them are no-ops but phtread_spin and
	  pthread_mutex do need their locks destroyed. ........ Merged
	  revisions 413102 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-29 11:27 +0000 [r413089]  Joshua Colp <jcolp@digium.com>

	* channels/chan_pjsip.c, /: chan_pjsip: Implement core ability to
	  get Call-ID of a channel. This changes implement the
	  "get_pvt_uniqueid" which is used to return the technology
	  specific unique identifier. In the case of SIP this is the
	  Call-ID of the dialog. Review:
	  https://reviewboard.asterisk.org/r/3480/ ........ Merged
	  revisions 413088 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-28 20:07 +0000 [r413074]  Kinsey Moore <kmoore@digium.com>

	* /, main/bridge.c, main/bridge_basic.c: Bridging: Don't lock NULL
	  bridges When bridge locking was added for bridge snapshot
	  creation, some locations where bridge locking was added were not
	  guaranteed to actually have a bridge and locking NULL AO2 objects
	  tends to cause segfaults. This ensures that NULL bridges aren't
	  locked. ........ Merged revisions 413073 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-28 14:40 +0000 [r413060]  Mark Michelson <mmichelson@digium.com>

	* res/res_manager_presencestate.c (added), main/devicestate.c,
	  CHANGES, main/presencestate.c, res/res_manager_devicestate.c
	  (added): Add DeviceStateChanged and PresenceStateChanged AMI
	  events. These events are controlled by two new modules,
	  res_manager_devicestate and res_manager_presencestate. Review:
	  https://reviewboard.asterisk.org/r/3417

2014-04-28 07:43 +0000 [r413048]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* UPGRADE.txt, CHANGES, channels/chan_unistim.c,
	  configs/unistim.conf.sample: Introducing changes proposed to
	  chan_unistim driver: 1) Added the unistim.conf variable
	  dtmf_duration which can select the DTMF playback duration from
	  0ms to 150ms (0 is off and is the new default) 2) Enabled the
	  transmission of month names, which are sent with the date and
	  changed the dateformat variable to accept the values 0-3 as per
	  the UNISTIM standard (2 & 3 match the previous 1 & 2 formats). 3)
	  Enabled the "Mute" packet so muting microphone works as expected
	  and microphone muted for all calls while LED light on 4) Changed
	  Duree to Timer on i2004 display (closes issue ASTERISK-23592)

2014-04-27 19:29 +0000 [r413036]  Olle Johansson <oej@edvina.net>

	* main/tcptls.c: tcptls.c : Log errors as ERROR, not warning or
	  something else.

2014-04-25 19:26 +0000 [r413012]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Add support for DTLS
	  handshake retransmissions On congested networks, it is possible
	  for the DTLS handshake messages to get lost. This patch adds a
	  timer to res_rtp_asterisk that will periodically check to see if
	  the handshake has succeeded. If not, it will retransmit the DTLS
	  handshake. Review: https://reviewboard.asterisk.org/r/3337
	  ASTERISK-23649 #close Reported by: Nitesh Bansal patches:
	  dtls_retransmission.patch uploaded by Nitesh Bansal (License
	  6418) ........ Merged revisions 413008 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 413009 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-24 14:37 +0000 [r412993]  Kevin Harwell <kharwell@digium.com>

	* /,
	  contrib/ast-db-manage/config/versions/e96a0b8071c_increase_pjsip_column_size.py
	  (added): pjsip realtime: increase the size of some columns The
	  string lengths on certain columns created through alembic for
	  PJSIP were too short. For instance, columns containing URIs are
	  currently set to 40 characters, but this can be too small and
	  result in truncated values. Added an alembic migration script
	  that increases the size of these columns and a few others to 255.
	  ASTERISK-23639 #close Reported by: Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/3475/ ........ Merged
	  revisions 412992 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-23 20:13 +0000 [r412977]  George Joseph <george.joseph@fairview5.com>

	* include/asterisk/spinlock.h (added), /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: This patch adds
	  support for spinlocks in Asterisk. There are cases in Asterisk
	  where it might be desirable to lock a short critical code section
	  but not incur the context switch and yield penalty of a mutex or
	  rwlock. The primary spinlock implementations execute exclusively
	  in userspace and therefore don't incur those penalties. Spinlocks
	  are NOT meant to be a general replacement for mutexes. They
	  should be used only for protecting short blocks of critical code
	  such as simple compares and assignments. Operations that may
	  block, hold a lock, or cause the thread to give up it's timeslice
	  should NEVER be attempted in a spinlock. The first use case for
	  spinlocks is in astobj2 - internal_ao2_ref. Currently the
	  manipulation of the reference counter is done with an
	  ast_atomic_fetchadd_int which works fine. When weak reference
	  containers are introduced however, there's an additional
	  comparison and assignment that'll need to be done while the lock
	  is held. A mutex would be way too expensive here, hence the
	  spinlock. Given that lock contention in this situation would be
	  infrequent, the overhead of the spinlock is only a few more
	  machine instructions than the current ast_atomic_fetchadd_int
	  call. ASTERISK-23553 #close Review:
	  https://reviewboard.asterisk.org/r/3405/ ........ Merged
	  revisions 412976 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-23 18:03 +0000 [r412925]  Richard Mudgett <rmudgett@digium.com>

	* /, main/http.c: http: Fix spurious ERROR message in responses
	  with no content. Backport -r411687 and fix the fix because
	  content_length is the length of out plus the length of the file
	  controlled by fd. When a response has an out content length of 0,
	  fwrite would be called to write a buffer with no data in it. This
	  resulted in the following classic error message: [Apr 3 11:49:17]
	  ERROR[26421] http.c: fwrite() failed: Success This patch makes it
	  so that we only attempt to write the content of out if the out
	  string is non-zero. ........ Merged revisions 412922 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 412923 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412924 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-23 15:02 +0000 [r412910]  Russell Bryant <russell@russellbryant.com>

	* res/res_monitor.c, funcs/func_periodic_hook.exports.in (added),
	  main/asterisk.dynamics, funcs/func_periodic_hook.c: Fix error
	  loading res_monitor. For some odd reason, loading app_mixmonitor
	  was fine, but res_monitor was not. This patch fixes a set of
	  issues related to func_periodic_hook exporting the beep functions
	  that gets res_monitor working again.

2014-04-22 10:09 +0000 [r412883]  Joshua Colp <jcolp@digium.com>

	* /, res/stasis/app.c: res_stasis: Fix crash when handling a failed
	  blind transfer message. This changes fixes a crash that occurs
	  when stasis determines if it should send a message out to an
	  application or not. The code incorrectly assumed that a bridge
	  snapshot would always be present when in reality for failure
	  cases it may not be. ASTERISK-23573 #close ........ Merged
	  revisions 412882 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-21 17:56 +0000 [r412759-412824]  Jonathan Rose <jrose@digium.com>

	* CHANGES, /: chan_sip: trust_id_outbound CHANGES message
	  improvement (closes issue AST-1301) (closes issue ASTERISK-19465)
	  Reported by: Krzysztof Chmielewski ........ Merged revisions
	  412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 412822 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412823 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
	  channels/sip/include/sip.h: chan_sip: Add sendrpid trust options
	  In r411189, some behavior was changed which made sendrpid
	  behavior act in a more trusting manner by sending full user data
	  for peers set with private caller presence in P-Asserted-Identity
	  headers. Since this changed long time expected behaviors, we
	  decided to pull that patch when that was pointed out by the
	  community. Instead, this patch provides a trust_id_outbound
	  setting which will expose the data per RFC-3325 if set to 'yes'
	  and simply not send the PAI/RPID headers at all if set to 'no'.
	  By default trust_id_outbound will be set to 'legacy' which will
	  preserve the behavior prior to these patches. Extra special
	  thanks to Walter Doekes for providing advice and feedback.
	  (closes issue AST-1301) (closes issue ASTERISK-19465) Reported
	  by: Krzysztof Chmielewski Review:
	  https://reviewboard.asterisk.org/r/3447/ ........ Merged
	  revisions 412744 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 412746 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412747 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-21 16:16 +0000 [r412729-412750]  Kinsey Moore <kmoore@digium.com>

	* main/http.c, main/manager.c, /: HTTP: Add TCP_NODELAY to accepted
	  connections This adds the TCP_NODELAY option to accepted
	  connections on the HTTP server built into Asterisk. This option
	  disables the Nagle algorithm which controls queueing of outbound
	  data and in some cases can cause delays on receipt of response by
	  the client due to how the Nagle algorithm interacts with TCP
	  delayed ACK. This option is already set on all non-HTTP AMI
	  connections and this change would cover standard HTTP requests,
	  manager HTTP connections, and ARI HTTP requests and websockets in
	  Asterisk 12+ along with any future use of the HTTP server.
	  Review: https://reviewboard.asterisk.org/r/3466/ ........ Merged
	  revisions 412745 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 412748 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412749 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_confbridge.c, /: Confbridge: Fix ConfbridgeKick AMI
	  documentation This adds documentation for the "all" channel
	  option for the ConfbridgeKick AMI action and adjusts AMI
	  responses accordingly. (issue ASTERISK-23282) Reported by: Dorian
	  Logan ........ Merged revisions 412730 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, apps/app_confbridge.c: Confbridge: Add references for kick all
	  option After the ability to kick all attendees from a conference
	  was added, a rework removed the comment about that feature from
	  the CLI documentation. This adds that documentation and adds
	  "all" to the participant tab completion list for the confbridge
	  kick command. (closes issue ASTERISK-23282) Reported by: Dorian
	  Logan ........ Merged revisions 412728 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-21 08:36 +0000 [r412714]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* /, channels/chan_unistim.c: Fix wrong dialtone. The "modulation"
	  should not be referenced for tone+tone as it refers to the on-off
	  characteristic - this often resulted in a single tone rather than
	  the multitone as in the UK. ........ Merged revisions 412712 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412713 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-19 02:14 +0000 [r412697-412699]  Matthew Jordan <mjordan@digium.com>

	* /, main/asterisk.c: main/asterisk: Fix startup sequence for
	  realtime features When ASTERISK-23265/ASTERISK-23320 was fixed,
	  it inadvertently led to realtime features breaking. This was due
	  to features loading prior to realtime. This patch fixes this by
	  loading features after loading dynamic modules. ASTERISK-23487
	  #close Reported by: Denis Tested by: Denis ........ Merged
	  revisions 412698 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, apps/app_sms.c: app_sms: Fix uninitialized values; hangup
	  channel when REL is sent successfully This patch fixes two issues
	  in app_sms: (1) Firstly, the 'flags' field on the stack in
	  sms_exec() is uninitialised, causing it to use the wrong protocol
	  in some cases. This patch correctly initializes the flags fields.
	  (2) Secondly, when disconnect supervision is not working or
	  inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was
	  failing to terminate the call after it sent the REL(ease) message
	  and the peer stopped talking to it. This patch fixes the code to
	  handle the 'bad stop bit' message more gracefully in that case,
	  and hang up the call. Review:
	  https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close
	  Reported by: David Woodhouse patches: asterisk-fix-sms.patch
	  uploaded by David Woodhouse (License 5754) ........ Merged
	  revisions 412655 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 412656 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412657 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-18 20:09 +0000 [r412641]  Jonathan Rose <jrose@digium.com>

	* /, res/ari/resource_bridges.h, res/stasis/control.c,
	  include/asterisk/stasis_app.h, res/stasis/control.h,
	  res/ari/resource_channels.c, CHANGES, res/res_stasis.c,
	  rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
	  res/res_ari_bridges.c, res/res_stasis_playback.c: ARI: Make
	  bridges/{bridgeID}/play queue sound files Previously multiple
	  play actions against a bridge at one time would cause the sounds
	  to play simultaneously on the bridge. Now if a sound is already
	  playing, the play action will queue playback to occur after the
	  completion of other sounds currently on the queue. (closes issue
	  ASTERISK-22677) Reported by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/3379/ ........ Merged
	  revisions 412639 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-18 17:17 +0000 [r412589]  Rusty Newton <rnewton@digium.com>

	* sounds/sounds.xml, sounds/Makefile, /: sounds: Fix Sounds
	  Makefile and XML that didn't support new sound prompt sets In
	  sounds/Makefile 1 Adds and moves some lines necessary for the
	  en_GB core set. I'm just following how the other sets are defined
	  here. 2 removes the ES extra sounds related lines as we don't
	  have ES extra sound sets. In sounds/sounds.xml 3 Adds member
	  definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in
	  extra sound sets ASTERISK-23550 #close Review:
	  https://reviewboard.asterisk.org/r/3464/ ........ Merged
	  revisions 412586 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412587 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-18 17:02 +0000 [r412584]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip/location.c: Allow for multiple contacts to be
	  configured in a single contact= line. This is useful for
	  configuring multiple permanent contacts for an AOR when using
	  realtime AORs. Review: https://reviewboard.asterisk.org/r/3462
	  ........ Merged revisions 412582 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-18 16:44 +0000 [r412580-412583]  Richard Mudgett <rmudgett@digium.com>

	* main/dial.c, main/pbx.c, /, apps/app_originate.c,
	  include/asterisk/pbx.h: Originated calls: Fix several originate
	  call problems. * Restore the reason value set by
	  pbx_outgoing_attempt() to use AST_CONTROL_xxx values as all the
	  consumers were expecting rather than cause codes. * Fixed the
	  dial routines to set cause codes for more than just ast_request()
	  so pbx_outgoing_attempt() reason codes will function. * Fix
	  inconsistent locked_channel return status in
	  pbx_outgoing_attempt(). The chanel may not have been locked or
	  the channel may have been a stale pointer. * Fixed the
	  OutgoingSpoolFailed channel to run dialplan whenever the dialing
	  fails for an originate exten and 1 < synchronous. * Fix incorrect
	  ast_cond_wait() usage in pbx_outgoing_attempt(). Indroduced by
	  issue ASTERISK-22212 patch. * Made struct pbx_outgoing use the
	  ao2 lock instead of its own lock for the cond wait mutex. No
	  sense in having two locks associated with the same struct when
	  only one is needed. Review:
	  https://reviewboard.asterisk.org/r/3421/ ........ Merged
	  revisions 412581 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis_channels.c, apps/app_queue.c, apps/app_dial.c, /:
	  app_dial and app_queue: Make lock the forwarding channel while
	  taking the channel snapshot. * Fixed
	  ast_channel_publish_dial_forward() not locking the forwarded
	  channel when taking the channel snapshot. * Fixed
	  app_dial.c:do_forward() using the wrong channel to get the
	  original call forwarding string. * Removed unnecessary locking
	  when calling ast_channel_publish_dial() and
	  ast_channel_publish_dial_forward() in app_dial and app_queue.
	  Holding channel locks when calling
	  ast_channel_publish_dial_forward() with a forwarded channel could
	  result in pausing the system while the stasis bus completes
	  processsing a forwarded channel subscription. Review:
	  https://reviewboard.asterisk.org/r/3451/ ........ Merged
	  revisions 412579 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-18 14:25 +0000 [r412566]  Kinsey Moore <kmoore@digium.com>

	* res/ari/ari_websockets.c, res/res_ari.c, main/manager.c, /: ARI:
	  Add debug logging for events and responses This adds DEBUG level
	  logging for ARI websocket events and HTTP responses similar to
	  what is available for AMI. Logging for ARI HTTP requests is
	  already adequate for debugging purposes. ........ Merged
	  revisions 412565 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-17 22:50 +0000 [r412552]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip/location.c, res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
	  res/res_pjsip_registrar.c: res_pjsip: Handle reloading when
	  permanent contacts exist and qualify is configured. This change
	  fixes a problem where permanent contacts being qualified were not
	  being updated. This was caused by the permanent contacts getting
	  a uuid and not a known identifier, causing an inability to look
	  them up when updating in the qualify code. A bug also existed
	  where the new configuration may not be available immediately when
	  updating qualifies. (closes issue ASTERISK-23514) Reported by:
	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/3448/
	  ........ Merged revisions 412551 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-17 22:42 +0000 [r412536-412550]  Jonathan Rose <jrose@digium.com>

	* /, main/app.c: Fix a silly shadowed variable mistake that was
	  missed from play tones patch ........ Merged revisions 412549
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/ari/resource_bridges.h, main/app.c,
	  rest-api/api-docs/channels.json, CHANGES,
	  rest-api/api-docs/bridges.json, res/ari/resource_channels.h,
	  include/asterisk/app.h, res/res_stasis_playback.c: ARI: Add tones
	  playback resource Adds a tones URI type to the playback resource.
	  The tone can be specified by name (from indications.conf) or by a
	  tone pattern. In addition, tonezone can be specified in the URI
	  (by appending ;tonezone=<zone>). Tones must be stopped manually
	  in order for a stasis control to move on from playback of the
	  tone. Tones may be paused, resumed, restarted, and stopped. They
	  may not be rewound or fast forwarded (tones can't be controlled
	  in a way that lets you skip around from note to note and pausing
	  and resuming will also restart the tone from the beginning).
	  Tests are currently in development for this feature
	  (https://reviewboard.asterisk.org/r/3428/). (closes issue
	  ASTERISK-23433) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3427/ ........ Merged
	  revisions 412535 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-17 20:25 +0000 [r412467-412484]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_oss.c, /, main/Makefile: main/Makefile: Fix build
	  failure on SmartOS/Illumos/SunOS This patch fixes two issues when
	  building on SmartOS: - channels/chan_oss.c: it makes sure
	  soundcard.h is found - main/Makefile: only use
	  "-Wl,--version-script" when GNU LD is used as the Sun Linker
	  doesn't support that. Similar checks are already used elswhere in
	  the Makefile Review: https://reviewboard.asterisk.org/r/3426
	  ASTERISK-23576 #close Reported by: Sebastian Wiedenroth patches:
	  fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
	  ........ Merged revisions 412468 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412483 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/sip/include/sip.h, channels/chan_sip.c, CHANGES:
	  chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL
	  URIs This patch is a continuation of
	  https://reviewboard.asterisk.org/r/3349/, committed in r412303.
	  It resolves a finding oej had that the phone-context be available
	  in a channel variable separate from SIPDOMAIN. This patch adds
	  that variable as SIPURIPHONECONTEXT. It also allows a local
	  number (or global number specified in the TEL URI) to be used to
	  look up as a peer. (issue ASTERISK-17179) Review:
	  https://reviewboard.asterisk.org/r/3349/

2014-04-17 15:17 +0000 [r412454]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_refer.c, /: res_pjsip_refer: Channel variable
	  SIPREFERTOHDR not being set during blind transfer The
	  SIPREFERTOHDR channel variable is not being set on any channel
	  when performing a blind transfer using PJSIP. The
	  'refer->refer_to' was not being set during a blind transfer.
	  Updated so the 'refer_to' is set to the target uri on a blind
	  transfer. (closes issue ASTERISK-23502) Reported by: John Bigelow
	  Review: https://reviewboard.asterisk.org/r/3445/ ........ Merged
	  revisions 412453 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-16 19:14 +0000 [r412440]  Kinsey Moore <kmoore@digium.com>

	* /, include/asterisk/stasis_app.h: Stasis: Add a usage note on
	  stasis_app_get_bridge This function returns an ast_bridge without
	  a refcount bump and the caller must increment the count if it
	  intends to hold the pointer. (closes issue ASTERISK-23588)
	  Review: https://reviewboard.asterisk.org/r/3450/ Reported by:
	  Matt Jordan ........ Merged revisions 412439 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-15 23:21 +0000 [r412427]  Russell Bryant <russell@russellbryant.com>

	* bridges/bridge_builtin_features.c, include/asterisk/monitor.h,
	  CHANGES, apps/app_queue.c, funcs/func_periodic_hook.c,
	  apps/app_mixmonitor.c, include/asterisk/beep.h (added),
	  res/res_monitor.c: (mix)monitor: Add options to enable a periodic
	  beep Add an option to enable a periodic beep to be played into a
	  call if it is being recorded. If enabled, it uses the
	  PERIODIC_HOOK() function internally to play the 'beep' prompt
	  into the call at a specified interval. This option is provided
	  for both Monitor() and MixMonitor(). Review:
	  https://reviewboard.asterisk.org/r/3424/

2014-04-15 18:30 +0000 [r412384-412414]  Richard Mudgett <rmudgett@digium.com>

	* main/stasis_channels.c, main/features_config.c,
	  res/res_parking.c, main/rtp_engine.c, /: Eliminate some more
	  unnecessary RAII_VAR() uses. RAII_VAR() is not a hammer
	  appropriate to pound all nails. ........ Merged revisions 412413
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis_playback.c, /, res/stasis/app.c, res/res_fax.c,
	  res/res_pjsip/security_events.c,
	  res/parking/parking_applications.c, channels/chan_oss.c,
	  main/stasis_bridges.c, res/res_pjsip_session.c,
	  res/stasis_recording/stored.c, main/cdr.c, res/res_parking.c,
	  channels/chan_skinny.c, res/res_pjsip/location.c,
	  res/res_stasis_recording.c, main/stasis_channels.c,
	  res/ari/resource_channels.c, res/parking/parking_manager.c,
	  res/ari/resource_recordings.c, res/res_pjsip_refer.c,
	  res/res_ari.c, main/pbx.c: Remove unused RAII_VAR() declarations.
	  * Remove unused RAII_VAR() declarations. The compiler cannot
	  catch these because the cleanup function "references" the unused
	  variable. Some actually allocated and released resources that
	  were never used. * Fixed some whitespace issues in
	  stasis_bridges.c. ........ Merged revisions 412399 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/rtp_engine.h, main/rtp_engine.c, /,
	  channels/chan_sip.c: chan_sip.c: Fix channel staging assertion
	  failure. The failing assertion ensures that the final snapshot
	  gets generated so CDR records can get finalized. The only place
	  where a channel staging snapshot flag could be left set is in
	  chan_sip.c:handle_request_bye(). The function could return before
	  clearing the flag because the channel could dissappear while the
	  function had to have the channel unlocked. * Fixed
	  handle_request_bye() channel snapshot staging coverage area to
	  not have a return in the middle of it and be unable to clear the
	  staging flag. * Pushed the channel snapshot staging coverage area
	  into ast_rtp_instance_set_stats_vars() to ensure that the staging
	  is not interrutped. * Made callers of
	  ast_rtp_instance_set_stats_vars() not call it with any channels
	  or channel driver private locks held to eliminate the deadlock
	  potential. The callers must hold references to the passed in
	  channel and rtp objects. * Eliminated sip_hangup() trying to get
	  the bridge peer. It is futile at this point because the channel
	  could never be in a bridge. Review:
	  https://reviewboard.asterisk.org/r/3431/ ........ Merged
	  revisions 412385 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: chan_sip.c: Moved some sip_pvt unrefs
	  after their last use. * Moved sip_pvt unref in ast_hangup() and
	  handle_request_do() to the end of the function. The unref needs
	  to happen after the last use of the pointer. ........ Merged
	  revisions 412348 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412383 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-15 16:13 +0000 [r412331]  Jonathan Rose <jrose@digium.com>

	* configs/sip.conf.sample, /, channels/chan_sip.c: Reverting
	  r411189 so that it can be put up for public review --- r411189 |
	  jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines
	  chan_sip: Send real CallerID information with
	  P-Assserted-Identity (RFC-3325) Prior to this patch, the
	  P-Asserted-Identity header would include anonymous caller id
	  information which seems to go against the point of the
	  P-Asserted-Identity header. Now the real caller ID information
	  will be included in this header. Also, no privacy header would be
	  included. This patch adds 'Privacy: id' to outgoing SIP messages
	  that include the P-Asserted-Identity header. (closes issue
	  AST-1301) --- ........ Merged revisions 412328 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 412329 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412330 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-14 15:54 +0000 [r412307]  Corey Farrell <git@cfware.com>

	* main/autoservice.c, /: autoservice: fix reference leak of logger
	  callid. autoservice acquires a local reference to the logger
	  callid of each channel in a loop. This local reference was not
	  released, causing the callid of every channel in autoservice to
	  leak. This change moves the callid unref inside the loop.
	  ASTERISK-23616 #close Reported by: ibercom ........ Merged
	  revisions 412305 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412306 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-12 02:27 +0000 [r412292]  Matthew Jordan <mjordan@digium.com>

	* channels/sip/reqresp_parser.c, CHANGES, channels/chan_sip.c:
	  chan_sip: Support RFC-3966 TEL URIs in inbound INVITE requests
	  This patch adds support for handling TEL URIs in inbound INVITE
	  requests. This includes the Request URI and the From URI. The
	  number specified in the Request URI will be the destination of
	  the inbound channel in the dialplan. The phone-context specified
	  in the Request URI will be stored in the TELPHONECONTEXT channel
	  variable. Review: https://reviewboard.asterisk.org/r/3349
	  ASTERISK-17179 #close Reported by: Geert Van Pamel Tested by:
	  Geert Van Pamel patches:
	  asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van
	  Pamel (License 6140)
	  asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by
	  Geert Van Pamel (License 6140)

2014-04-12 01:35 +0000 [r412279-412280]  Russell Bryant <russell@russellbryant.com>

	* funcs/func_periodic_hook.c: func_periodic_hook: move module ref
	  The previous code left one error path where the module would be
	  unref'd twice instead of once. It was done once in the error
	  handling block, and again inside of datastore destruction. Now
	  the module ref is only released in the datastore destructor and
	  only acquired when the datastore has been successfully allocated.

	* funcs/func_periodic_hook.c: func_periodic_hook: add module ref
	  counting This module lacked necessary module ref count
	  incrementing and decrementing when used. This patch adds it.
	  There's already a datastore used, so doing the ref counting along
	  with the lifetime of the datastore provides a convenient place to
	  do it.

2014-04-11 21:43 +0000 [r412213-412228]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_stack.c, /: app_stack: Add missing unlock in off-nominal
	  path of STACK_PEEK function. ASTERISK-23620 #close Reported by:
	  Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch
	  (license #5021) patch uploaded by Bradley Watkins ........ Merged
	  revisions 412225 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 412226 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412227 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* utils/Makefile, utils: utils dir: Remove no longer needed traces
	  of refcounter except in the clean make target. * Removed no
	  longer needed files from the svn:ignore property to make them
	  visible.

2014-04-11 12:43 +0000 [r412194]  Kinsey Moore <kmoore@digium.com>

	* /, main/bridge.c, main/bridge_basic.c,
	  include/asterisk/stasis_bridges.h, tests/test_cel.c,
	  apps/app_confbridge.c, res/ari/resource_bridges.c: bridging:
	  Ensure locking during snapshot creation While the vast majority
	  of bridge snapshot creation is locked properly, there are
	  currently some instances that are not. This adds the missing
	  locking to ensure bridge state is not malleable during snapshot
	  creation. (closes issue ASTERISK-22904) Review:
	  https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan
	  ........ Merged revisions 412193 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-11 08:28 +0000 [r412168-412180]  Olle Johansson <oej@edvina.net>

	* main/audiohook.c: Formatting: Remove invisible characters

	* main/audiohook.c: Formatting only.

2014-04-11 02:59 +0000 [r412154]  Matthew Jordan <mjordan@digium.com>

	* main/astobj2.c, contrib/scripts/refcounter.py (added),
	  main/asterisk.c, utils/refcounter.c (removed),
	  build_tools/cflags.xml, utils/utils.xml, /, channels/chan_sip.c,
	  channels/sip/security_events.c, include/asterisk/astobj2.h,
	  UPGRADE.txt: main/astobj2: Make REF_DEBUG a menuselect item;
	  improve REF_DEBUG output This patch does the following: (1) It
	  makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
	  REF_DEBUG globally throughout Asterisk. (2) The ref debug log
	  file is now created in the AST_LOG_DIR directory. Every run will
	  now blow away the previous run (as large ref files sometimes
	  caused issues). We now also no longer open/close the file on each
	  write, instead relying on fflush to make sure data gets written
	  to the file (in case the ao2 call being performed is about to
	  cause a crash) (3) It goes with a comma delineated format for the
	  ref debug file. This makes parsing much easier. This also now
	  includes the thread ID of the thread that caused ref change. (4)
	  A new python script instead for refcounting has been added in the
	  contrib/scripts folder. (5) The old refcounter implementation in
	  utils/ has been removed. Review:
	  https://reviewboard.asterisk.org/r/3377/ ........ Merged
	  revisions 412114 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 412115 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 412153 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-11 01:12 +0000 [r412102]  Russell Bryant <russell@russellbryant.com>

	* res/res_monitor.c: monitor: use app options parsing helper code
	  This app is pretty ancient, so it was never converted to use the
	  option parsing helper code. I'd like to add an option to this app
	  that takes an argument, and that's a pain to do when not using
	  this helper, so start by doing this conversion. Review:
	  https://reviewboard.asterisk.org/r/3429/

2014-04-10 21:28 +0000 [r412089]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_hep_pjsip.c: res_hep_pjsip: Use the channel name
	  instead of the call ID when it is available During discussions
	  with Alexandr Dubovikov at Kamailio World, it became apparent
	  that while the SIP call ID is a useful identifier prior to an
	  Asterisk channel being created, it is far more preferable to use
	  the channel name (or some channel based identifier) when the
	  channel is available. Homer is smart enough to tie the various
	  messages together. This patch opts to use the channel name when
	  it is available, falling back to the call ID otherwise. ........
	  Merged revisions 412088 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-10 21:10 +0000 [r412075]  Kevin Harwell <kharwell@digium.com>

	* /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Set the body
	  generation result to 0 for a valid path The result of the
	  "ast_sip_pubsub_generate_body_content" was not set/initialized.
	  Consequently, the nominal path potentially returned an invalid
	  value, thus not sending mwi notifications. ........ Merged
	  revisions 412074 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-09 21:43 +0000 [r412050]  Mark Michelson <mmichelson@digium.com>

	* /, CHANGES, apps/app_mixmonitor.c: Add a Command header to the
	  AMI Mixmonitor action. This fixes a parsing error that occurred
	  during the processing of the AMI action. The error did not result
	  in MixMonitor itself misbehaving, but it could result in the AMI
	  response not giving correct information back. The new header
	  allows for one to specify a post-process command to run when
	  recording finishes. Previously, in order to do this, the
	  post-process command would have to be placed at the end of the
	  Options: header. Patches: mixmonitor_command_2.patch by jhardin
	  (License #6512) ........ Merged revisions 412048 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-09 18:17 +0000 [r412035]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_stasis_answer.c: res_stasis_answer: Add missing
	  newlines ........ Merged revisions 412034 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-08 21:25 +0000 [r411946-411990]  Richard Mudgett <rmudgett@digium.com>

	* /, main/asterisk.c: Internal timing: Add notice that the -I and
	  internal_timing option are no longer needed. Add notice messages
	  during execution that the -I command line option and the
	  astersik.conf internal_timing option are no longer needed. The
	  internal timing functionality is now always enabled if there is a
	  timing module loaded. NOTE: Since the command line options and
	  the asterisk.conf config file are processed before the logging
	  system is initialized, the messages are output to stderr. Change
	  requested as a result of asterisk-dev list comments about the
	  commit for ASTERISK-22846 that removed the -I and internal_timing
	  options. Review: https://reviewboard.asterisk.org/r/3423/
	  ........ Merged revisions 411964 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411974 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411985 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/config.c, /: config: Fix CB_ADD_LEN() to work as originally
	  intended. Fix a long standing bug in CB_ADD_LEN() behaving like
	  CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes
	  ........ Merged revisions 411960 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411961 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411962 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix
	  confbridge.conf dsp_talking_threshold option setting wrong
	  parameter. Fixed copy pasta error. ASTERISK-23545 #close Reported
	  by: John Knott ........ Merged revisions 411944 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411945 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-08 14:49 +0000 [r411928]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip.c: res_pjsip: Ignore explicit transport
	  configuration if a WebSocket transport is specified. This change
	  makes it so if a transport is configured on an endpoint that is a
	  WebSocket type the option will be ignored. In practice this is
	  fine because the WebSocket transport can not create outgoing
	  connections, it can only reuse existing ones. By ignoring the
	  option the existing PJSIP logic for using the existing connection
	  will be invoked and stuff will proceed. (closes issue
	  ASTERISK-23584) Reported by: Rusty Newton ........ Merged
	  revisions 411927 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-08 00:26 +0000 [r411897]  Russell Bryant <russell@russellbryant.com>

	* funcs/func_periodic_hook.c: func_periodic_hook: List more modules
	  as dependencies This module makes use of some existing Asterisk
	  components. app_chanspy was already listed as a dependency. There
	  are a few function modules used, as well, so list them.

2014-04-07 20:41 +0000 [r411884]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_pjsip_pubsub.c: PJSIP: Ensure test event has new state
	  The change that fixed the pubsub test event's use of a dangling
	  pointer also changed when it was processed relative to the pjsip
	  subscription state change processing. This change corrects the
	  order of events while holding a reference to the pointer that was
	  previously dangling. ........ Merged revisions 411883 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-07 16:15 +0000 [r411870]  Jonathan Rose <jrose@digium.com>

	* main/manager_channels.c, /: AGI/Manager: Prevent multiple
	  NewExten events during AGI application changes AGI applications
	  would trigger NewExten events every time the state of the AGI
	  application changed. This has historically not been the behavior
	  and this behavior was introduced with a CDR patch. This patch
	  corrects that. (closes issue ASTERISK-23390) Reported by:
	  Benjamin Keith Ford Review:
	  https://reviewboard.asterisk.org/r/3406/ ........ Merged
	  revisions 411868 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-07 14:57 +0000 [r411812]  Walter Doekes <walter+asterisk@wjd.nu>

	* apps/app_queue.c, /: app_queue: Re-add HoldTime to
	  QueueCallerAbandon event (simple typo during ast12 refactor).
	  Reported by: Ibrahim22 (on IRC) Tested by: Ibrahim22 ........
	  Merged revisions 411811 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-07 14:29 +0000 [r411791-411806]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_stasis.c: Stasis: Fix Stasis() bridge refcount issue
	  The Stasis() dialplan application monitors what bridge a channel
	  is in and so necessarily holds on to a bridge pointer. This
	  change ensures that it also holds on to a reference for that
	  bridge to prevent the bridge pointer from becoming a dangling
	  pointer. ........ Merged revisions 411804 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_pubsub.c, /: PJSIP: Fix crash introduced in r411671
	  The test event introduced in revision 411671 uses a dangling
	  pointer to access information about pubsub state changes. This
	  moves the event to within the lifetime of the pointer. ........
	  Merged revisions 411790 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-05 13:06 +0000 [r411768]  Russell Bryant <russell@russellbryant.com>

	* CHANGES, funcs/func_periodic_hook.c (added): func_periodic_hook:
	  New function for periodic hooks. This commit introduces a new
	  dialplan function, PERIODIC_HOOK(). It allows you run to a
	  dialplan hook on a channel periodically. The original use case
	  that inspired this was the ability to play a beep periodically
	  into a call being recorded. The implementation is much more
	  generic though and could be used for many other things. The
	  implementation makes heavy use of existing Asterisk components.
	  It uses a combination of Local channels and ChanSpy() to run some
	  custom dialplan and inject any audio it generates into an active
	  call. The other important bit of the implementation is how it
	  figures out when to trigger the beep playback. This
	  implementation uses the audiohook API, even though it's not
	  actually touching the audio in any way. It's a convenient way to
	  get a callback and check if it's time to kick off another beep.
	  It would be nice if this was timer event based instead of polling
	  based, but unfortunately I don't see a way to do it that won't
	  interfere with other things. Review:
	  https://reviewboard.asterisk.org/r/3362/

2014-04-04 19:19 +0000 [r411702-411724]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/options.h, main/asterisk.c, main/channel.c, /,
	  channels/chan_sip.c, configs/asterisk.conf.sample, UPGRADE.txt,
	  include/asterisk/channel.h, utils/extconf.c: internal_timing:
	  Remove the option and always make it enabled if a timing module
	  is loaded. The masquerade supertest frequently fails because
	  either the local channel chain doesn't completely optimize out or
	  the DTMF handshake doesn't completely get accross. Local channel
	  optimization requires frames flowing to trigger when optimization
	  can happen. When optimization happens the media frame that
	  triggered the optimization is dropped. Sending DTMF requires
	  frames to flow in the other direction for timing purposes while
	  sending nothing. If internal timing is not enabled when MOH is
	  playing, Asterisk switches to received timing when an audio frame
	  is received. With optimization dropping media frames and MOH not
	  sending frames unless it receives frames, occasionaly there are
	  no more frames being passed and the test fails. * The asterisk
	  command line -I option and the asterisk.conf internal_timing
	  option are removed. Asterisk now always uses internal timing when
	  needed if any timing module is loaded. The issue ASTERISK-14861
	  did this quite awhile ago in v1.4 but effectively is broken if
	  other internal timing modules besides DAHDI are used. The
	  ast_read_generator_actions() now only does received timing if it
	  has no choice for frame generators like MOH, silence, and
	  playback streaming. * Cleaned up some code dealing with frame
	  generators in ast_deactivate_generator(),
	  generator_write_format_change(), ast_activate_generator(), and
	  ast_channel_stop_silence_generator(). * Removed
	  ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and
	  ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........
	  Merged revisions 411715 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411716 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411717 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/utils.c, res/res_musiconhold.c, main/channel.c,
	  main/stasis_cache.c, /: Add some asserts that were handy when
	  looking for a stasis cache problem. * Assert if a channel is
	  destroyed but has the snapshot staging flag set. In this case the
	  final channel destruction snapshot would never get taken. *
	  Assert if what we just got out of the stasis cache is not what we
	  were looking for. This assert would have saved several days
	  searching for a bug and a lot of my hair. * Assert if the music
	  on hold message posts could not find the associated channel. A
	  crash will happen later when manager tries to send the MOH AMI
	  message. This assert catches the problem when the stasis message
	  is posted instead of by the thread processing the defective
	  message. * Always generate a backtrace when an ast_assert()
	  fails. Review: https://reviewboard.asterisk.org/r/3411/ ........
	  Merged revisions 411701 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-04 15:13 +0000 [r411688]  Matthew Jordan <mjordan@digium.com>

	* /, main/http.c: http: Fix spurious ERROR message in responses
	  with no content When a response has a content length of 0, fwrite
	  would be called to write a buffer with no data in it. This
	  resulted in the following classic error message: [Apr 3 11:49:17]
	  ERROR[26421] http.c: fwrite() failed: Success This patch makes it
	  so that we only attempt to write out the content if the
	  calculated content_length is non-zero. ........ Merged revisions
	  411687 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-03 12:06 +0000 [r411671]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Add test event for
	  state change This adds a test event when subscription state
	  changes so that integration tests may trigger new actions at the
	  appropriate times. Review:
	  https://reviewboard.asterisk.org/r/3383/ ........ Merged
	  revisions 411670 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-03 11:47 +0000 [r411669]  Matthew Jordan <mjordan@digium.com>

	* res/res_hep.c, /: res_hep: Fix crash when hep.conf not available
	  Parts of res_hep properly checked for a valid configuration
	  object before attempting to access the configuration. A check,
	  however, was missed when a packet is sent. This patch fixes the
	  crash caused by not checking if the configuration object is
	  valid. ........ Merged revisions 411668 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-02 18:57 +0000 [r411656]  Mark Michelson <mmichelson@digium.com>

	* main/sorcery.c, /, res/res_mwi_external.c,
	  res/res_pjsip/config_system.c, configs/sorcery.conf.sample,
	  main/bucket.c, include/asterisk/sorcery.h,
	  res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c,
	  tests/test_sorcery.c, tests/test_sorcery_realtime.c: Prevent
	  duplicate sorcery wizards from being applied to sorcery object
	  types. This commit contains several changes to sorcery: 1)
	  Application of sorcery configuration based on module name is
	  automatically performed when sorcery is opened for a module. 2)
	  Sorcery will not attempt to apply the same wizard to an object
	  type more than once. 3) Sorcery gives more exact results when
	  attempting to apply a wizard, whether as the default or based on
	  configuration. Sorcery unit tests still pass for me after making
	  these changes. Review: https://reviewboard.asterisk.org/r/3326
	  ........ Merged revisions 411159 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-01 22:42 +0000 [r411637-411639]  Richard Mudgett <rmudgett@digium.com>

	* res/parking/parking_bridge.c, /: res_parking: Minor tweaks. * Use
	  ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of
	  ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables.
	  * Use ast_copy_string() instead of inlining it. * Remove an
	  already done TODO comment. * Some whitespace tweaks. ........
	  Merged revisions 411638 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis_channels.c, /: stasis_channels.c: Eliminate another
	  overuse of RAII_VAR(). ........ Merged revisions 411636 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-04-01 16:52 +0000 [r411587]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: app_queue: Fix a bug where realtime members
	  would be deleted during reload causing waiting callers to get
	  ejected. This patch causes realtime queue members to remain in
	  queues during the reload process. Previously these members would
	  be removed causing any waiting callers to be ejected from the
	  queue with a reason of "EXITEMPTY". ASTERISK-23547 #close
	  ASTERISK-23547 #comment Patch
	  app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo
	  Rossi (license 6409) Review:
	  https://reviewboard.asterisk.org/r/3404/ ........ Merged
	  revisions 411584 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411585 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411586 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-28 18:32 +0000 [r411556]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/res_hep.h (added), res/res_hep_pjsip.c (added),
	  res/res_hep.exports.in (added), configs/hep.conf.sample (added),
	  CHANGES, res/res_hep.c (added), /: res_hep/res_hep_pjsip: Add a
	  HEPv3 capture agent module and a logger for PJSIP This patch adds
	  the following: (1) A new module, res_hep, which implements a
	  generic packet capture agent for the Homer Encapsulation Protocol
	  (HEP) version 3. Note that this code is based on a patch provided
	  by Alexandr Dubovikov; I basically just wrapped it up, added
	  configuration via the configuration framework, and threw in a
	  taskprocessor. (2) A new module, res_hep_pjsip, which forwards
	  all SIP message traffic that passes through the res_pjsip stack
	  over to res_hep for encapsulation and transmission to a HEPv3
	  capture server. Much thanks to Alexandr for his Asterisk patch
	  for this code and for a *lot* of patience waiting for me to port
	  it to 12/trunk. Due to some dithering on my part, this has taken
	  the better part of a year to port forward (I still blame CDRs for
	  the delay). ASTERISK-23557 #close Review:
	  https://reviewboard.asterisk.org/r/3207/ ........ Merged
	  revisions 411534 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-28 18:00 +0000 [r411533]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
	  addons/chan_ooh323.c, /, addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c:
	  process stack command even if gatekeeper client isn't register
	  don't destroy gatekeeper client if it is not started don't
	  destroy gatekeeper client in some sort of gatekeeper errors
	  signal rtp create condition when call cleared before rtp
	  structure created (closes issue ASTERISK-23460) Reported by:
	  Dmitry Melekhov Patches: ASTERISK-23460-2.patch Tested by: Dmitry
	  Melekhov ........ Merged revisions 411531 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411532 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-28 17:41 +0000 [r411515-411530]  Matthew Jordan <mjordan@digium.com>

	* rest-api/api-docs/channels.json,
	  rest-api/api-docs/recordings.json,
	  rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
	  /, rest-api/api-docs/playbacks.json, UPGRADE.txt,
	  rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES,
	  include/asterisk/manager.h, rest-api/api-docs/bridges.json,
	  rest-api/api-docs/deviceStates.json,
	  rest-api/api-docs/mailboxes.json,
	  rest-api/api-docs/asterisk.json,
	  rest-api/api-docs/applications.json: Update API versions and
	  UPGRADE/CHANGES for 12.2.0 This patch does the following: * It
	  updates the AMI version to 2.2.0 to indicate backwards compatible
	  changes have been made since the last release * It updates the
	  ARI version to 1.2.0 to indicate backwards compatible changes
	  have been made since the last release * It updates the
	  UPGRADE/CHANGES files with changes that were not mentioned
	  ........ Merged revisions 411529 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* UPGRADE.txt, res/res_config_odbc.c: res_config_odbc: Fix for
	  nullable integer columns and keyfield existence check in
	  update_odbc. This patch fixes setting nullable integer columns to
	  NULL instead of an empty string, which fails for PostgreSQL, for
	  example. The current code is supposed to do so, but the check is
	  broken. The patch also allows the first column in the list to be
	  a nullable integer. Also, the check for existence of a mandatory
	  column checked for the first column in the list instead of the
	  key field lookup column. This patch fixes that issue as well.
	  Finally, the compatibility option allow_empty_string_in_nontext,
	  which was added to previous revisions to allow for some database
	  backends with certain schemas to function, has been removed.
	  Review: https://reviewboard.asterisk.org/r/3335 ASTERISK-23459
	  #close ASTERISK-23351 #close (closes issue ASTERISK-23459)
	  Reported by: zvision patches: res_config_odbc.diff uploaded by
	  zvision (License 5755)

2014-03-28 16:18 +0000 [r411469]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/tcptls.c, main/manager.c, /, main/http.c: http: response
	  body often missing after specific request This patch works around
	  a problem with the HTTP body being dropped from the response to a
	  specific client and under specific circumstances: a) Client
	  request comes from node.js user agent "Shred" via use of
	  swagger-client library. b) Asterisk and Client are *not* on the
	  same host or TCP/IP stack In testing this problem, it has been
	  determined that the write of the HTTP body is lost, even if the
	  data is written using low level write function. The only solution
	  found is to instruct the TCP stack with the shutdown function to
	  flush the last write and finish the transmission. See review for
	  more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
	  Reported by: Sam Galarneau Review:
	  https://reviewboard.asterisk.org/r/3402/ ........ Merged
	  revisions 411462 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411463 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411465 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-28 15:48 +0000 [r411375-411460]  Matthew Jordan <mjordan@digium.com>

	* UPGRADE.txt, /: UPGRADE: Note IAX2 compatibility issue between
	  1.4 and 1.8+ systems. ........ Merged revisions 411457 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411458 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411459 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* contrib/realtime/mysql/voicemail_messages.sql (removed),
	  contrib/realtime/postgresql/realtime.sql (removed),
	  contrib/realtime/mysql/voicemail_data.sql (removed),
	  contrib/realtime/mysql/musiconhold.sql (removed),
	  contrib/realtime/mysql/queue_log.sql (removed),
	  contrib/realtime/mysql/voicemail.sql (removed),
	  contrib/realtime/mysql/sippeers.sql (removed), /,
	  contrib/realtime/mysql/iaxfriends.sql (removed),
	  contrib/realtime/mysql/meetme.sql (removed): contrib/realtime:
	  Remove empty SQL script files Since the relatime scripts are now
	  managed by Alembic, the previous realtime scripts were previously
	  removed. However, the removal process messed up, as the files
	  were still in the repository. The contents were just empty. This
	  removes the files from the tree. ........ Merged revisions 411442
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/sip/include/sip.h: chan_sip: Add MESSAGE request to
	  allowed methods The allowed methods advertised by chan_sip did
	  not previously note the MESSAGE request. Even in Asterisk 1.8, we
	  do accept in-dialog MESSAGE requests; we should advertise that we
	  support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
	  #comment Reported by: Martin Kontsek ASTERISK-23504 #comment
	  Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
	  Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged
	  revisions 411372 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411373 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411374 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-27 19:21 +0000 [r411312-411328]  Corey Farrell <git@cfware.com>

	* funcs/func_global.c, apps/app_speech_utils.c,
	  apps/confbridge/conf_config_parser.c,
	  funcs/func_callcompletion.c, funcs/func_frame_trace.c,
	  funcs/func_callerid.c, main/message.c, /, res/res_mutestream.c,
	  channels/pjsip/dialplan_functions.c,
	  res/res_pjsip_header_funcs.c, funcs/func_pitchshift.c,
	  funcs/func_groupcount.c, funcs/func_volume.c, funcs/func_odbc.c,
	  funcs/func_channel.c, funcs/func_cdr.c, funcs/func_blacklist.c,
	  apps/app_stack.c, apps/app_voicemail.c, res/res_calendar.c,
	  apps/app_jack.c, funcs/func_dialplan.c, funcs/func_speex.c,
	  channels/chan_sip.c, funcs/func_math.c, funcs/func_strings.c,
	  funcs/func_jitterbuffer.c, res/res_xmpp.c, channels/chan_iax2.c,
	  main/features_config.c, res/res_jabber.c: Fix dialplan function
	  NULL channel safety issues (closes issue ASTERISK-23391) Reported
	  by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3386/ ........ Merged
	  revisions 411313 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411314 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411315 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/format.c, include/asterisk.h, /: main/formats: Fix crash in
	  ast_format_cmp during non-clean shutdown. * Update asterisk.h to
	  reflect availability of ast_register_cleanup in 11.9. * Use
	  ast_register_cleanup for format_attr_shutdown. (closes issue
	  ASTERISK-23103) Reported by: JoshE ........ Merged revisions
	  411310 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........ Merged revisions 411311 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-27 14:21 +0000 [r411296]  Mark Michelson <mmichelson@digium.com>

	* main/sorcery.c, /: Give sorcery instances a reference to their
	  wizards. On graceful shutdown, sorcery wizards are all killed
	  off, but it is possible for sorcery instances to still have
	  dangling pointers after this, possibly causing a crash. Giving
	  the sorcery instances a reference to their wizards ensures that
	  the wizard reference will remain valid for the lifetime of the
	  sorcery instance. Review: https://reviewboard.asterisk.org/r/3401
	  ........ Merged revisions 411295 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-26 22:45 +0000 [r411246]  Joshua Colp <jcolp@digium.com>

	* /, main/say.c: say: Fix a bug where SayNumber in Polish tries to
	  play incorrect sound. This change fixes a bug where calling
	  SayNumber with a number divisible by 100 using the Polish
	  language would cause the code to attempt to play a sound file
	  with an empty name. (closes issue ASTERISK-23509) Reported by:
	  zvision Review: https://reviewboard.asterisk.org/r/3378/ ........
	  Merged revisions 411243 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411244 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411245 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-26 16:15 +0000 [r411194]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Send
	  real CallerID information with P-Assserted-Identity (RFC-3325)
	  Prior too this patch, the P-Asserted-Identity header would
	  include anonymous caller id information which seems to go against
	  the point of the P-Asserted-Identity header. Now the real caller
	  ID information will be included in this header. Also, no privacy
	  header would be included. This patch adds 'Privacy: id' to
	  outgoing SIP messages that include the P-Asserted-Identity
	  header. (closes issue AST-1301) ........ Merged revisions 411189
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 411190 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411193 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-26 16:05 +0000 [r411192]  Richard Mudgett <rmudgett@digium.com>

	* /,
	  contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py:
	  Fix 'alembic branches' merge conflict as described by the web
	  page. ........ Merged revisions 411191 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-25 18:44 +0000 [r411174]  Sean Bright <sean@malleable.com>

	* /, res/ari/config.c: ARI: Don't complain about missing ARI users
	  when we aren't enabled Currently, if ARI is not enabled it will
	  still complain that there are no configured users. This patch
	  checks to see if ARI is enabled before logging and error or
	  iterating the container to validate the users. Review:
	  https://reviewboard.asterisk.org/r/3391/ ........ Merged
	  revisions 411173 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-25 17:40 +0000 [r411158]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
	  res/res_pjsip_messaging.c, res/res_pjsip.c,
	  include/asterisk/res_pjsip.h: Add a "message_context" option for
	  PJSIP endpoints. ........ Merged revisions 411157 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-25 16:57 +0000 [r411142]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
	  include/asterisk/res_pjsip.h, /: res_pjsip: Fix contact
	  authenticate_qualify endpoint lookup when qualifing a contact. *
	  Fixed bad use of ao2_find() in on_endpoint(). * Replaced use of
	  find_endpoints() with find_an_endpoint() since only the first
	  found endpoint is ever needed. * Fixed qualify_contact_cb() to
	  update the contact with the aor authenticate_qualify setting.
	  Otherwise, permanent contacts in the aor type sections would have
	  a config line order dependancy. * Fixed off nominal path contact
	  ref leak in qualify_contact(). The comment saying the unref is
	  not needed was wrong. * Fixed off nominal path use of the
	  endpoint parameter if it is NULL in send_out_of_dialog_request().
	  * Added missing off nominal path unref of pjsip tdata in
	  send_out_of_dialog_request(). * Fixed off nominal path failing to
	  call the callback in send_request_cb() when the request is
	  challenged for authentication. * Eliminated silly RAII_VAR() use
	  in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen
	  to better reflect reality. (closes issue ASTERISK-23254) Reported
	  by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/
	  ........ Merged revisions 411141 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-25 16:06 +0000 [r411092]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
	  update_provisional_keepalive() is called while
	  send_provisional_keepalive_full() is waiting on the PVT lock,
	  then pvt->provisional_keepalive_sched_id will be changed to a new
	  sched_id value by update_provisional_keepalive(), but that new
	  sched_id then may be overwritten with -1 by
	  send_provisional_keepalive_full(), killing the pvt's reference to
	  a schedule and "leaking" the reference. (closes issue
	  ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
	  Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
	  Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
	  (license 5012) ........ Merged revisions 411088 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411089 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411091 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-25 15:56 +0000 [r411090]  Jonathan Rose <jrose@digium.com>

	* /, res/res_stasis.c: ARI: Resolve a subscription leak against
	  implicit bridge subscriptions When a channel in a stasis
	  application is joined to a bridge, a subscription for that bridge
	  is created implicitly for the stasis application serving the
	  channel. Prior to this patch, subsequent removals of the channel
	  from the bridge would leave the subscription open. Review:
	  https://reviewboard.asterisk.org/r/3380/ ........ Merged
	  revisions 411086 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-25 15:47 +0000 [r411073-411087]  Richard Mudgett <rmudgett@digium.com>

	* utils/conf2ael.c, main/lock.c, utils/ael_main.c: Revert -r411073.
	  It didn't help and blew up the system.

	* utils/ael_main.c, utils/conf2ael.c, main/lock.c: locking: Add
	  temporary sanity checks. Add some temporary sanity checks to hunt
	  for locking problems with the masquerade supertest.

2014-03-24 21:39 +0000 [r411024]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: chan_sip: Always use fromdomain if set
	  for domain, even if callerid is set to restricted. (closes issue
	  ASTERISK-20841) Reported by: Kelly Goedert ........ Merged
	  revisions 411021 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 411022 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 411023 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-21 16:04 +0000 [r410996]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_pjsip_registrar.c: res_pjsip_registrar.c:
	  Miscellaneous cleanup in rx_task(). * Fix variable shadowing of
	  'updated' by renaming it to 'contact_update'. * Checked
	  'contact_update' for ast_sorcery_copy() failure. * Removed silly
	  use of RAII_VAR() for 'contact_update'. ........ Merged revisions
	  410995 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-21 15:50 +0000 [r410981-410994]  Sean Bright <sean@malleable.com>

	* res/ael/ael.flex, utils/Makefile, pbx/pbx_ael.c,
	  res/ael/ael_lex.c: Make the AEL load process less chatty.
	  Switched a bunch of LOG_NOTICEs to ast_debug. This time without
	  breaking the build.

	* pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Revert
	  r410981. aelparse blew up.

	* main/config.c: Remove a LOG_NOTICE from
	  ast_config_engine_register. There is enough indication from the
	  CLI that we are loading a realtime engine as it is.

	* pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Make the AEL
	  load process less chatty. Switched a bunch of LOG_NOTICEs to
	  ast_debug.

2014-03-20 23:02 +0000 [r410967]  Jonathan Rose <jrose@digium.com>

	* apps/app_confbridge.c, /: app_confbridge: Fix bug - users with
	  startmuted set don't start muted (closes issue ASTERISK-23461)
	  Reported by: Chico Manobela Review:
	  https://reviewboard.asterisk.org/r/3373/ ........ Merged
	  revisions 410965 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410966 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-20 16:35 +0000 [r410950]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/rtp_engine.h, main/dial.c, main/manager.c, /,
	  main/channel_internal_api.c, main/core_unreal.c,
	  include/asterisk/channel.h, res/ari/resource_channels.c,
	  res/res_stasis_snoop.c: assigned-uniqueids: Miscellaneous cleanup
	  and fixes. * Fix memory leak in ast_unreal_new_channels(). Made
	  it generate the ;2 uniqueid on a stack variable instead of
	  mallocing it. * Made send error response to ARI and AMI requests
	  instead of just logging excessive uniqueid length and allowing
	  truncation. action_originate() and
	  ari_channels_handle_originate_with_id(). * Fixed minor truncating
	  uniqueid hole when generating the ;2 uniqueid string length.
	  Created public and internal lengths of uniqueid. The internal
	  length can handle a max public uniqueid plus an appended ;2. *
	  free() and ast_free() are NULL tolerant so they don't need a NULL
	  test before calling. * Made use better struct initialization
	  format instead of the position dependent initialization format.
	  Also anything not explicitly initialized in the struct is
	  initialized to zero by the compiler. * Made
	  ast_channel_internal_set_fake_ids() use the safer
	  ast_copy_string() instead of strncpy(). Review:
	  https://reviewboard.asterisk.org/r/3371/ ........ Merged
	  revisions 410949 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-19 17:27 +0000 [r410934]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip_endpoint_identifier_ip.c: PJSIP: Allow for
	  identify sections to be specified in sorcery.conf. "identify" is
	  a special type of configuration object in PJSIP because unlike
	  the other objects, it is not provided by the base res_pjsip
	  module. Instead, it is provided by the
	  res_pjsip_endpoint_identifier_ip module. If using the default
	  sorcery wizard (config,criteria=type=identify) then things work
	  because the module that applies the default wizard is the correct
	  module. However, if attempting to use sorcery.conf to apply an
	  alternate wizard, it was not possible. If you attempted to
	  specify the identify object type in the res_pjsip section, then
	  the object could not be registered since the object was
	  undocumented for the res_pjsip module. There was no alternate
	  configuration section defined for it, so you were out of luck if
	  you wanted to override the default wizard. With this change, the
	  identify section will properly have a sorcery.conf-based wizard
	  applied when the identify definition is within the
	  res_pjsip_endpoint_identifier_ip section. ........ Merged
	  revisions 410933 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-19 14:25 +0000 [r410905-410919]  Joshua Colp <jcolp@digium.com>

	* res/res_stasis.c, /: res_stasis: Fix a bug where the default
	  bridge type was not set. ........ Merged revisions 410918 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* CHANGES, res/res_stasis.c, rest-api/api-docs/bridges.json, /,
	  res/ari/resource_bridges.h: res_stasis: Extend bridge type to be
	  a comma separated list of bridge attributes. This change turns
	  the bridge type field into a comma separated list of attributes.
	  These attributes include: mixing, holding, dtmf_events, and
	  proxy_media. By setting the various attributes a user can control
	  the type of bridge created with the behavior they need for their
	  application. (closes issue ASTERISK-23437) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/3359/ ........
	  Merged revisions 410904 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-19 02:33 +0000 [r410891]  Matthew Jordan <mjordan@digium.com>

	* res/res_ari.c, /: res_ari: Fix documentation schema error
	  ........ Merged revisions 410890 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-18 23:32 +0000 [r410877]  Rusty Newton <rnewton@digium.com>

	* res/res_ari.c, /: res_ari: Add notes about Asterisk HTTP server
	  to the "enabled" config option for the res_ari general section
	  Added note and see-also reminding user to enable the HTTP server.
	  (closes issue ASTERISK-22499) Reported by: Rusty Newton ........
	  Merged revisions 410876 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-18 15:45 +0000 [r410863]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, main/http.c: ARI: allow json content type with zero length
	  body When a request was received with a Content-type of json, the
	  body was sent for json parsing - even if it was zero length. This
	  resulted in ARI requests failing that were valid, such as a
	  channel DELETE with no parameters. The code has now been changed
	  to skip json parsing with zero content length. (closes issue
	  SWP-6748) Reported by: Samuel Galarneau Review:
	  https://reviewboard.asterisk.org/r/3360/ ........ Merged
	  revisions 410858 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-18 15:28 +0000 [r410862]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /: cdr: Add asserts for when we don't know about a
	  CDR for a channel In the CDR core, every channel should either be
	  filtered out (due to being an 'internal' channel used as an
	  implementation detail, such as playing media back into a bridge)
	  or it should get a CDR. Even if that CDR ends up being discarded,
	  we still give the channel a CDR in case we end up needing it. If
	  we hit a situation where a channel does not have a CDR, we should
	  blow up in -dev-mode. Asserts are appropriate for that. This
	  patch adds those asserts, as they would have quickly caught the
	  error fixed by r410814. ........ Merged revisions 410861 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-18 12:45 +0000 [r410845]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip/config_system.c: res_pjsip: Fix memory leak of
	  nameservers in off-nominal resolver creation failure. Thanks
	  Walter Doekes! ........ Merged revisions 410844 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-18 11:52 +0000 [r410831]  Sean Bright <sean@malleable.com>

	* res/res_fax_spandsp.c, /: res_fax_spandsp: Use g711_free() when
	  available. Per Johann Steinwendtner on the asterisk-dev mailing
	  list:
	  http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html
	  g711_free() was introduced in spandsp 0.0.6pre4 and
	  g711_release() became a noop. I opted not to remove the call to
	  g711_release() since it is harmless and to call g711_free() if we
	  have a sufficiently recent version of spandsp. (issue
	  ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged
	  revisions 410829 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410830 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-18 02:09 +0000 [r410814]  Richard Mudgett <rmudgett@digium.com>

	* main/stasis_cache.c, /: stasis_cache: Use the right variable in
	  the cache entry ao2 cmp function. ........ Merged revisions
	  410813 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-17 22:54 +0000 [r410794-410796]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/dns.h, CHANGES,
	  res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c,
	  main/dns.c, /, res/res_pjsip/config_system.c: res_pjsip: Enable
	  PJSIP DNS client support. This change enables DNS client support
	  within PJSIP. System nameservers are automatically discovered
	  using res_init or res_ninit. If this fails then PJSIP will resort
	  to using gethostbyname for resolution. By enabling this support
	  we gain SRV support, failover, and weight support. (closes issue
	  ASTERISK-23435) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3343/ ........ Merged
	  revisions 410795 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Make address
	  replacement less aggressive. This change makes the
	  res_pjsip_multihomed module less aggressive when changing the
	  address in messages. It will now only occur if the transport in
	  use is bound to the any address OR if the system determined
	  source address matches the bound address of the transport in use.
	  Review: https://reviewboard.asterisk.org/r/3369/ ........ Merged
	  revisions 410793 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-17 22:24 +0000 [r410775]  Russ Meyerriecks <rmeyerreicks@digium.com>

	* /, main/callerid.c: callerid: Logic error in checksum processing
	  Callerid checksum-ing was being handled incorrectly here. When
	  the checksum is calculated to be 0x00, it will perform 0x100-0x00
	  which results in 0x100. This value will then fail the otherwise
	  correct callerid message. This patch changes the logic to simply
	  add the calculated checksum to the transmitted 2's compliment
	  checksum. Review: https://reviewboard.asterisk.org/r/3356/
	  (closes issue ASTERISK-23488) ........ This is a merge of merged
	  revisions 410750 410747 from
	  http://svn.asterisk.org/svn/asterisk/branches/12 I didn't want a
	  broken patch to be comitted to trunk so I pre-merge merged them.

2014-03-17 19:35 +0000 [r410684-410699]  Mark Michelson <mmichelson@digium.com>

	* res/res_mwi_external.c, res/res_pjsip/config_system.c,
	  configs/sorcery.conf.sample, include/asterisk/sorcery.h,
	  res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c,
	  tests/test_sorcery.c, tests/test_sorcery_realtime.c,
	  main/sorcery.c, /: Revert changes to sorcery that accidentally
	  got committed. These changes were still up for review and have
	  not been approved yet. I must have had the changes in my working
	  copy when making a different change. ........ Merged revisions
	  410696 from http://svn.asterisk.org/svn/asterisk/branches/12

	* bridges/bridge_softmix.c, tests/test_sorcery.c, main/channel.c,
	  res/res_pjsip/config_system.c, res/res_mwi_external.c,
	  include/asterisk/bridge_channel.h, funcs/func_frame_trace.c,
	  configs/sorcery.conf.sample, res/res_pjsip/pjsip_configuration.c,
	  include/asterisk/sorcery.h, tests/test_sorcery_astdb.c,
	  include/asterisk/frame.h, main/bridge_channel.c,
	  tests/test_sorcery_realtime.c, main/sorcery.c,
	  res/res_stasis_playback.c, main/frame.c, /: Fix stuck channel in
	  ARI through the introduction of synchronous bridge actions.
	  Playing back a file to a channel in an ARI bridge would attempt
	  to wait until the playback concluded before returning. The method
	  used involved signaling the waiting thread in the ARI custom
	  playback function. The problem with this is that there were some
	  corner cases that were not accounted for: * If a bridge channel
	  could not be found, then we never would attempt the playback but
	  would still attempt to wait for the playback to complete. * If
	  the bridge playfile action failed to queue, we would still
	  attempt to wait for the playback to complete. * If the bridge
	  playfile action were queued but some circumstance caused the
	  playback not to occur (the bridge dies, the channel is removed
	  from the bridge), then we would never be notified. The solution
	  to this is to move the waiting logic into the bridge code. A new
	  bridge API function is added to queue a synchronous action on a
	  bridge. The waiting thread is notified when the queued frame has
	  been freed, either due to an error occurring or due to successful
	  playback. As a failsafe, the waiting thread has a 10 minute
	  timeout just in case there is a frame leak somewhere. Review:
	  https://reviewboard.asterisk.org/r/3338 ........ Merged revisions
	  410673 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-17 16:48 +0000 [r410672]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/confbridge/conf_chan_announce.c: app_confbridge: Add
	  missing destructor call to announcer channel destructor. ........
	  Merged revisions 410671 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-16 20:27 +0000 [r410651]  Matthew Jordan <mjordan@digium.com>

	* /, res/stasis/app.c: stasis/app.c: Add some extra debugging for
	  subscription counts Events are sent to a connected ARI
	  application based on the things that ARI application cares about.
	  These subscriptions can be set up implicitly - such as when that
	  ARI application creates a new object - or explicitly, via the
	  application resource's subscription operations. Debugging *why*
	  something was being sent to an application - or why something was
	  not being sent to an application - was a bit tricky, as there was
	  no debug information for the subscriptions. This patch adds some
	  debug level 3 statements that show the subscription counts for
	  applications. (Level 3 was chosen as it matches the verbose level
	  3 statements elsewhere) ........ Merged revisions 410650 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-15 15:24 +0000 [r410639]  Russell Bryant <russell@russellbryant.com>

	* include/asterisk/framehook.h: framehook.h: Fix some doc typos.
	  There were a number of instances in this header file where
	  "function all" was intended to be "function call". This patch
	  fixes that up.

2014-03-14 21:56 +0000 [r410626]  Mark Michelson <mmichelson@digium.com>

	* /, tests/test_sorcery_realtime.c: Fix failing realtime sorcery
	  tests. The store realtime callback needs to return a positive
	  value for sorcery to treat the store as a success. ........
	  Merged revisions 410625 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-14 21:36 +0000 [r410624]  Jonathan Rose <jrose@digium.com>

	* main/manager.c, /: manager: fix memory leak in manager_add_filter
	  function (closes issue ASTERISK-23420) Reported by: Etienne
	  Lessard Patches: manager_eventfilter_leak uploaded by Etienne
	  Lessard (license 6394) ........ Merged revisions 410609 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410623 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-14 20:55 +0000 [r410591-410608]  Mark Michelson <mmichelson@digium.com>

	* /, main/db.c: Remove an extra ast_cond_wait() that slipped
	  through the patch. ........ Merged revisions 410606 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410607 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/config.c, res/res_sorcery_realtime.c: Handle the return
	  values of realtime updates and stores more accurately. Realtime
	  backends' update and store callbacks return the number of rows
	  affected, or -1 if there was a failure. There were a couple of
	  issues: * The config API was treating 0 as a successful return,
	  and positive values as a failure. Now the config API treats
	  anything >= 0 as a success. * res_sorcery_realtime was treating 0
	  as a successful return from the store procedure, and any positive
	  values as a failure. Now sorcery treats anything > 0 as a
	  success. It still considers 0 a "failure" since there is no
	  change to report to observers. Review:
	  https://reviewboard.asterisk.org/r/3341 ........ Merged revisions
	  410592 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_mwi.c: Prevent conflicts regarding unsolicited
	  and solicited MWI to an endpoint. If an endpoint is receiving
	  unsolicited MWI for a mailbox and then attempts to subscribe to
	  an AOR that provides MWI for the same mailbox, then the SUBSCRIBE
	  is rejected with a 500 response. Review:
	  https://reviewboard.asterisk.org/r/3345 ........ Merged revisions
	  410590 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-14 17:56 +0000 [r410589]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, CHANGES: uniqueid: Update CHANGES to reflect new features Note
	  the new features provided by uniqueid in the CHANGES file. (issue
	  ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3316/
	  ........ Merged revisions 410588 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-14 16:42 +0000 [r410575]  Jonathan Rose <jrose@digium.com>

	* /, main/acl.c, res/res_pjsip/pjsip_configuration.c,
	  contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py,
	  CHANGES, res/res_pjsip/config_transport.c,
	  include/asterisk/acl.h: PJSIP: TOS values should be represented
	  as decimals in sorcery objects (closes issue ASTERISK-23235)
	  Reported by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3324/ ........ Merged
	  revisions 410574 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-14 16:19 +0000 [r410567]  Mark Michelson <mmichelson@digium.com>

	* /, main/db.c: Prevent delayed astdb syncs. The syncing thread
	  sleeps for a second before waiting to be told to attempt to sync
	  again. If a signal were sent during this sleeping period, we
	  would end up having to wait until the next sync signal occurred
	  in order to sync up the astdb. This code rearrangement also
	  ensures that any pending transactions will be synced prior to
	  Asterisk shutting down. Patches: db_sync.patch by John Hardin
	  (License #6512) ........ Merged revisions 410556 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410559 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-14 16:17 +0000 [r410560]  Jonathan Rose <jrose@digium.com>

	* res/ari/resource_bridges.c, /: ARI/bridges: Forward
	  Playback/Recording Started/Finished to bridge topic (closes issue
	  ASTERISK-23444) Reported by: Ben Merrills Review:
	  https://reviewboard.asterisk.org/r/3340/ ........ Merged
	  revisions 410558 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-14 16:01 +0000 [r410542-410557]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/app.h, /, res/res_mwi_external.c, main/app.c:
	  res_mwi_external: Clear the stasis cache entry when the external
	  MWI is deleted. One of the things missing when external MWI
	  support was added was the ability to clear the stasis cache entry
	  of deleted external MWI mailboxes. Review:
	  https://reviewboard.asterisk.org/r/3325/ ........ Merged
	  revisions 410555 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/cdr.c: cdr.c: Add missing aow_unlock(cdr) in off nominal
	  path of handle_dial_message(). * Trivial common code hoisting in
	  handle_bridge_leave_message(). * Some whitespace fixing. ........
	  Merged revisions 410541 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-13 19:33 +0000 [r410528]  Kinsey Moore <kmoore@digium.com>

	* res/stasis/control.h, res/res_stasis.c, /, res/stasis/control.c:
	  ARI: Ensure managing application receives ChannelEnteredBridge
	  messages This fixes an issue where a Stasis application running
	  over ARI and subscribed to ari/events could miss the
	  ChannelEnteredBridge event because it did not subscribe to the
	  new bridge fast enough. To accomplish this, it subscribes the
	  application controlling the channel to the new bridge before
	  adding it to that bridge which required the stasis_app_control
	  structure to maintain a reference to the stasis_app. (closes
	  issue ASTERISK-23295) Review:
	  https://reviewboard.asterisk.org/r/3336/ ........ Merged
	  revisions 410527 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-13 13:25 +0000 [r410511]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_multihomed.c, /: Multiple revisions 410509-410510
	  ........ r410509 | file | 2014-03-13 06:23:14 -0700 (Thu, 13 Mar
	  2014) | 2 lines res_pjsip_multihomed: Fix a bug where the 200 OK
	  for a REGISTER would contain the wrong contact. ........ r410510
	  | file | 2014-03-13 06:24:17 -0700 (Thu, 13 Mar 2014) | 2 lines
	  res_pjsip_multihomed: Remove change for testing fix. ........
	  Merged revisions 410509-410510 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-12 19:06 +0000 [r410492-410494]  Richard Mudgett <rmudgett@digium.com>

	* res/res_musiconhold.c, main/channel.c, /: res_musiconhold.c:
	  Generate MOH start/stop events whenever the MOH stream is
	  started/stopped. * Made res_musiconhold.c always post the
	  MusicOnHoldStart/MusicOnHoldStop events when it actually
	  starts/stops the music streams. This allows the events to always
	  happen when MOH starts/stops. The event posting code was moved to
	  the MOH alloc/release routines. * Made channel_do_masquerade()
	  stop any MOH on the original channel before masquerading so the
	  original channel will get a stop event with correct information.
	  * Cleaned up a couple odd codings in moh_files_alloc() and
	  moh_alloc() dealing with the music state variable. (issue
	  ASTERISK-23311) Reported by: Benjamin Keith Ford Review:
	  https://reviewboard.asterisk.org/r/3306/ ........ Merged
	  revisions 410493 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/confbridge/conf_state.c,
	  apps/confbridge/conf_state_single.c,
	  apps/confbridge/conf_state_inactive.c,
	  apps/confbridge/conf_state_single_marked.c, /: app_confbridge:
	  Make explicitly stop MOH if a user is kicked or hangs up while
	  MOH is playing. When MOH is playing to a user in a conference and
	  the user is kicked or hangs up from the conference then the AMI
	  MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event:
	  MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported
	  by: Benjamin Keith Ford Review:
	  https://reviewboard.asterisk.org/r/3306/ ........ Merged
	  revisions 410490 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410491 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-12 12:51 +0000 [r410452-410472]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Fix a bug
	  where outgoing messages for TCP would go out using UDP. This
	  change fixes a bug where the code which changes the transport did
	  not check whether the message is going out over UDP or not before
	  changing it. For TCP and TLS transports we don't need to change
	  the transport as the correct one is already chosen. ........
	  Merged revisions 410471 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_multihomed.c (added), /: res_pjsip_multihomed: Add
	  module which places the correct address within messages. Due to
	  how messages are handled within PJSIP it is not until a message
	  is actually sent that the destination is reliably known. This
	  means that the addresses placed within the message may not be of
	  the interface the message is being sent out on. This module
	  determines what interface a message is being sent on and updates
	  the message to contain the correct address if applicable. This
	  module was tested by myself in a virtualized environment with
	  multiple interfaces and also by Kinsey Moore in the following
	  configuration: Networks: * 10.24.16.0/21 ** hard phone ** default
	  gateway * 10.24.64.0/21 ** softphone with pjsip-based stack
	  Transport details: bind address: 0.0.0.0 protocol: UDP All
	  endpoints were tested with explicitly configured transports and
	  unconfigured transports. This was tested with inbound and
	  outbound calls, both of which were experiencing detrimental
	  effects from incorrect IP addresses in SIP messages. These
	  effects were only experienced by the soft phone on the 10.24.64.0
	  network since the messages to the hard phone on the 10.24.16.0
	  network had the correct IP address. (closes issue ASTERISK-23020)
	  Reported by: xrobau Review:
	  https://reviewboard.asterisk.org/r/3102/ ........ Merged
	  revisions 410451 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-10 17:21 +0000 [r410395]  Richard Mudgett <rmudgett@digium.com>

	* /, main/http.c: AST-2014-001: Stack overflow in HTTP processing
	  of Cookie headers. Sending a HTTP request that is handled by
	  Asterisk with a large number of Cookie headers could overflow the
	  stack. Another vulnerability along similar lines is any HTTP
	  request with a ridiculous number of headers in the request could
	  exhaust system memory. (closes issue ASTERISK-23340) Reported by:
	  Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
	  Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions
	  410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 410381 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410383 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-10 16:33 +0000 [r410369]  Scott Griepentrog <sgriepentrog@digium.com>

	* res/ari/resource_channels.c, main/manager.c, /: unqiueid: correct
	  max uniqueid length test This patch adds null string test prior
	  to checking for a max uniqueid value that was added in r410157.
	  ........ Merged revisions 410368 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-10 13:30 +0000 [r410346]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
	  session timers request This change allows chan_sip to avoid
	  creation of the channel and consumption of associated file
	  descriptors altogether if the inbound request is going to be
	  rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
	  Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
	  Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
	  Corey Farrell (license 5909) ........ Merged revisions 410308
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 410311 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410329 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-10 12:53 +0000 [r410307]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c: AST-2014-003:
	  res_pjsip: When handling 401/407 responses don't assume a request
	  will have an endpoint. This change removes the assumption that an
	  outgoing request will always have an endpoint and makes the
	  authenticate_qualify option work once again. (closes issue
	  ASTERISK-23210) Reported by: Joshua Colp ........ Merged
	  revisions 410306 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-08 16:50 +0000 [r410288]  George Joseph <george.joseph@fairview5.com>

	* res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c,
	  res/res_pjsip_outbound_registration.c,
	  res/res_pjsip_endpoint_identifier_ip.c,
	  include/asterisk/res_pjsip_cli.h, include/asterisk/sorcery.h,
	  res/res_pjsip/pjsip_cli.c, res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip/config_transport.c, main/sorcery.c,
	  include/asterisk/res_pjsip.h: pjsip_cli: Create pjsip show
	  channel and contact, and general cli code cleanup. Created the
	  'pjsip show channel' and 'pjsip show contact' commands.
	  Refactored out the hated ast_hashtab. Replaced with
	  ao2_container. Cleaned up function naming. Internal only, no
	  public name changes. Cleaned up whitespace and brace formatting
	  in cli code. Changed some NULL checking from "if"s to
	  ast_asserts. Fixed some register/unregister ordering to reduce
	  deadlock potential. Fixed ast_sip_location_add_contact where the
	  'name' buffer was too short. Fixed some self-assignment issues in
	  res_pjsip_outbound_registration. (closes issue ASTERISK-23276)
	  Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged
	  revisions 410287 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-08 15:45 +0000 [r410275]  Matthew Jordan <mjordan@digium.com>

	* /, res/ari/resource_channels.c: resource_channels: Check if a
	  passed in ID is NULL before checking its length Calling strlen on
	  a NULL string is explosive. This patch checks whether or not the
	  passed in string is NULL or zero length before checking to see if
	  the string is too long. ........ Merged revisions 410274 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 22:56 +0000 [r410227]  Corey Farrell <git@cfware.com>

	* /, channels/chan_sip.c: chan_sip: Fix deadlock of monlock between
	  unload_module and do_monitor Release monlock before calling
	  pthread_join. This ensures do_monitor cannot freeze by locking
	  monlock during module unload. (closes issue ASTERISK-21406)
	  Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3284/ ........ Merged
	  revisions 410224 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 410225 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410226 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 22:08 +0000 [r410212]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, include/asterisk/sorcery.h: sorcery: correct field register
	  argument list This fixes mistakes I previously made in merging
	  gtjoseph's changes with mine. ........ Merged revisions 410211
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 21:54 +0000 [r410208-410210]  Matthew Jordan <mjordan@digium.com>

	* /, main/config_options.c: config_options: Display the see-also
	  information for CLI config option help The config option help
	  information has always parsed the <see-also> tags in the XML
	  documentation. Unfortunately, it just never bothered displaying
	  them on the CLI. With this patch, when you execute 'config show
	  help [module] [obj] [option]', it will display what other options
	  are useful to you. (closes issue ASTERISK-22008) Reported by:
	  Richard Mudgett ........ Merged revisions 410209 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip.c, /: res_pjsip: Fix documentation for one touch
	  recording see-also links The one touch recording options have
	  several see-also links between the various configuration options.
	  These were 'broken' by the snake casing of those options. This
	  patch corrects the see-also links such that they reference the
	  correct option names. ........ Merged revisions 410194 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 21:23 +0000 [r410207]  Mark Michelson <mmichelson@digium.com>

	* main/sorcery.c, res/res_sorcery_realtime.c, /,
	  include/asterisk/sorcery.h, tests/test_sorcery_realtime.c: Make
	  res_sorcery_realtime filter unknown retrieved results. When
	  retrieving data from a database or other realtime backend, it's
	  quite possible to retrieve variables that Asterisk does not care
	  about but that are legitimate to exist. Asterisk does not need to
	  throw a hissy fit when these variables are encountered but rather
	  just filter them out. Review:
	  https://reviewboard.asterisk.org/r/3305 ........ Merged revisions
	  410187 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 21:11 +0000 [r410191]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/sorcery.c, /, include/asterisk/sorcery.h,
	  res/res_pjsip/pjsip_configuration.c: pjsip: allow and disallow
	  show same codecs In order to prevent confusion over the allow and
	  disallow list of codecs being the same an option for registering
	  a field as an alias is added. The alias field will be read from
	  the configuration file, but afterwards is not listed as a known
	  field. With disallow set as an alias, the CLI command pjsip show
	  endpoint # will list the allow= field, but not the disallow
	  field. (closes issue ASTERISK-23092) Review:
	  https://reviewboard.asterisk.org/r/3193/ ........ Merged
	  revisions 410190 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 20:41 +0000 [r410174-410185]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/devicestate.h, main/stasis_cache.c,
	  main/stasis_message.c, /, tests/test_devicestate.c,
	  include/asterisk/stasis.h, main/app.c, main/devicestate.c,
	  tests/test_stasis.c: stasis cache: Enhance to keep track of an
	  item from different entities. A stasis cache entry now contains
	  more than a single message/snapshot. It contains
	  messages/snapshots for the local entity as well as any remote
	  entities that post to the cached item. In addition callbacks can
	  be supplied when the cache is created to compute and post the
	  aggregate message/snapshot representing all entities stored in
	  the cache entry. * All stasis messages now have an eid to
	  indicate what entity posted it. * The stasis cache enhancements
	  allow device state to cache and aggregate the device states from
	  local and remote entities in a single operation. The cached
	  aggregate device state is available immediately after it is
	  posted to the stasis bus. This improves performance by
	  eliminating a cache dump and associated ao2 container traversals
	  to calculate the aggregate state. (closes issue ASTERISK-23204)
	  Reported by: Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/3281/ ........ Merged
	  revisions 410184 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* tests/test_cel.c, channels/sig_pri.c, channels/sig_ss7.c,
	  include/asterisk/bridge.h, tests/test_cdr.c, channels/sig_pri.h,
	  channels/chan_dahdi.c, channels/sig_ss7.h, /: uniqueid: Fix
	  chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler
	  errors. (issue ASTERISK-23120) ........ Merged revisions 410171
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 15:47 +0000 [r410158]  Scott Griepentrog <sgriepentrog@digium.com>

	* tests/test_cdr.c, res/res_clioriginate.c, res/res_ari_bridges.c,
	  tests/test_substitution.c, res/res_stasis_playback.c,
	  channels/chan_multicast_rtp.c, apps/app_meetme.c, /,
	  main/bridge_basic.c, include/asterisk/channel_internal.h,
	  tests/test_app.c, apps/confbridge/conf_chan_record.c,
	  main/core_unreal.c, channels/chan_gtalk.c,
	  include/asterisk/stasis_app_playback.h,
	  res/ari/resource_bridges.c, channels/chan_jingle.c,
	  channels/chan_phone.c, pbx/pbx_spool.c,
	  res/ari/resource_bridges.h, res/parking/parking_tests.c,
	  channels/chan_motif.c, apps/app_confbridge.c,
	  res/ari/resource_channels.c, include/asterisk/pbx.h,
	  res/res_stasis.c, include/asterisk/bridge.h,
	  apps/app_voicemail.c, res/ari/resource_channels.h,
	  apps/app_dial.c, res/res_calendar_exchange.c,
	  channels/chan_vpb.cc, apps/app_page.c, apps/app_chanisavail.c,
	  include/asterisk/dial.h, main/core_local.c,
	  res/parking/parking_bridge_features.c,
	  tests/test_stasis_endpoints.c, res/parking/parking_bridge.c,
	  channels/chan_skinny.c, include/asterisk/stasis_app_snoop.h,
	  addons/chan_mobile.c, main/bridge_channel.c,
	  channels/chan_pjsip.c, channels/chan_mgcp.c,
	  channels/chan_unistim.c, main/pbx.c,
	  res/res_calendar_icalendar.c, main/ccss.c,
	  channels/chan_bridge_media.c, main/bridge.c,
	  tests/test_stasis_channels.c, apps/app_bridgewait.c,
	  apps/app_originate.c, res/res_calendar_caldav.c,
	  include/asterisk/channel.h, res/parking/parking_applications.c,
	  apps/app_followme.c, main/cel.c, apps/app_queue.c,
	  res/res_ari_channels.c, res/res_calendar_ews.c,
	  rest-api/api-docs/bridges.json, main/dial.c,
	  channels/chan_dahdi.c, channels/chan_h323.c, tests/test_cel.c,
	  rest-api/api-docs/channels.json,
	  include/asterisk/bridge_internal.h,
	  apps/confbridge/conf_chan_announce.c, res/res_calendar.c,
	  include/asterisk/core_unreal.h, addons/chan_ooh323.c,
	  res/stasis/control.c, channels/chan_sip.c,
	  main/channel_internal_api.c, include/asterisk/stasis_app.h,
	  res/res_stasis_snoop.c, channels/chan_console.c,
	  channels/chan_iax2.c, channels/chan_oss.c, apps/app_agent_pool.c,
	  main/channel.c, main/manager.c, channels/chan_misdn.c,
	  tests/test_voicemail_api.c, channels/chan_alsa.c,
	  channels/chan_nbs.c, main/message.c: uniqueid: channel linkedid,
	  ami, ari object creation with id's Much needed was a way to
	  assign id to objects on creation, and much change was necessary
	  to accomplish it. Channel uniqueids and linkedids are split into
	  separate string and creation time components without breaking
	  linkedid propgation. This allowed the uniqueid to be specified by
	  the user interface - and those values are now carried through to
	  channel creation, adding the assignedids value to every function
	  in the chain including the channel drivers. For local channels,
	  the second channel can be specified or left to default to a ;2
	  suffix of first. In ARI, bridge, playback, and snoop objects can
	  also be created with a specified uniqueid. Along the way, the
	  args order to allocating channels was fixed in chan_mgcp and
	  chan_gtalk, and linkedid is no longer lost as masquerade occurs.
	  (closes issue ASTERISK-23120) Review:
	  https://reviewboard.asterisk.org/r/3191/ ........ Merged
	  revisions 410157 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-07 05:04 +0000 [r410108]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: chan_sip: Allow static realtime members
	  to be qualified during module load. When a static realtime peer
	  with qualify=yes is loaded, Asterisk will fail to send an OPTIONS
	  request due to the lastms being equal to 0. This results in the
	  peer being unable to receive calls from Asterisk because the
	  status is permanently UNKNOWN. This patch allows an OPTIONS
	  request to be sent during module load by ignoring the lastms
	  value on startup only. Review:
	  https://reviewboard.asterisk.org/r/3294/ (closes issue
	  ASTERISK-17523) Reported by: Maciej Krajewski Tested by:
	  wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor
	  Peirce (license 6112) ........ Merged revisions 410105 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 410106 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410107 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-06 23:47 +0000 [r410092]  Richard Mudgett <rmudgett@digium.com>

	* main/sorcery.c, /: sorcery.c: Fix off-nominal path ref and memory
	  leak in ast_sorcery_objectset_json_create(). * Made exit a loop
	  early on error in ast_sorcery_objectset_json_create(). * Removed
	  some dead code in ast_sorcery_objectset_create2(). ........
	  Merged revisions 410089 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-06 23:43 +0000 [r410091]  Russell Bryant <russell@russellbryant.com>

	* /, res/res_musiconhold.c: moh: fix a refcount error with realtime
	  MOH I observed a crash in res_musiconhold on an Asterisk 11
	  system using realtime MOH. Investigation of the backtrace showed
	  a corrupt mohclass, implying that it got destroyed before the
	  code expected it to. I went looking for reference counting errors
	  that could have caused this crash and this patch this result. It
	  contains 2 changes. 1) Remove a usless block of code that was
	  impossible to reach. There was even a comment indicating that it
	  was impossible to reach. The conditional includes
	  "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's
	  inside of an if block with the opposite check
	  "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no
	  good reason to keep it around. 2) A similar block to #1 contained
	  a reference counting error. It stores state->class in the local
	  variable mohclass without increasing its reference count. The
	  reference count on mohclass is decremented at the end of the
	  function. This block of code probably very rarely runs, which
	  would help explain why this system was working fine for many
	  months before experiencing a crash. Review:
	  https://reviewboard.asterisk.org/r/3282/ ........ Merged
	  revisions 410043 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 410044 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 410090 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-06 22:39 +0000 [r410042]  George Joseph <george.joseph@fairview5.com>

	* res/res_pjsip/config_auth.c, funcs/func_sorcery.c (added),
	  res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c,
	  main/bucket.c, res/res_pjsip_endpoint_identifier_ip.c,
	  include/asterisk/config.h, include/asterisk/sorcery.h,
	  res/res_pjsip/pjsip_configuration.c, res/res_pjsip_acl.c,
	  CHANGES, tests/test_sorcery.c, res/res_pjsip/config_transport.c,
	  main/config.c, main/sorcery.c: sorcery: Create AST_SORCERY
	  dialplan function. This patch creates the AST_SORCERY dialplan
	  function which allows someone to retrieve any value from a
	  sorcery-based config file. It's similar to AST_CONFIG. The
	  creation of the function itself was fairly straightforward but it
	  required changes to the underlying sorcery infrastructure that
	  rippled into individual sorcery objects. The changes stemmed from
	  inconsistencies in how sorcery created ast_variable objectsets
	  from sorcery objects and the inconsistency in how individual
	  objects used that feature especially when it came to parameters
	  that can be specified multiple times like contact in aor and
	  match in identify. You can read more here...
	  http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html
	  So, what this patch does, besides actually creating the
	  AST_SORCERY function, is the following... * Creates
	  ast_variable_list_append which is a helper to append one
	  ast_variable list to another. * Modifies the
	  ast_sorcery_object_field_register functions to accept the
	  already-defined sorcery_fields_handler callback. * Modifies
	  ast_sorcery_objectset_create to accept a parameter indicating
	  return type preference...a single ast_variable with all values
	  concatenated or an ast_variable list with multiple entries. Also
	  fixed a few bugs. * Modifies individual sorcery object
	  implementations to use the new function definition of the
	  ast_sorcery_object_field_register functions. * Modifies
	  location.c and res_pjsip_endpoint_identifier_ip.c to implement
	  sorcery_fields_handler handlers so they return multiple
	  occurrences as an ast_variable_list. * Added a whole bunch of
	  tests to test_sorcery. (closes issue ASTERISK-22537) Review:
	  http://reviewboard.asterisk.org/r/3254/

2014-03-06 19:04 +0000 [r410029]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/acl.h, /, main/acl.c,
	  res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
	  contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py
	  (added), res/res_pjsip/config_transport.c: pjsip configuration:
	  Make transport TOS values consistent with endpoints Transport TOS
	  values were interpreted as DSCP values without being documented
	  as such. Endpoint TOS values (tos_audio/tos_video) behaved
	  normally as TOS values have historically. This patch makes the
	  transport TOS values behave as TOS values and makes all TOS
	  values readable as string values (e.g. AF11). In addition,
	  alembic scripts have been updated to use the proper field types
	  for all TOS/COS values. (issue ASTERISK-23235) Reported by:
	  George Joseph Review: https://reviewboard.asterisk.org/r/3304/
	  ........ Merged revisions 410028 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-06 18:20 +0000 [r410027]  Joshua Colp <jcolp@digium.com>

	* res/ari/resource_channels.c, CHANGES,
	  res/ari/ari_model_validators.c,
	  rest-api/api-docs/recordings.json, res/ari/resource_bridges.c,
	  res/ari/ari_model_validators.h, /,
	  include/asterisk/stasis_app_recording.h,
	  res/res_stasis_recording.c: res_stasis_recording: Add a
	  "target_uri" field to recording events. This change adds a
	  target_uri field to the live recording object. It contains the
	  URI of what is being recorded. (closes issue ASTERISK-23258)
	  Reported by: Ben Merrills Review:
	  https://reviewboard.asterisk.org/r/3299/ ........ Merged
	  revisions 410025 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-06 15:58 +0000 [r410012]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_mwi.c, /: Don't attempt to link in an aggregate MWI
	  subscription if an endpoint does not aggregate MWI. Attempting to
	  link a NULL object into an ao2 container had been benign
	  previously, but since enabling DO_CRASH in the testsuite, this is
	  now causing a crash. It's better to be right here anyway.
	  ........ Merged revisions 410011 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-06 02:22 +0000 [r409996]  Matthew Jordan <mjordan@digium.com>

	* res/res_fax_spandsp.c, /: res_fax_spandsp: Fix crash when passing
	  ulaw/alaw data to spandsp When acting as a T.38 fax gateway,
	  res_fax_spandsp would at times cause a crash in libspandsp. This
	  would occur when, during fax tone detection, a ulaw/alaw frame
	  would be passed to modem_connect_tones_rx. That particular
	  routine expects the data to be in slin format. This patch looks
	  at the frame type and, if the data is ulaw/alaw, converts the
	  format to slin before passing it to modem_connect_tones_rx.
	  Review: https://reviewboard.asterisk.org/r/3296 (closes issue
	  ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal
	  Rybarik patches: spandsp_g711decode.diff uploaded by Michal
	  Rybarik (license 6578) ........ Merged revisions 409990 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409991 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-06 00:33 +0000 [r409970-409977]  Richard Mudgett <rmudgett@digium.com>

	* apps/confbridge/conf_state_multi.c,
	  apps/confbridge/conf_state_inactive.c, /: app_confbridge: Remove
	  some noop code. ........ Merged revisions 409976 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_musiconhold.c: res_musiconhold.c: Remove some
	  unnecessary RAII_VAR() usage. * Made the moh_register() define
	  use useful parameter names. ........ Merged revisions 409967 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-05 20:41 +0000 [r409904-409919]  Kinsey Moore <kmoore@digium.com>

	* main/config.c, /: config: Fix inverted test The test of the
	  result of the stat() call was inverted such that its output was
	  only used if the call failed. This inverts the test so that the
	  output of stat() is used correctly. This was causing full reloads
	  on unchanged files. (closes issue ASTERISK-23383) Reported by:
	  David Woolley ........ Merged revisions 409916 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409917 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409918 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* bridges/bridge_native_rtp.c, /: bridge_native_rtp: Fix crash
	  involving masquerade It is possible for a channel to be
	  masqueraded out of a bridge which means it may no longer have RTP
	  glue to check upon leaving said bridge. If this situation
	  occurred (it's possible at least during dial and call pickup)
	  then Asterisk would crash. This change makes sure the glue is
	  checked before use. (closes issue AST-1290) Reported by: John
	  Bigelow ........ Merged revisions 409900 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-05 18:51 +0000 [r409889]  Richard Mudgett <rmudgett@digium.com>

	* contrib/ast-db-manage/cdr/versions,
	  contrib/ast-db-manage/cdr/versions/210693f3123d_create_cdr_table.py,
	  /,
	  contrib/ast-db-manage/config/versions/28887f25a46f_create_queue_tables.py
	  (added), contrib/ast-db-manage/cdr.ini.sample (added),
	  contrib/ast-db-manage/cdr/env.py, contrib/ast-db-manage/cdr
	  (added), contrib/ast-db-manage/cdr/script.py.mako: alembic: Add
	  missing queue and CDR table creation scripts. * Added the queues
	  and queue_members tables to the config alembic scripts. * Added
	  the CDR table alembic creation script. The CDR table is more of
	  an example for new setups since the actual table can be fully
	  customized in cdr_adaptive_odbc.conf. (closes issue
	  ASTERISK-23233) Reported by: jmls Review:
	  https://reviewboard.asterisk.org/r/3227/ ........ Merged
	  revisions 409885 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-05 18:47 +0000 [r409888]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_presencestate.c, /: Fix documentation for
	  PRESENCE_STATE to properly illustrate how to create a presence
	  hint. There was a missing comma. This was discovered by Dan
	  Kaplan. ........ Merged revisions 409886 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409887 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-05 16:58 +0000 [r409836]  David M. Lee <dlee@digium.com>

	* main/config.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Corrected cross-platform stat nanosecond code When
	  nanosecond time resolution was added for identifying config file
	  changes, it didn't cover all of the myriad of ways that one might
	  obtain nanosecond time resolution off of struct stat. Rather than
	  complicate the #if even further figuring out one system from the
	  next, this patch directly tests for the three struct members I
	  know about today, and #ifdef's accordingly. Review:
	  https://reviewboard.asterisk.org/r/3273/ ........ Merged
	  revisions 409833 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409834 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409835 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-05 16:26 +0000 [r409831-409832]  Moises Silva <moises.silva@gmail.com>

	* res/res_http_websocket.c: Fix res/res_http_websocket.c build
	  failure in 32bit due to incorrect print format for uint64_t

	* res/res_http_websocket.c, /: Fix WebRTC over WSS not working
	  Several fixes for the WebSockets implementation in
	  res/res_http_websocket.c * Flush the websocket session FILE* as
	  fwrite() may not actually guarantee sending the data to the
	  network. If we do not flush, it seems that buffering on the SSL
	  socket for outbound messages causes issues * Refactored
	  ast_websocket_read to take into account that SSL file descriptors
	  may be ready to read via fread() but poll() will not actually say
	  so because the data was already read from the network buffers and
	  is now in the libc buffers (closes issue ASTERISK-23099) (closes
	  issue ASTERISK-21930) Review:
	  https://reviewboard.asterisk.org/r/3248/ ........ Merged
	  revisions 409681 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409697 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-05 12:06 +0000 [r409780]  Sean Bright <sean@malleable.com>

	* contrib/scripts/astgenkey, contrib/scripts/astgenkey.8, /: Fix
	  references to 'keys' CLI commands in astgenkey ........ Merged
	  revisions 409777 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409778 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409779 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-05 06:17 +0000 [r409747]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Add update_peer function to
	  unistim_rtp_glue, improve other unistim_rtp_glue functions
	  conforming to other channel drivers. Do not forget auto-detected
	  and user-selected phone settings on 'unistim reload' ........
	  Merged revisions 409705 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409745 from
	  http://svn.asterisk.org/svn/asterisk/branches/11

2014-03-05 01:05 +0000 [r409683]  Richard Mudgett <rmudgett@digium.com>

	* /, include/asterisk/stasis_internal.h: stasis: Made
	  internal_stasis_subscribe() prototype and definition match
	  exactly. ........ Merged revisions 409682 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-04 19:34 +0000 [r409627]  Michael L. Young <elgueromexicano@gmail.com>

	* funcs/func_audiohookinherit.c, /: func_audiohookinheritance:
	  Check If A Channel Was Specified This patch prevents a crash when
	  using the function audiohookinheritance without setting the
	  channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal
	  Tested by: Joel Vandal Patches:
	  asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/3272/ ........ Merged
	  revisions 409623 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409625 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409626 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-04 17:22 +0000 [r409587]  Jonathan Rose <jrose@digium.com>

	* /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix one way audio
	  problems with hold/unhold when using ICE ICE sessions will now be
	  restarted if sessions are changed to use new sets of remote
	  candidates. (closes issue ASTERISK-22911) Reported by: Vytis
	  Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/
	  ........ Merged revisions 409565 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409570 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-04 16:55 +0000 [r409569]  Kinsey Moore <kmoore@digium.com>

	* /, main/astobj2.c: AO2: Add an assert for bad objects This adds
	  an assert that will only be active if Asterisk is compiled with
	  DO_CRASH and allows the testsuite to fail tests that would
	  otherwise require log file parsing. ........ Merged revisions
	  409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 409567 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409568 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-04 14:55 +0000 [r409475]  Sean Bright <sean@malleable.com>

	* /, channels/chan_sip.c: Minor whitespace change to 'sip show
	  peers' output. (closes issue ASTERISK-23406) Reported by: ibercom
	  Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom
	  ........ Merged revisions 409472 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409473 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409474 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-03 19:44 +0000 [r409423]  Joshua Colp <jcolp@digium.com>

	* /, res/res_stasis_recording.c: res_stasis_recording: Fix memory
	  leak of the absolute name. ........ Merged revisions 409422 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-03 02:08 +0000 [r409364]  Matthew Jordan <mjordan@digium.com>

	* main/asterisk.c, /: doxygen: Tweak the link back to ye olde
	  Digium website ........ Merged revisions 409361 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409362 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409363 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-02 17:03 +0000 [r409350]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* /, Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a
	  legal option of gcc. Unofficially gcc considers it to be
	  equivalent of -O3. clang chalks on it, though. This commit sets
	  the default optimization flag to be -O3, like gcc actually
	  considered it. Review: https://reviewboard.asterisk.org/r/3280/
	  ........ Merged revisions 409308 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409344 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409346 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-01 20:28 +0000 [r409288]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_session.c, /: res_pjsip_session: Set options
	  (100rel, timers) on incoming sessions. This change passes options
	  to the UAS creation function. This in turn sets up 100rel and
	  session timer properties on the incoming session. Reported by
	  Julian Russell on asterisk-users mailing list. ........ Merged
	  revisions 409287 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-03-01 00:05 +0000 [r409257-409275]  Richard Mudgett <rmudgett@digium.com>

	* /, main/devicestate.c: devicestate.c: Simplified some logic in
	  _ast_device_state(). ........ Merged revisions 409274 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis_cache.c, /: stasis_cache.c: Remove some unnecessary
	  RAII_VAR() usage. ........ Merged revisions 409272 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis.c, /: stasis.c: Misc code cleanups. * Remove some
	  unnecessary RAII_VAR() usage. * Made the struct
	  stasis_subscription ao2 object use the ao2 lock instead of a
	  redundant join_lock in the struct for ast_cond_wait(). * Removed
	  locks on some ao2 objects that don't need the lock. * Made the
	  topic pool entries container use the ao2 template functions. *
	  Add some missing allocation failure checks. * Add missing cleanup
	  in off nominal path of dispatch_message(). ........ Merged
	  revisions 409270 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: chan_sip: Add precautionary p->owner
	  checks. * Add precautionary p->owner checks in sip_hangup(),
	  get_refer_info(), get_also_info(), and
	  interpret_t38_parameters(). * Simplify some tangled logic in
	  get_refer_info(), get_also_info(), and add_rpid(). * Removed some
	  dead code in handle_request_invite(). (closes issue
	  ASTERISK-23323) Reported by: Walter Doekes Patches:
	  issueA23323-more_p_owner_checks-1.8.x.patch (license #5674)
	  uploaded by wdoekes (modified)
	  issueA23323-more_p_owner_checks-11.x.patch (license #5674)
	  uploaded by wdoekes (modified)
	  issueA23323-more_p_owner_checks-12.x.patch (license #5674)
	  uploaded by wdoekes (modified)
	  issueA23323-more_p_owner_checks-trunk.patch (license #5674)
	  uploaded by wdoekes (modified) ........ Merged revisions 409207
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 409255 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409256 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-28 21:24 +0000 [r409237]  Kinsey Moore <kmoore@digium.com>

	* apps/app_queue.c, /: app_queue: Fix documented AMI event name
	  During the rewrite of AMI events to use the Stasis bus, the name
	  of the QueueMemberPaused event was changed to QueueMemberPause.
	  This corrects documentation to reflect that. ........ Merged
	  revisions 409234 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-28 18:03 +0000 [r409159]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: chan_sip: Fix crash in
	  ast_channel_hangupcause_set(). * Fix crash in
	  ast_channel_hangupcause_set() because p->owner not checked before
	  calling. Regression introduced by the fix for ASTERISK-22621.
	  (closes issue ASTERISK-23135) Reported by: OK (issue
	  ASTERISK-23323) Reported by: Walter Doekes ........ Merged
	  revisions 409156 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409157 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409158 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-27 19:54 +0000 [r409132]  Jonathan Rose <jrose@digium.com>

	* res/res_rtp_asterisk.c, /: Multiple revisions 409129-409130
	  ........ r409129 | jrose | 2014-02-27 13:19:02 -0600 (Thu, 27 Feb
	  2014) | 15 lines res_rtp_asterisk: Fix checklist creating
	  problems in ICE sessions Prior to this patch, local candidate
	  lists including SRFLX would fail to start properly when building
	  ICE candidate check lists. This patch fixes that problem by
	  making sure that each SRFLX candidate is associated with the
	  proper base address so that the check list can create matches
	  properly. This patch was written by jcolp. The issue will be left
	  open to await testing by the issue participants. (issue
	  ASTERISK-23213) Reported by: Andrea Suisani Review:
	  https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose
	  | 2014-02-27 13:38:10 -0600 (Thu, 27 Feb 2014) | 8 lines
	  res_rtp_asterisk: correct build error from r409129 Accidentally
	  placed a declaration below functional code (issue ASTERISK-23213)
	  Reported by: Andrea Suisani Review:
	  https://reviewboard.asterisk.org/r/3256/ ........ Merged
	  revisions 409129-409130 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409131 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-27 16:26 +0000 [r409091]  David M. Lee <dlee@digium.com>

	* utils/astman.c, /: Fix memory stomping bug in astman. This memset
	  complained in dev mod on my Ubuntu box. The memset is both
	  unnecessary and dangerous. At this point, m hasn't been
	  initialized yet, so the memset will write off to whatever address
	  happens to be on the stack at the time. ........ Merged revisions
	  409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 409083 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409087 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-27 16:08 +0000 [r409055]  Corey Farrell <git@cfware.com>

	* /, configs/res_fax.conf.sample: res_fax: Comment out default
	  settings from res_fax.conf. Comment out many settings in
	  res_fax.conf.sample. The defaults are set in res_fax.c, so
	  setting the same value in sample config does nothing but make the
	  sample config more fragile. (closes issue ASTERISK-23231)
	  Reported by: David Brillert Review:
	  https://reviewboard.asterisk.org/r/3261/ ........ Merged
	  revisions 409052 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 409053 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 409054 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-27 12:29 +0000 [r409000]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Apply
	  packetization rules on inbound SDP handling The setting
	  'use_ptime' is supposed to tell Asterisk to honour the ptime
	  attribute in an offer, preferring it to whatever packetization
	  preferences have been set internally. Currently, however,
	  something rather quirky will happen: (1) The SDP answer will be
	  constructed in create_outgoing_sdp_stream. This will use the
	  preferences from the endpoint, such that the 200 OK response will
	  add the packetization preferences from the endpoint, and not what
	  was offered. (2) When the 200 response is issued,
	  apply_negotiated_sdp_stream is called. This will call
	  apply_packetization, which will use the ptime attribute from the
	  offer internally. We end up telling the offerer to use the
	  internal ptime attribute, but we end up using the offered ptime
	  attribute. Hilarity ensues. This patch modifies the behaviour by
	  calling apply_packetization from negotiate_incoming_sdp_stream,
	  which is called prior to create_outgoing_sdp_stream. This causes
	  the format preferences on the session's media object to be set to
	  the inbound ptime value (if 'use_ptime' is enabled), such that
	  the construction of the answer gets the right value immediately.
	  Review: https://reviewboard.asterisk.org/r/3244/ ........ Merged
	  revisions 408999 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-26 23:35 +0000 [r408984]  Richard Mudgett <rmudgett@digium.com>

	* /, tests/test_stasis.c: test_stasis.c: Misc cleanups. * Make the
	  consumer ao2 object use the ao2 lock instead of a redundant lock
	  in the struct for ast_cond_wait(). * Fixed some curly brace
	  placements. * Fixed use of malloc(0). malloc(0) has variant
	  behavior. It is up to the implementation to determine if it
	  returns NULL or a valid pointer that can be later passed to
	  free(). ........ Merged revisions 408983 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-26 19:00 +0000 [r408971]  Scott Griepentrog <sgriepentrog@digium.com>

	* channels/chan_pjsip.c, /: pjsip: avoid edge case potential crash
	  in answer() When accidentally compiling against a wrong version
	  of pjsip headers with a different pjsip_inv_session size, the
	  invite_tsx structure could be null in the answer() function. This
	  led to a crash because it attempted to send the session response
	  with an uninitialized packet pointer. This patch presets packet
	  to null and adds a diagnostic log message to explain why the call
	  fails. Review: https://reviewboard.asterisk.org/r/3267/ ........
	  Merged revisions 408970 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-26 17:04 +0000 [r408958]  Joshua Colp <jcolp@digium.com>

	* res/res_ari.c, /: res_ari: Make some additional error responses
	  consistent with the rest of the system. This change makes some
	  error cases use ast_ari_response_error to construct their error
	  responses instead of manually doing it. This ensures they are
	  consistent with the other error responses. Based on the original
	  patch as done by Paul Belanger on the associated review. Review:
	  https://reviewboard.asterisk.org/r/2904/ ........ Merged
	  revisions 408957 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-26 13:47 +0000 [r408942-408944]  Kinsey Moore <kmoore@digium.com>

	* include/asterisk/res_pjsip_session.h, /: PJSIP: Fix some bad
	  spacing ........ Merged revisions 408943 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_refer.c: PJSIP: Prevent crash if channel has
	  gone away It is currently possible for an ast_sip_session to
	  exist without an associated channel as is the case when a new
	  invite is coming in or just after a hangup is issued on a
	  chan_pjsip channel. Part of the attended transfer code assumed
	  the channel would be non-NULL and used it as such causing a
	  crash. This bug was exposed thanks to the attended transfer ARI
	  test in the test suite. (closes issue ASTERISK-23287) Reported
	  by: Matt Jordan ........ Merged revisions 408941 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-26 08:57 +0000 [r408932]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Implement functions handling keypress,
	  display icons and text for i2004 KEM support.

2014-02-25 17:51 +0000 [r408881-408883]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_exten_state.c, /,
	  res/res_pjsip_pidf_digium_body_supplement.c (added),
	  include/asterisk/res_pjsip_body_generator_types.h:
	  res_pjsip_exten_state: Presence for digium phones Added presence
	  support for digium phones. Review:
	  https://reviewboard.asterisk.org/r/3239/ ........ Merged
	  revisions 408882 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_send_to_voicemail.c (added),
	  res/res_pjsip_header_funcs.c: res_pjsip_send_to_voicemail:
	  transferring to voicemail for digium phones Added the ability for
	  transferring directly to voicemail on digium phones. Added a new
	  module that checks for the presence of a custom header and/or
	  diversion header within a sip REFER. If either is found and they
	  specify a sending to voicemail action then variables are added to
	  the channel allowing the user access to them in the dialplan.
	  Dialplan can then be written that branches based upon these
	  values allowing, for instace, for a single number to be used for
	  dialing and/or accessing voicemail directly. Also fixed a problem
	  where the PJSIP_HEADER function was allowing non pjsip channels
	  through (checked to make sure it has the correct channel type
	  before proceeding). Review:
	  https://reviewboard.asterisk.org/r/3245/ ........ Merged
	  revisions 408880 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-25 17:44 +0000 [r408879]  Rusty Newton <rnewton@digium.com>

	* configs/voicemail.conf.sample, /: configs/voicemail.conf.sample -
	  Make mailcmd sample text more explicit Made the wording a bit
	  more explicit. Didn't really change the meaning. ........ Merged
	  revisions 408876 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408877 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408878 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-22 23:31 +0000 [r408859]  Matthew Jordan <mjordan@digium.com>

	* /, main/asterisk.c: main: Initialize dialplan providing core
	  components prior to module pre-load It is possible to pre-load
	  pbx_config. As a result, pbx_config - which will load and parse
	  the dialplan - will attempt to use various dialplan components,
	  such as device state providers and presence state providers,
	  prior to them being initialized by the core. This would lead to a
	  crash, as the components had not created their Stasis cache
	  entries. This patch moves a number of core component
	  initializations before the module pre-load. This guarantees that
	  if someone does pre-load pbx_config - or other pbx modules - that
	  the Stasis caches for the various core components are created.
	  (closes issue ASTERISK-23320) Reported by: xrobau (closes issue
	  ASTERISK-23265) Reported by: Andrew Nagy Tested by: Andrew Nagy,
	  Rusty Newton ........ Merged revisions 408855 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-22 18:01 +0000 [r408840]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: ignore AST_CONTROL_PVT_CAUSE_CODE
	  without any messages (closes issue ASTERISK-23336) Reported by:
	  Alexander Semych ........ Merged revisions 408838 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408839 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-22 02:31 +0000 [r408788]  Corey Farrell <git@cfware.com>

	* /, utils/extconf.c, utils/conf2ael.c, res/ael/pval.c, main/pbx.c:
	  Remove extra defines of AST_PBX_MAX_STACK. * Ensure
	  AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix
	  incorrect function parameters in utils/extconf.c. (closes issue
	  ASTERISK-23141) Reported by: Maxim Review:
	  https://reviewboard.asterisk.org/r/3241/ ........ Merged
	  revisions 408785 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408786 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408787 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-21 18:37 +0000 [r408731]  Kevin Harwell <kharwell@digium.com>

	* main/rtp_engine.c, /: rtp_engine: Dynamic payload change in rtp
	  mapping not supported Asterisk didn't support the dynamic payload
	  change in rtp mapping in the 200 OK response. Scenario: Asterisk
	  sends the INVITE proposing alaw and telephone-event, it proposes
	  rtpmap:101 for telephone-event. Peer responds with 2xx, it
	  answers with alaw and telephone-event also, but it proposes a
	  different rtpmap number (rtpmap:103) for telephone-event.
	  Expected Behaviour: Asterisk should honour the rtpmapping in the
	  response and send DTMF packets using 103 as payload type for
	  DTMF. Actual Behaviour: Asterisk sends DTMF packets using payload
	  type 101. With this patch asterisk now supports changes that can
	  occur in the rtp mapping in the response. (closes issue
	  ASTERISK-23279) Reported by: NITESH BANSAL Review:
	  https://reviewboard.asterisk.org/r/3225/ Patches:
	  dynamic_payload_change.patch uploaded by nbansal (license 6418)
	  ........ Merged revisions 408729 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408730 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-21 18:19 +0000 [r408712-408723]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, /: manager: Fix AMI Status action of a single
	  channel. Fixed use of uninitialized ao2 container iterator in an
	  off-nominal condition. Either a memory allocation error or the
	  requested channel is an internal channel not exposed to the
	  outside. ........ Merged revisions 408715 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/sorcery.c, res/ari/resource_endpoints.c, /,
	  apps/app_meetme.c, res/res_fax.c, res/res_stasis_recording.c,
	  main/stasis_channels.c, res/res_sorcery_astdb.c,
	  include/asterisk/json.h: json: Fix off-nominal json ref counting
	  issues. * Fixed off-nominal json ref counting issue with using
	  the following API calls: ast_json_object_set() and
	  ast_json_array_append(). * Fixed off-nominal error reporting in
	  ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal
	  json ref counting issues in report_receive_fax_status() and
	  dial_to_json(). ........ Merged revisions 408713 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/json.c, /: json: Fix json API wrapper code for json library
	  versions earlier than 2.3.0. * Fixed json ref counting issue with
	  json API wrapper code for ast_json_object_update_existing() and
	  ast_json_object_update_missing() when the json library is earlier
	  than version 2.3.0. ........ Merged revisions 408711 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-21 16:49 +0000 [r408699]  Corey Farrell <git@cfware.com>

	* channels/chan_sip.c: chan_sip: prevent add_route from adding
	  empty header. Fix regression caused by ASTERISK-22582. Empty
	  Route headers were added when the route had a single strict hop.
	  (closes issue ASTERISK-23306) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3236/

2014-02-21 16:27 +0000 [r408645-408652]  Kevin Harwell <kharwell@digium.com>

	* main/rtp_engine.c, /: rtp_engine: Output mixup in
	  ${CHANNEL(rtpqos,audio,all)} Fixed the output of
	  CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter.
	  (closes issue ASTERISK-23261) Reported by: rsw686 Patches:
	  rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged
	  revisions 408646 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408647 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408649 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/channel.c, /: channel.c: MOH is not working for transferee
	  after attended transfer Updated the code to check to see if MOH
	  is playing on the transferor and if so then start it on the
	  channel that replaces it during a masquerade. Example scenario of
	  the problem: Alice calls Bob and then Bob begins the attended
	  transfer process into a queue. Upon going on hold Alice hears
	  music and so does Bob once he is in the queue. Bob then transfers
	  Alice into the queue and then music for Alice stops even though
	  she should be hearing it since has now replaced Bob in the queue.
	  The problem that was occurring is that once the channel was
	  masqueraded the app (queues, confbridge, etc...) had no way of
	  knowing that the channel had just been swapped out thus it did
	  not start music for the present channel. Credit to Olle Johansson
	  for pointing me in the right direction on this issue. (closes
	  issue ASTERISK-19499) Reported by: Timo Teräs Review:
	  https://reviewboard.asterisk.org/r/3226/ ........ Merged
	  revisions 408642 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408643 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408644 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-21 10:45 +0000 [r408592]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay
	  variables ........ Merged revisions 408589 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408590 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408591 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-21 00:50 +0000 [r408539]  Michael L. Young <elgueromexicano@gmail.com>

	* /, apps/app_chanspy.c: app_chanspy: Documentation Update To
	  Clarify "x" Option When using the "x" option (specify a DTMF
	  digit to exit the application), it is not obvious in the
	  documentation that this only works when spying on a channel. If a
	  channel being used to spy on other channels is waiting to connect
	  to a channel or is no longer attached to a channel, the DTMF is
	  ignored. As noted on the issue tracker, since there are
	  workarounds available and this is a rarely used option we are
	  opting for a documentation change here. (closes issue
	  ASTERISK-22661) Reported by: Chris Hillman Patches:
	  asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2990/ ........ Merged
	  revisions 408536 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408537 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408538 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-20 21:12 +0000 [r408519-408523]  George Joseph <george.joseph@fairview5.com>

	* /, res/res_pjsip/location.c,
	  res/res_pjsip_outbound_registration.c: pjsip_cli: Add pjsip
	  commands 'show registrations' and 'show contacts'. Added 'show
	  registrations' and 'show contacts' to pjsip cli to make things a
	  little more consistent. The output is exactly the same as the
	  list command. Just needed to add entries to their respective
	  ast_cli_entry structures. (closes issue ASTERISK-23275) Review:
	  http://reviewboard.asterisk.org/r/3210/ ........ Merged revisions
	  408522 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip/pjsip_cli.c, main/config.c: pjsip_cli: Fix
	  memory leak in ast_sip_cli_print_sorcery_objectset. Fixed memory
	  leaks in ast_sip_cli_print_sorcery_objectset and
	  ast_variable_list_sort. (closes issue ASTERISK-23266) Review:
	  http://reviewboard.asterisk.org/r/3200/ ........ Merged revisions
	  408520 from http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/sorcery.h,
	  res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c,
	  tests/test_sorcery.c, main/sorcery.c, /,
	  res/res_pjsip/config_system.c: sorcery: Create sorcery instance
	  registry. In order to retrieve an arbitrary sorcery instance from
	  a dialplan function (or any place else) there needs to be a
	  registry of sorcery instances. ast_sorcery_init now creates a
	  hashtab as a registry. ast_sorcery_open now checks the hashtab
	  for an existing sorcery instance matching the caller's module
	  name. If it finds one, it bumps the refcount and returns it. If
	  not, it creates a new sorcery instance, adds it to the hashtab,
	  then returns it. ast_sorcery_retrieve_by_module_name is a new
	  function that does a hashtab lookup by module name. It can be
	  called by the future dialplan function. res_pjsip/config_system
	  needed a small change to share the main res_pjsip sorcery
	  instance. tests/test_sorcery was updated to include a test for
	  the registry. (closes issue ASTERISK-22537) Review:
	  http://reviewboard.asterisk.org/r/3184/ ........ Merged revisions
	  408518 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-20 19:02 +0000 [r408503]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip.c, /: res_pjsip: Update documentation for
	  'use_avpf' option When 'use_avpf' is set to True, inbound offers
	  must use the AVPF/SAVPF RTP profile. However, when 'use_avpf' is
	  set to False, Asterisk will accept both AVP/SAVP or AVPF/SAVPF
	  RTP profiles in inbound offers. The documentation previously
	  implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was
	  set to False and a UA offered said profile in an INVITE request.
	  ........ Merged revisions 408502 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-20 02:44 +0000 [r408450]  Rusty Newton <rnewton@digium.com>

	* /, apps/app_queue.c: apps/app_queue - Fix incorrect Macro
	  parameter documentation Macro is executed on the called channel,
	  not the calling channel. (closes issue ASTERISK-23069) Reported
	  By: Bryan Anderson ........ Merged revisions 408447 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408448 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408449 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-19 19:09 +0000 [r408386-408390]  Richard Mudgett <rmudgett@digium.com>

	* /, main/config.c: config: Add file size and nanosecond resolution
	  fields to the cached modified config file information. Repeatedly
	  modifying config files and reloading too fast sometimes fails to
	  reload the configuration because the cached modification
	  timestamp has one second resolution. * Added file size and
	  nanosecond resolution fields to the cached config file
	  modification timestamp information. Now if the file size changes
	  or the file system supports nanosecond resolution the modified
	  file has a better chance of being detected for reload. * Added a
	  missing unlock in an off-nominal code path. (closes issue
	  AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
	  ........ Merged revisions 408387 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408388 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408389 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix regex
	  handling and keep simple prefix matching performance. The sorcery
	  astDB wizzard does not handle regex correctly if the pattern
	  begins with an anchor character. This patch attempts to convert
	  the anchored regex pattern to a prefix pattern supported by astDB
	  for performance reasons. If it is not able to convert the pattern
	  it falls back to getting all astDB members of the family and
	  doing a normal regex pattern matching on the retrieved records.
	  Review: https://reviewboard.asterisk.org/r/3161/ ........ Merged
	  revisions 408385 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-19 12:04 +0000 [r408315-408332]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooCapability.c, /,
	  addons/ooh323c/src/ooh245.c: process receiveAndTransmit user
	  input remote caps instead of receive only send receiveAndTransmit
	  user input our caps instead of receive only ........ Merged
	  revisions 408328 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408330 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408331 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* addons/ooh323c/src/ooh323.c, /: Allow different socket and
	  signalling ip on h.323 connection if gk mode is active Reported
	  by: Gabriele Odone Patches: ASTERISK-22738-1.patch Tested by:
	  Gabriele Odone (closes issue ASTERISK-22738) ........ Merged
	  revisions 408312 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408314 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-18 19:19 +0000 [r408299]  Richard Mudgett <rmudgett@digium.com>

	* contrib/ast-db-manage/config/env.py,
	  contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py,
	  contrib/ast-db-manage/config,
	  contrib/ast-db-manage/voicemail/env.py,
	  contrib/ast-db-manage/voicemail,
	  contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py,
	  contrib/ast-db-manage/config/versions,
	  contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py,
	  contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py,
	  contrib/ast-db-manage/voicemail/versions, contrib/ast-db-manage,
	  /: alembic: Add svn:ignore *.pyc to directories and
	  svn:executable to *.py files. ........ Merged revisions 408297
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-17 15:36 +0000 [r408272]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip/location.c, UPGRADE.txt, res/res_pjsip.c,
	  res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h: Store
	  SIP User-Agent information in contacts. When an endpoint sends a
	  REGISTER request to Asterisk, we now will associate the
	  User-Agent header with all contacts that were bound in that
	  REGISTER request. ........ Merged revisions 408270 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-16 03:25 +0000 [r408199-408227]  Matthew Jordan <mjordan@digium.com>

	* /, main/pbx.c: pbx: Handle a completely empty dialplan during a
	  context merge It is highly unlikely, but - at least in Asterisk
	  12 - theoretically possible to load Asterisk with no dialplan
	  whatsoever. If that occurs, and some other module (that is not a
	  pbx module) attempts to merge its contexts into the dialplan, the
	  existing merge routine will crash. This is because it is not
	  insane, and rightly believes that you provided some sort of
	  dialplan, somewhere. This patch will gracefully merge the
	  contexts in such a case. Note that this is highly unlikely to
	  occur in 1.8/11, as features will most likely provide some
	  dialplan via parking. However, in Asterisk 12, parking is now
	  provided by res_parking, and hence may create its dialplan later.
	  (closes issue ASTERISK-23297) Reported by: CJ Oster Review:
	  https://reviewboard.asterisk.org/r/3222 ........ Merged revisions
	  408200 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 408201 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408220 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, Makefile: buildsystem: Unbreak the build (infloop) on Asterisk
	  11+ Apparently r408084 ( https://reviewboard.asterisk.org/r/3212/
	  ) broke the build. This patch fixes it by ignoring the .lastclean
	  dependencies if the MENUSELECT_EMBED variable is not defined.
	  patches: tmp.diff uploaded by wdoekes (License 5674) Review:
	  https://reviewboard.asterisk.org/r/3228/ ........ Merged
	  revisions 408193 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408194 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-14 21:44 +0000 [r408139-408141]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/stasis_endpoints.c, /: ARI: correct upper/lower case URI
	  discrepancies URI's are supposed to be case sensitive and all
	  lower case. In practice some portions of URI's in ARI are case
	  insensitive and others are not, such as TECH, which in one
	  instance would match a lower case name and in another would not.
	  In this patch, the ast_endpoint_lastest_snapshot() function is
	  modified to change the TECH portion to full upper case before
	  lookup. This resolves the discrepancy noted by the reporter.
	  However I chose to avoid forcing the /ari prefix of the URI's to
	  be lower case for now. Except for the two cases here, all URI's
	  should be lower case, unless they are part of a resource name or
	  id. Review: https://reviewboard.asterisk.org/r/3211/ Reported by:
	  Zane Conkle (closes issue ASTERISK-23125) ........ Merged
	  revisions 408140 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/format.c, /: format.c: correct possible null pointer
	  dereference In ast_format_sdp_parse and ast_format_sdp_generate
	  the check checks for a valid interface and function were
	  potentially confusing, and hid an error in the test of the
	  presence of the function that is called later. This patch clears
	  up and corrects the test. Review:
	  https://reviewboard.asterisk.org/r/3208/ (closes issue
	  ASTERISK-23098) Reported by: marcelloceschia Patches:
	  main_format.patch uploaded by marcelloceschia (license 6036)
	  ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
	  ........ Merged revisions 408137 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408138 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-14 13:31 +0000 [r408086]  Walter Doekes <walter+asterisk@wjd.nu>

	* Makefile, /: buildsystem: Don't force main to depend on
	  everything else. Directory 'main' only needs to depend on
	  embedded modules. If no module embedding is selected, the
	  dependency is dropped. Review:
	  https://reviewboard.asterisk.org/r/3212/ ........ Merged
	  revisions 408083 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 408084 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 408085 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-14 12:41 +0000 [r408070]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: chnan_sip: Set SIP_DEFER_BYE_ON_TRANSFER
	  prior to calling bridge blind transfer This patch moves setting
	  SIP_DEFER_BY_ON_TRANSFER prior to calling
	  ast_bridge_transfer_blind. This prevents a BYE from being sent
	  prior to the NOTIFY request that informs the transferor if the
	  transfer succeeded or failed. This patch also clears said flag
	  from the off nominal NOTIFY paths in the local_attended_transfer
	  code, as once we've sent the NOTIFY request it is safe to send by
	  the BYE request. This was caught by the
	  blind-transfer-accountcode test in the Asterisk Test Suite.
	  (closes issue ASTERISK-23290) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3214/ ........ Merged
	  revisions 408069 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-14 08:52 +0000 [r408059]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* Makefile, build_tools/install_subst (added): install_subst:
	  helper script for installing with path substitution A helper
	  script to copy a source file substituting any
	  __ASTERISK_<foo>_DIR__ with the content of $AST<foo>DIR. Review:
	  https://reviewboard.asterisk.org/r/3202/

2014-02-13 18:52 +0000 [r407990-408006]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pubsub.c, /, res/res_pjsip_mwi.c: Remove all PJSIP
	  MWI-specific use from our MWI code. PJSIP has built-in MWI code
	  that could be useful to some degree, but our utilization of the
	  API actually made our code a bit more cluttered since we had to
	  have special cases peppered throughout. With this change, we move
	  to using the pjsip_evsub API instead, which streamlines the code
	  by removing special cases. Review:
	  https://reviewboard.asterisk.org/r/3205 ........ Merged revisions
	  408005 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip/location.c: Fix crash in AMI PJSIPShowEndpoint
	  action. If an AOR has no permanent contacts, then the
	  permanent_contacts container is never allocated. This makes the
	  code safe in the face of NULLs. I also changed the variable that
	  counts contacts from "num" to "total_contacts" since there are
	  now two variables that are indicate numbers of things. ........
	  Merged revisions 407988 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-13 15:51 +0000 [r407989]  Kinsey Moore <kmoore@digium.com>

	* main/logger.c, CHANGES: Logger: Add dynamic logger channels This
	  adds the ability to dynamically add and remove logger channels
	  from Asterisk via the CLI. (closes issue AST-1150) Review:
	  https://reviewboard.asterisk.org/r/3185/

2014-02-12 08:25 +0000 [r407970]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, main/config.c: realtime: Fix ast_update2_realtime() on
	  raspberry pi. The old code depended on undefined va_arg
	  behaviour: calling a function twice with the same va_list
	  parameter and expecting it to continue where it left off. The
	  changed code behaves like the manpage says it should. Also added
	  a bunch of early returns to trap errors (e.g. OOM) instead of
	  crashing. The problem was found by Julian Lyndon-Smith. The
	  deviant behaviour on the raspberry PI also uncovered another bug
	  (fixed in r407875) in the res_config_pgsql.so driver. Reported
	  by: jmls Tested by: jmls Review:
	  https://reviewboard.asterisk.org/r/3201/ ........ Merged
	  revisions 407968 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-11 20:17 +0000 [r407958]  Joshua Colp <jcolp@digium.com>

	* main/sched.c: scheduler: Remove hashtab usage. This is a first
	  stab at tweaking the performance profile of the scheduler.
	  Removing the hashtab usage removes an extra memory allocation
	  when scheduling something and makes it so rescheduling does not
	  incur any memory allocation at all. Review:
	  https://reviewboard.asterisk.org/r/3199/

2014-02-11 03:18 +0000 [r407940]  Matthew Jordan <mjordan@digium.com>

	* res/ari/resource_channels.c, /: ari/resource_channels: Add
	  channel variables earlier in the creation process This patch
	  tweaks the behaviour of POST /channels with channel variables
	  such that the variables are passed into the pbx.c routines that
	  perform the origination. This allows the variables to be assigned
	  to the newly created channels immediately upon their
	  construction, as opposed to be assigned after the originate has
	  completed. The upshot of this is that the variables are available
	  on the channels if they execute in the dialplan, as opposed to
	  only being available once the channels are answered. Review:
	  https://reviewboard.asterisk.org/r/3183/ ........ Merged
	  revisions 407937 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-10 18:28 +0000 [r407926]  Corey Farrell <git@cfware.com>

	* channels/sip/include/reqresp_parser.h,
	  channels/sip/include/route.h (added), channels/chan_sip.c,
	  channels/sip/route.c (added), channels/sip/include/sip.h:
	  chan_sip: Isolate code that manages struct sip_route. * Move
	  route code to sip/route.c + sip/include/route.h * Rename
	  functions to sip_route_* * Replace ad-hoc list code with macro's
	  from linkedlists.h * Create sip_route_process_header() to
	  processes Path and Record-Route headers (previously done with
	  different code in build_route and build_path) * Add use of const
	  where possible * Move struct uriparams, struct contact and
	  contactliststruct from sip.h to reqresp_parser.h. sip/route.c
	  uses reqresp_parser.h but not sip.h, this was a problem. These
	  moved declares are not used outside of reqresp_parser. * While
	  modifying reqprep() the lack of {} caused me trouble. I added
	  them. * Code outside route.c treats sip_route as an opaque
	  structure, using macro's or procedures for all access. (closes
	  issue ASTERISK-22582) Reported by: Corey Farrell Review:
	  https://reviewboard.asterisk.org/r/3173/

2014-02-10 16:49 +0000 [r407876]  Walter Doekes <walter+asterisk@wjd.nu>

	* res/res_config_pgsql.c, /: res_config_pgsql: Fix
	  ast_update2_realtime calls. Fix so multiple updates from a single
	  call works (add missing ','). Remove bogus ast_free's that
	  weren't supposed to be there. Moved a few spaces for readability.
	  Review: https://reviewboard.asterisk.org/r/3194/ ........ Merged
	  revisions 407873 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407874 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407875 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-10 16:01 +0000 [r407859]  Kinsey Moore <kmoore@digium.com>

	* apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c,
	  apps/confbridge/conf_state_empty.c,
	  apps/confbridge/conf_config_parser.c,
	  configs/confbridge.conf.sample, /,
	  apps/confbridge/include/confbridge.h, UPGRADE.txt: ConfBridge:
	  Correct prompt playback target Currently, when the first marked
	  user enters the conference that contains waitmarked users, a
	  prompt is played indicating that the user is being placed into
	  the conference. Unfortunately, this prompt is played to the
	  marked user and not the waitmarked users which is not very
	  helpful. This patch changes that behavior to play a prompt
	  stating "The conference will now begin" to the entire conference
	  after adding and unmuting the waitmarked users since the design
	  of confbridge is not conducive to playing a prompt to a subset of
	  users in a conference in an asynchronous manner. (closes issue
	  PQ-1396) Review: https://reviewboard.asterisk.org/r/3155/
	  Reported by: Steve Pitts ........ Merged revisions 407857 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407858 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-07 20:52 +0000 [r407767]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_iax2.c: chan_iax2: Add some more iaxs[] NULL
	  checks to a routine already full of them. ........ Merged
	  revisions 407764 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407765 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407766 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-07 20:17 +0000 [r407752]  Matthew Jordan <mjordan@digium.com>

	* /, main/security_events.c: security_events: Fix assertion failure
	  in dev-mode on optional IE parsing When formatting an optional
	  IE, the value is, of course, optional. As such, it is entirely
	  appropriate for ast_json_object_get to return NULL. If that
	  occurs, we now simply skip the IE that was requested, as it was
	  not provided by the entity that raised the event. Thanks to
	  George Joseph (gtjoseph) for catching this and reporting it in
	  #asterisk-dev ........ Merged revisions 407750 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-07 20:01 +0000 [r407749]  Joshua Colp <jcolp@digium.com>

	* main/timing.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
	  res/res_timing_timerfd.c, include/asterisk/timing.h,
	  res/res_timing_kqueue.c: timing: Improve performance for most
	  timing implementations. This change allows timing implementation
	  data to be stored directly on the timer itself thus removing the
	  requirement for many implementations to do a container lookup for
	  the same information. This means that API calls into timing
	  implementations can directly access the information they need
	  instead of having to find it. Review:
	  https://reviewboard.asterisk.org/r/3175/

2014-02-07 19:40 +0000 [r407748]  Matthew Jordan <mjordan@digium.com>

	* /, funcs/func_cdr.c: funcs/func_cdr: Handle empty time values
	  when extracting parsed values When extracting timestamps that are
	  parsed, time stamp values that are not set (time values of
	  0.000000) should not actually result in a parsed string. The
	  value should be skipped, and the result of the CDR function
	  should be an empty string. Prior to this patch, the result was
	  fed to the time formatting, which would result in an output of a
	  date/time in 1969. ........ Merged revisions 407747 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-07 18:29 +0000 [r407731]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c, include/asterisk/frame.h,
	  configs/iax.conf.sample, /: chan_iax2: Block unnecessary control
	  frames to/from the wire. Establishing an IAX2 call between
	  Asterisk v1.4 and v1.8 (or later) results in an unexpected call
	  disconnect. The problem happens because newer values in the enum
	  ast_control_frame_type are not consistent between the branch
	  versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later)
	  using IAX2 2) v1.8 answers and sends a connected line update
	  control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4
	  receives the control frame as an end-of-q (on v1.4
	  AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the
	  receive queue becomes empty. Several things are done by this
	  patch to fix the problem and attempt to prevent it from happening
	  again in the future: * Added a warning at the definition of enum
	  ast_control_frame_type about how to add new control frame values.
	  * Made block sending and receiving control frames that have no
	  reason to go over the wire. * Extended the connectedline iax.conf
	  parameter to also include the redirecting information updates. *
	  Updated the connectedline iax.conf parameter documentation to
	  include a notice that the parameter must be "no" when the peer is
	  an Asterisk v1.4 instance. (closes issue AST-1302) Review:
	  https://reviewboard.asterisk.org/r/3174/ ........ Merged
	  revisions 407678 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407727 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407729 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-07 16:47 +0000 [r407677]  Matthew Jordan <mjordan@digium.com>

	* /, main/security_events.c: security_events: Fix error caused by
	  DTD validation error The appdocsxml.dtd specifies that a
	  "required" attribute in a parameter may have a value of yes, no,
	  true, or false. On some systems, specifying "False" instead of
	  "false" would cause a validation error. This patch fixes the
	  casing to explicitly match the DTD. ........ Merged revisions
	  407676 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-07 13:15 +0000 [r407625]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* /, configs/indications.conf.sample: indications.conf: add stutter
	  tone; end properly * If the "stutter" (voicemail indication) tone
	  is indeed a stutter tone, and it ends with a constant tone, make
	  sure that it is the dial tone. This was done for India (in),
	  Mexico (mx) and the Philippines (ph). * If no "stutter" tone
	  exists for a country, provide one. This was done for Spain (es),
	  Malaysia (my) and Venezuela (ve). Review:
	  https://reviewboard.asterisk.org/r/3158/ ........ Merged
	  revisions 407622 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407623 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407624 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-06 21:24 +0000 [r407602]  Matthew Jordan <mjordan@digium.com>

	* /, main/security_events.c, UPGRADE.txt, CHANGES: security_events:
	  Add AMI documentation; output optional fields This patch adds
	  documentation for the Security Events that are emited over AMI.
	  It also notes these events in the UPGRADE/CHANGES file. ........
	  Merged revisions 407589 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-06 19:58 +0000 [r407588]  Rusty Newton <rnewton@digium.com>

	* /, configs/pjsip.conf.sample: configs/pjsip.conf.sample:
	  Configuration section naming in pjsip.conf.sample needs a little
	  clarification There is a bit of nuance to how you name things in
	  pjsip.conf. This is a documentation patch to at least clear it up
	  a little for users. Review:
	  https://reviewboard.asterisk.org/r/3180/ ........ Merged
	  revisions 407587 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-06 18:11 +0000 [r407574]  Kevin Harwell <kharwell@digium.com>

	* /,
	  contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py:
	  pjsip realtime: already created enum failure for postgresql If an
	  enum had been previously created the alembic script would attempt
	  to re-create it and an error would be generated while running
	  migrations for a postgresql server. The work around for this is
	  to use the ENUM object type for postgres as opposed to the
	  generic enum type used by sqlalchemy. Using this type in the
	  script seems to work properly for both postgres and mysql.
	  ........ Merged revisions 407572 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-06 17:55 +0000 [r407573]  Richard Mudgett <rmudgett@digium.com>

	* res/res_pjsip_logger.c,
	  res/res_pjsip/include/res_pjsip_private.h,
	  res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c,
	  include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
	  res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c,
	  res/res_pjsip_outbound_registration.c,
	  res/res_pjsip_endpoint_identifier_ip.c,
	  include/asterisk/res_pjsip_cli.h, res/res_pjsip/pjsip_cli.c,
	  res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip/config_domain_aliases.c: res_pjsip: Updates and
	  adds more PJSIP CLI commands. * Adds identify, transport, and
	  registration support to the PJSIP CLI. * Creates three additional
	  callbacks, one for an iterator, one for a comparator, and one for
	  a container. This eliminates the link dependency from higher
	  level modules to lower level ones. * Eliminates duplicate sorting
	  in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. *
	  Pushes CLI command registration down to the implementing source
	  file. * Adds several ast_sip_destroy_sorcery functions to
	  complement existing ast_sip_sorcery_initialize functions. The
	  destroy functions unregister PJSIP CLI commands and PJSIP CLI
	  formatters. Reported by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3104/ ........ Merged
	  revisions 407568 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-05 23:04 +0000 [r407514]  Rusty Newton <rnewton@digium.com>

	* /, formats/format_wav.c: formats/format_wav: enhancing log
	  message "Not a wav file" to be clear on what is supported
	  Modifying the log message to be more specific as to what is
	  supported. Specifically it seems format_wav supports only PCM
	  encoded versions with a lower-case '.wav' extension. (closes
	  issues ASTERISK-22310) Reported by: Jim Credland Review:
	  https://reviewboard.asterisk.org/r/3188/ ........ Merged
	  revisions 407511 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407512 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407513 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-05 20:56 +0000 [r407462]  Jonathan Rose <jrose@digium.com>

	* CHANGES, /: CHANGES: Improved description of Name/Creator changes
	  to bridge ARI, adds AMI The changes log was written with language
	  that was a little too internal Asterisk specific, so it's been
	  changed to be more in the frame of reference of an ARI user.
	  Also, previously the AMI event changes were omitted from the
	  change log as well as the ability to include a bridge name in the
	  ARI post bridges command. ........ Merged revisions 407461 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-05 20:43 +0000 [r407459]  Kinsey Moore <kmoore@digium.com>

	* main/logger.c, /: Logger: Fix handling of absolute paths This
	  fixes path handling for log files so that an extra / is not
	  appended to the file path when the path is absolute (begins with
	  /). This would previously result in different but functionally
	  equivalent paths in the output of 'logger show channels'.
	  ........ Merged revisions 407455 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407456 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407458 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-05 19:42 +0000 [r407443]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip/config_global.c, /: res_pjsip: When no global type
	  the debug option defaults to "yes" If the global section was not
	  specified in pjsip.conf then the configuration object does not
	  exist in sorcery so when retrieving "debug" option it would
	  return NULL. Then the NULL result was passed to ast_false utils
	  function which would return false because it wasn't set to some
	  representation of false, thus enabling sip debug logging. Made it
	  so if the global config object does not exist then it will return
	  a default of "no" for sip debugging. (issue ASTERISK-23038)
	  Reported by: Rusty Newton ........ Merged revisions 407442 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-05 17:42 +0000 [r407422-407425]  Jonathan Rose <jrose@digium.com>

	* CHANGES: CHANGES: Update changes log to include r403414 entry
	  Adds note of additional 0 for operator option on app_record

	* CHANGES, /: CHANGES: Update changes log to include new bridge
	  fields added in r404042 ........ Merged revisions 407419 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-05 15:29 +0000 [r407407]  Matthew Jordan <mjordan@digium.com>

	* rest-api/api-docs/playbacks.json, UPGRADE.txt,
	  rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES,
	  include/asterisk/manager.h, rest-api/api-docs/bridges.json,
	  rest-api/api-docs/deviceStates.json,
	  rest-api/api-docs/mailboxes.json,
	  rest-api/api-docs/asterisk.json,
	  rest-api/api-docs/applications.json,
	  rest-api/api-docs/channels.json,
	  rest-api/api-docs/recordings.json,
	  rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
	  /: ARI/AMI: Update versions; update UPGRADE/CHANGES notes for
	  12.1.0 changes Due to backwards compatible changes made to
	  AMI/ARI, the version needs to be bumped to 1.1.0/2.1.0,
	  respectively. ........ Merged revisions 407402 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-04 20:15 +0000 [r407275-407340]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/devicestate.h, /, main/devicestate.c:
	  devicestate: Make ast_devstate_changed_literal() return value and
	  doxygen consistent. Nothing actually cares about the value
	  anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose
	  ........ Merged revisions 407337 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407338 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407339 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix assertion
	  for pjsip.conf authorization list options. (closes issue
	  ASTERISK-23168) Reported by: George Joseph Review:
	  https://reviewboard.asterisk.org/r/3143/ ........ Merged
	  revisions 407324 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* configs/sip.conf.sample, main/tcptls.c, /: tcptls.c: Made TLS
	  handle a certificate chain file. Thanks to Guillaume Martres for
	  doing the necessary research to validate the change. (closes
	  issue ASTERISK-17727) Reported by: LN Patches:
	  use_certificate_chain.patch (license #5864) patch uploaded by st
	  documente_certificate_chain.patch (license #6576) patch uploaded
	  by Guillaume Martres ........ Merged revisions 407272 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407273 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407274 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-04 16:55 +0000 [r407260]  Matthew Jordan <mjordan@digium.com>

	* /, funcs/func_cdr.c: funcs/func_cdr: Fix non-epoch timestamps
	  broken by improper char array deref Thanks to snuffy for pointing
	  this issue out and fixing it. (closes issue ASTERISK-23250)
	  Reported by: snuffy patches: func_cdr-fix.diff uploaded by snuffy
	  (License 5024) ........ Merged revisions 407259 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-04 02:22 +0000 [r407217]  Joshua Colp <jcolp@digium.com>

	* res/res_clialiases.c, /: res_clialiases: Fix crash when reloading
	  and re-aliasing an alias that is in use. The code assumed that
	  unregistering the alias would always succeed while in practice
	  this is not actually true. A common case is the "reload" command
	  itself. If the cli_aliases.conf configuration file was changed
	  and reload executed the command would fail to unregister and
	  ultimately point to freed memory. The reload process now checks
	  whether unregistering succeeded or not and if not the old CLI
	  alias is retained. (closes issue ASTERISK-19773) Reported by:
	  Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth
	  Blades ........ Merged revisions 407205 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407210 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407213 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-04 02:07 +0000 [r407198]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Skinny - Fix deadlock when pickup of
	  no call. Locking issues in skinny when picking up a call that
	  doesn't exist. Cleaned up sub locking by fully removing and using
	  the chan lock instead. Also changed ast_call_pickup to check
	  whether chan was masq'd. (closes issue ASTERISK-23249) Reported
	  by: wedhorn Tested by: snuffy, myself Patches:
	  skinny-locking01.diff uploaded by wedhorn (license 5019) ........
	  Merged revisions 407197 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-03 01:31 +0000 [r407169]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /: cdrs: Check for applications to lock onto during
	  dial begin handling This patch brings CDR processing further in
	  line with r407085. During some dial operations, the application
	  would not be locked to the Dial application and would instead
	  continue to show the previously known application. In particular,
	  this would occur when a Parked call would time out. This was due
	  to a previous snapshot already locking the application to Park -
	  processing this in a Dial Begin allows the Dial application to
	  reassert its rightful place. (CDRs. Ugh.) But hooray for the
	  Parked Call tests for catching this in the Asterisk Test Suite.
	  ........ Merged revisions 407166 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-01 16:26 +0000 [r407154]  Joshua Colp <jcolp@digium.com>

	* res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /,
	  res/stasis/app.c, res/ari/ari_model_validators.c,
	  res/res_stasis.c, main/stasis_bridges.c: res_stasis: Enable
	  transfers and provide events when they occur. This change enables
	  transfers within ARI created bridges and adds events for when
	  they occur. Unlike other events these will be received if *any*
	  subscribed object is involved in the transfer. (closes issue
	  ASTERISK-22984) Reported by: David M. Lee Review:
	  https://reviewboard.asterisk.org/r/3120/ ........ Merged
	  revisions 407153 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-02-01 00:25 +0000 [r407105]  Corey Farrell <git@cfware.com>

	* apps/app_stack.c, /: app_stack: protect against missing
	  parameters to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2
	  parameters and LOCAL_PEEK requires 1 parameter. This protects
	  against situations where those parameters are blank or missing by
	  logging an error and returning. (closes issue ASTERISK-23220)
	  Reported by: James Sharp ........ Merged revisions 407100 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 407103 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407104 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-31 23:40 +0000 [r407083-407085]  Matthew Jordan <mjordan@digium.com>

	* apps/app_dial.c, main/cdr.c, main/pbx.c, /, main/bridge_after.c,
	  UPGRADE.txt, main/manager_channels.c: CDRs: fix a variety of dial
	  status problems, h/hangup handler creating CDRs This patch fixes
	  a number of small-ish problems that were noticed when witnessing
	  the records that the FreePBX dialplan produces: (1) Mid-call
	  events (as well as privacy options) have the ability to change
	  the overall state of the Dial operation after the called party
	  answers. This means that publishing the DialEnd event when the
	  called party is premature; we have to wait for the execution of
	  these subroutines to complete before we can signal the overall
	  status of the DialEnd. This patch moves that publication and adds
	  handlers for the mid-call events. (2) The AST_FLAG_OUTGOING
	  channel flag is cleared if an after bridge goto datastore is
	  detected. This flag was preventing CDRs from being recorded for
	  all outbound channels that had a 'continue' option enabled on
	  them by the Dial application. (3) The CDR engine now locks the
	  'Dial' application as being the CDR application if it detects
	  that the current CDR has entered that app. This is similar to the
	  logic that is done for Parking. In general, if we entered into
	  Dial, then we want that CDR to record the application as such -
	  this prevents pre-dial handlers, mid-call handlers, and other
	  shenaniganry from changing the application value. (4) The CDR
	  engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more
	  places to determine if the channel is in hangup logic or dead. In
	  either case, we don't want to record changes in the channel. (5)
	  The default option for "endbeforehexten" has been changed to
	  "yes". In general, you don't want to see CDRs in the 'h' exten or
	  in hangup logic. Since the semantics of that option changed in
	  12, it made sense to update the default value as well. (6)
	  Finally, because we now have the ability to synchronize on the
	  messages published to the CDR topic, on shutdown the CDR engine
	  will now synchronize to the messages currently in flight. This
	  helps to ensure that all in-flight CDRs are written before
	  shutting down. (closes issue ASTERISK-23164) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/3154 ........
	  Merged revisions 407084 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_dial.c, /: app_dial: Allow macro/gosub pre-bridge
	  execution to occur on priorities The parsing for the destination
	  of the macro/gosub uses the '^' character to separate out
	  context, extension, and priority. However, the logic for the
	  macro/gosub execution was written such that it would only do the
	  actual macro/gosub jump if a '^' character existed. This doesn't
	  apply when the macro/gosub jump occurs in a priority/priority
	  label. This patch changes the logic so that the parsing still
	  occurs, but the jump will occur even for priorities/priority
	  labels. (issue ASTERISK-23164) Review:
	  https://reviewboard.asterisk.org/r/3154 ........ Merged revisions
	  407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 407074 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 407082 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-31 23:15 +0000 [r407035-407037]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_logger.c, CHANGES, res/res_pjsip.c,
	  include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
	  contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py
	  (added), /, configs/pjsip.conf.sample, UPGRADE.txt: res_pjsip:
	  Config option to enable PJSIP logger at load time. Added a
	  "debug" configuration option for res_pjsip that when set to "yes"
	  enables SIP messages to be logged. It is specified under the
	  "system" type. Also added an alembic script to add the option to
	  realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton
	  Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged
	  revisions 407036 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_exten_state.c, /: res_pjsip_exten_state: Exporting
	  global symbols caused load order issues Removed the exportation
	  of global symbols from the module as it is no longer needed and
	  it could potentially cause load problems as on some systems it
	  would try to load before res_pjsip_pubsub ........ Merged
	  revisions 407034 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-31 23:04 +0000 [r407033]  Richard Mudgett <rmudgett@digium.com>

	* CHANGES, apps/app_chanspy.c: ChanSpy: Add ability to specify
	  channel uniqueids as well as channel names. * Made ChanSpy accept
	  a channel uniqueid or a fully specified channel name as the
	  chanprefix parameter if the 'u' option is specified. (closes
	  issue AFS-42) Review: https://reviewboard.asterisk.org/r/3160/

2014-01-31 22:39 +0000 [r407030-407032]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/res_pjsip_presence_xml.h (added), /: Add file
	  that apparently got missed in the merge. ........ Merged
	  revisions 407031 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_pidf_body_generator.c (added),
	  include/asterisk/res_pjsip_exten_state.h (removed),
	  res/res_pjsip_pubsub.exports.in, /,
	  include/asterisk/res_pjsip_body_generator_types.h (added),
	  res/res_pjsip_mwi.c, res/res_pjsip_xpidf_body_generator.c
	  (added), res/res_pjsip_mwi_body_generator.c (added),
	  res/res_pjsip_pubsub.c, res/res_pjsip_pidf.c (removed),
	  res/res_pjsip_pidf_eyebeam_body_supplement.c (added),
	  res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c
	  (added), include/asterisk/res_pjsip_pubsub.h: Decouple
	  subscription handling from NOTIFY/PUBLISH body generation. When
	  the PJSIP pubsub framework was created, subscription handlers
	  were required to state what event they handled along with what
	  body types they knew how to generate. While this serves well when
	  implementing a base RFC, it has problems when trying to extend
	  the body to support non-standard or proprietary body elements.
	  The code also was NOTIFY-specific, meaning that when the time
	  comes that we start writing code to send out PUBLISH requests
	  with MWI or presence bodies, we would likely find ourselves
	  duplicating code that had previously been written. This changeset
	  introduces the concept of body generators and body supplements. A
	  body generator is responsible for allocating a native structure
	  for a given body type, providing the primary body content,
	  converting the native structure to a string, and deallocating
	  resources. A body supplement takes the primary body content (the
	  native structure, not a string) generated by the body generator
	  and adds nonstandard elements to the body. With these elements
	  living in their own module, it becomes easy to extend our support
	  for body types and to re-use resources when sending a PUBLISH
	  request. Body generators and body supplements register themselves
	  with the pubsub core, similar to how subscription and publish
	  handlers had done. Now, subscription handlers do not need to know
	  what type of body content they generate, but they still need to
	  inform the pubsub core about what the default body type for a
	  given event package is. The pubsub core keeps track of what body
	  generators and body supplements have been registered. When a
	  SUBSCRIBE arrives, the pubsub core will check that there is a
	  subscription handler for the event in the SUBSCRIBE, then it will
	  check that there is a body generator that can provide the content
	  specified in the Accept header(s). Because of the nature of body
	  generators and supplements, it means res_pjsip_exten_state and
	  res_pjsip_mwi have been completely gutted. They no longer worry
	  about body types, instead calling
	  ast_sip_pubsub_generate_body_content() when they need to generate
	  a NOTIFY body. Review: https://reviewboard.asterisk.org/r/3150
	  ........ Merged revisions 407016 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-31 22:23 +0000 [r407015-407029]  Kevin Harwell <kharwell@digium.com>

	* contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py,
	  contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py,
	  /, UPGRADE.txt: alembic: script modifications due to errors A
	  couple of the scripts had errors that would not allow a full
	  migration to take place. The extensions table needed to make its
	  'id' column a primary key in order to work with mysql. The other
	  script ...add_endpoints... was missing tables that it was trying
	  to add columns to. Added the primary key on id for extensions and
	  added the tables in for the missing pjsip configuration options.
	  While it is not ideal to modify already released scripts this was
	  a case where it had to be done due to errors in the script and
	  lacking a better alternative. Review:
	  https://reviewboard.asterisk.org/r/3167/ ........ Merged
	  revisions 407019 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_mwi.c: res_pjsip_mwi: Subscribe fails when
	  missing aor name When subscribing to MWI (res_pjsip_mwi) and the
	  sip uri did not contain a name (ex: sip:<ip address>) then the
	  subscription would fail since it would be unable to locate an
	  associated aor. This patch makes it so that when a subscribe
	  comes with no aor name then it will subscribe to all aors on the
	  located endpoint. (closes issue ASTERISK-23072) Reported by: Bob
	  M Review: https://reviewboard.asterisk.org/r/3164/ ........
	  Merged revisions 407014 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-31 15:08 +0000 [r407001]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip_nat.c, /: PJSIP: Fix address for ACK in NAT
	  situations In NAT scenarios where a call is placed to a
	  Grandstream phone, res_pjsip will sometimes send the ACK to a 200
	  OK to the private address of the device behind the NAT instead of
	  the address of the NAT device. This corrects that behavior by
	  rewriting the address in the Contact header in the incoming 200
	  OK and the dialog's target address if necessary (since it has
	  already been rewritten to the incorrect private address). (closes
	  issue ASTERISK-23106) Review:
	  https://reviewboard.asterisk.org/r/3168/ Reported by: Matt Jordan
	  ........ Merged revisions 407000 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-31 05:31 +0000 [r406988]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Skinny: fix up possible double unlock
	  of chan. Return before chan is possibly unlocked a second time
	  when hanging up a channel in SUBSTATE_OFFHOOK. ........ Merged
	  revisions 406987 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-30 20:36 +0000 [r406936]  Corey Farrell <git@cfware.com>

	* main/udptl.c, res/res_rtp_asterisk.c, /: res_rtp_asterisk &
	  udptl: fix port selection to work with SELinux restrictions
	  ast_bind to a port reserved for another program by SELinux causes
	  errno == EACCES. This caused random failures when binding rtp or
	  udptl sockets. Treat EACCES as a non-fatal error, try next port.
	  (closes issue ASTERISK-23134) Reported by: Corey Farrell ........
	  Merged revisions 406933 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406934 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406935 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-30 17:35 +0000 [r406920]  Sean Bright <sean@malleable.com>

	* main/manager.c, /: Make a NOTICE about an invalid channel name
	  more useful. ........ Merged revisions 406918 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406919 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-29 00:44 +0000 [r406863]  Russell Bryant <russell@russellbryant.com>

	* /, configs/queues.conf.sample: queues.conf.sample Fix documented
	  default for persistentmembers Closes issue ASTERISK-22662
	  ........ Merged revisions 406860 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406861 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406862 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-28 23:40 +0000 [r406789-406848]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: potential crash on
	  timeout What seems to be happening is if a subscription has been
	  terminated and the subscription timeout/expires is less than the
	  time it takes for all pending transactions (currently on the
	  subscription) to end then the subscription timer will not have
	  been canceled yet and sub will be null. Since the subscription
	  has already been canceled nothing needs to be done so a null
	  check in the asterisk code is sufficient in working around this
	  problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins
	  ........ Merged revisions 406847 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* cdr/cdr_radius.c, cel/cel_radius.c, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: cdr_radius,
	  cel_radius: build agains libfreeradius-client Asterisk's RADIUS
	  module currently build against libradiusclient-ng, but this
	  project has been superseeded by libfreeradius-client. The API is
	  99% compatible except that the header name has changed, the
	  library name has changed, and the configuration file location has
	  changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé
	  Patches: freeradius-client.patch uploaded by sharky (license
	  6561) ........ Merged revisions 406801 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406802 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406803 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip/include/res_pjsip_private.h, /,
	  include/asterisk/compat.h: res_pjsip,compat: INFINITY and NAN
	  undefined On some systems the values for INFINITY and NAN are not
	  defined thus causing a build error on those systems. Added
	  definitions for those if they had not previously been defined.
	  (closes issue ASTERISK-23056) Reported by: capouch Patches:
	  inf-nan-patch.txt uploaded by capouch (license 6564) ........
	  Merged revisions 406788 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-28 19:19 +0000 [r406778]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_stasis_device_state.c: ARI: Make double subscribe
	  respond with success Currently, attempting to subscribe an
	  application to a device state that it has already subscribed to
	  will generate a 500 error response. This will now be treated as a
	  subscription refresh even though ARI subscriptions don't
	  currently support lifetimes and will respond with the normal
	  response for a successful subscription (200 OK). (closes issue
	  ASTERISK-23143) Reported by: Matt Jordan ........ Merged
	  revisions 406775 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-28 16:43 +0000 [r406724]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/rtp_engine.c, /: rtp_engine: improved handling of
	  get_rtp_info failure In ast_rtp_instance_make_compatible(), after
	  a failure of channel tech call get_rtp_info() to return
	  peer_instance, the null pointer would be passed to ao2_ref,
	  producing an error that looked like a refernce counting problem
	  but is not. This patch corrects that and adds helpful LOG_ERROR
	  messages to indicate which failure path occurred. (issue
	  AST-1276) Review: https://reviewboard.asterisk.org/r/3156/
	  ........ Merged revisions 406721 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406722 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406723 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-28 00:20 +0000 [r406710]  Richard Mudgett <rmudgett@digium.com>

	* /, tests/test_cel.c, tests/test_cdr.c: test_cdr.c, test_cel.c:
	  Correctly destroy created bridges. * Fixed the
	  test_cel_attended_transfer_bridges_link unit test to also account
	  for the local channel link being destroyed now that the bridges
	  are actually destroyed. * Made CDR unit test use its own version
	  of do_sleep() from the CEL unit tests. ........ Merged revisions
	  406707 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-27 22:54 +0000 [r406647-406696]  Kevin Harwell <kharwell@digium.com>

	* CHANGES: manager: ExtensionStatus event status human readable
	  Added a note in the changes file about the new 'StatusText' field
	  that was added to the 'ExtensionStatus' event. (issue
	  ASTERISK-23154) Reported by: Jonathan Rose

	* main/manager.c: manager: ExtensionStatus event status human
	  readable When an 'ExtensionStatus' event was raised it included
	  the status as a numerical value, but did not include a text
	  description of the status. Added a 'StatusText' field to the
	  event which is a string representation of the extension status.
	  Also added this to the 'Extension State' command response.
	  (closes issue ASTERISK-23154) Reported by: Jonathan Rose

2014-01-27 20:38 +0000 [r406646]  Russell Bryant <russell@russellbryant.com>

	* main/config.c, /: Allow nested #includes in extconfig.conf
	  extconfig.conf was hard-coded to not allow nested includes for
	  some reason. The code has been this way since a patch was merged
	  for ASTERISK-3333 (revision 4889), which was a significant update
	  to this code ("Merge config updates"). I can't figure out any
	  good reason why this should be limited. This patch just removes
	  the limit and uses the default nesting depth limit. Closes issue
	  ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/
	  ........ Merged revisions 406643 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406644 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406645 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-27 08:17 +0000 [r406618]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/manager.c, UPGRADE.txt, configs/manager.conf.sample:
	  manager: The eventfilter= option now takes an extended regex. In
	  pre-trunk versions (...12) it accepts a basic regex, which is
	  confusing because all other regexes in asterisk are of the
	  extended kind. Review: https://reviewboard.asterisk.org/r/3147/

2014-01-27 01:25 +0000 [r406595]  Russell Bryant <russell@russellbryant.com>

	* main/file.c, include/asterisk/channel.h, main/channel.c, /:
	  Protect ast_filestream object when on a channel The
	  ast_filestream object gets tacked on to a channel via
	  chan->timingdata. It's a reference counted object, but the
	  reference count isn't used when putting it on a channel. It's
	  theoretically possible for another thread to interfere with the
	  channel while it's unlocked and cause the filestream to get
	  destroyed. Use the astobj2 reference count to make sure that as
	  long as this code path is holding on the ast_filestream and
	  passing it into the file.c playback code, that it knows it's
	  valid. Bug reported by Leif Madsen. Review:
	  https://reviewboard.asterisk.org/r/3135/ ........ Merged
	  revisions 406566 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406567 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406574 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-26 23:04 +0000 [r406517]  Richard Mudgett <rmudgett@digium.com>

	* /, main/tcptls.c: tcptls.c: Add missing cleanup on off nominal
	  path. ........ Merged revisions 406514 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406515 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406516 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-26 14:19 +0000 [r406503]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/scripts/live_ast: live_ast: run wrapped programs with
	  exec live_ast can be used as a wrapper script to run asterisk,
	  gdb or valgrind. In those cases it runs them and returns the
	  result. It is more useful to use 'exec' to avoid having another
	  odd process in the chain. Review:
	  https://reviewboard.asterisk.org/r/3110/

2014-01-26 02:11 +0000 [r406490]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_session.c, /: res_pjsip_session: Be less strict
	  with core requested outgoing capabilities. The core may
	  (depending on circumstances) request a single codec on outgoing
	  calls. Many channel drivers ignore or treat this as a suggestion
	  while still including configured codecs. The res_pjsip_session
	  logic treated this as an explicit request, leaving out other
	  configured codecs. This change makes res_pjsip_session behave
	  like other channel driver and simply adds the requested codec to
	  the list. (closes issue ASTERISK-23082) Reported by: xrobau
	  Review: https://reviewboard.asterisk.org/r/3140/ ........ Merged
	  revisions 406489 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-24 23:33 +0000 [r406466]  Richard Mudgett <rmudgett@digium.com>

	* /, main/cel.c: CEL: Protect data structures during reload and
	  shutdown. The CEL data structures need to be protected during a
	  configuration reload and shutdown. Asterisk crashed during a
	  shutdown because CEL events were still in flight and the CEL data
	  structures were already destroyed. * Protected the cel_backends,
	  cel_dialstatus_store, and cel_linkedids ao2 containers with a
	  global ao2 object wrapper. * Added NULL checks before use of the
	  cel_backends, cel_dialstatus_store, and cel_linkedids ao2
	  containers in case the CEL module is already shutdown. * Fixed
	  overloading of the cel_linkedids held objects reference count.
	  During shutdown any held objects would be leaked. * Fixed memory
	  leak of cel_linkedids held objects if the LINKEDID_END is not
	  being tracked. The objects in the cel_linkedids container were
	  not removed if the LINKEDID_END event is not used. * Added access
	  protection to the cel_backends container during the CLI "cel show
	  status" command. * Made cel_backends, cel_dialstatus_store, and
	  cel_linkedids use the standard ao2 callback templates for the
	  hash and cmp functions. * Eliminated unnecessary uses of
	  RAII_VAR(). * Made ast_cel_engine_init() cleanup alocated
	  resources on failure. (closes issue AST-1253) Reported by:
	  Guenther Kelleter Review:
	  https://reviewboard.asterisk.org/r/3128/ ........ Merged
	  revisions 406417 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406418 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406465 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-24 22:34 +0000 [r406416]  Jonathan Rose <jrose@digium.com>

	* main/utils.c, CHANGES: Thread Debugging: Add LWP to core show
	  locks output This patch adds the LWP to core show locks output if
	  it is available. Review: https://reviewboard.asterisk.org/r/3142/

2014-01-24 22:18 +0000 [r406407]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, /: manager: Register atexit shutdown routine only
	  once. * Made register atexit shutdown routine only once in
	  __init_manager(). * Fixed some initial load failure conditions in
	  __init_manager(). * Made reset options to defaults on reload when
	  the reload will actually happen. * Removed unnecessary container
	  traversals of the white/black filters during manager_free_user().
	  * ast_free() does not need a NULL check before calling. ........
	  Merged revisions 406359 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406400 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406401 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-24 21:46 +0000 [r406399]  Jonathan Rose <jrose@digium.com>

	* res/res_config_pgsql.c, /: res_config_pgsql: Fix a memory leak
	  and use RAII_VAR for cleanup when practical Review:
	  https://reviewboard.asterisk.org/r/3141/ ........ Merged
	  revisions 406360 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406361 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406389 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-24 18:13 +0000 [r406343]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, /: manager: Protect data structures during
	  shutdown. Occasionally, the manager module would get an
	  "INTERNAL_OBJ: bad magic number" error on a "core restart
	  gracefully" command if an AMI connection is established. * Added
	  ao2_global_obj protection to the sessions global container. *
	  Fixed the order of unreferencing a session object in
	  session_destroy(). * Removed unnecessary container traversals of
	  the white/black filters during session_destructor(). (closes
	  issue AST-1242) Reported by: Guenther Kelleter Review:
	  https://reviewboard.asterisk.org/r/3144/ ........ Merged
	  revisions 406341 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406342 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-23 23:43 +0000 [r406328]  Mark Michelson <mmichelson@digium.com>

	* /: Today is not my day for writing code that compiles. ........
	  Merged revisions 406327 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-23 22:56 +0000 [r406312]  Michael L. Young <elgueromexicano@gmail.com>

	* /, addons/res_config_mysql.c: res_config_mysql: Fix Setting The
	  Column Name Incorrectly When support for a realtime sorcery
	  module was added in revision 386731, the wrong property was
	  accidentally used for setting the column name to be updated in
	  the database table. This patch fixes the typo. (closes issue
	  ASTERISK-23177) Reported by: Denis Tested by: Denis Patches:
	  asterisk-23177-use-field-name.diff by Michael L. Young (license
	  5026) ........ Merged revisions 406311 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-23 21:18 +0000 [r406298]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pidf.c, /: Multiple revisions 406294-406295
	  ........ r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu,
	  23 Jan 2014) | 11 lines Fix presence body errors found during
	  testing: * PIDF bodies were reporting an "open" state in many
	  cases where it should have been reporting "closed" * XPIDF bodies
	  had XML nodes placed incorrectly within the hierarchy. * SIP URIs
	  in XPIDF bodies did not go through XML sanitization * XML
	  sanitization had some errors: * Right angle bracket was being
	  replaced with "&rt;" instead of "&gt;" * Double quote,
	  apostrophe, and ampersand were not being escaped. ........
	  r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan
	  2014) | 11 lines Fix presence body errors found during testing: *
	  PIDF bodies were reporting an "open" state in many cases where it
	  should have been reporting "closed" * XPIDF bodies had XML nodes
	  placed incorrectly within the hierarchy. * SIP URIs in XPIDF
	  bodies did not go through XML sanitization * XML sanitization had
	  some errors: * Right angle bracket was being replaced with "&rt;"
	  instead of "&gt;" * Double quote, apostrophe, and ampersand were
	  not being escaped. ........ Merged revisions 406294-406295 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-22 22:24 +0000 [r406269]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/pbx.c, /, utils/extconf.c: pbx.c: Pre-initialize timezone to
	  avoid crash on destroy In ast_build_timing, initialize the
	  timezone value to NULL in order to avoid deferencing an
	  uninitialized value later when calling ast_destroy_timing. The
	  timezone value could be uninitialized if ast_build_timing were to
	  fail due to a zero length time string. (closes issue
	  ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review:
	  https://reviewboard.asterisk.org/r/3134/ Patches:
	  ast_build_timing-initialize-timezone.patch uploaded by
	  coreyfarrell (license 5909) ........ Merged revisions 406241 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406245 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406264 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-22 19:36 +0000 [r406153-406224]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_confbridge.c: ConfBridge: Fix channel parameter
	  documentation Confbridge AMI and CLI commands for mute, unmute,
	  and setting the single video source can accept channel prefixes
	  in lieu of a full channel name, but documentation states only
	  that it is required and is a channel name. This corrects the
	  documentation. (closes issue PQ-1397) Reported by: Steve Pitts
	  ........ Merged revisions 406217 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406223 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: chan_sip: Decline image streams on
	  unsupported transports This change allows chan_sip to decline
	  individual image streams over unsupported transports in the SDP
	  of the 200 response. Previously, an image stream offer with
	  RTP/AVP as the transport would cause chan_sip to respond with a
	  488. (closes issue ASTERISK-22988) Reported by: adomjan Original
	  patch by: adomjan ........ Merged revisions 406170 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406171 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406172 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis_playback.c, /: res_stasis_playback: Correct error
	  argument order Several of the playback error messages for invalid
	  media input in res_stasis_playback.c had the media name and
	  channel name reversed. They now correctly identify the channel
	  name and media name. Reported by: skrusty ........ Merged
	  revisions 406152 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-21 21:48 +0000 [r406134]  Rusty Newton <rnewton@digium.com>

	* /, res/res_pjsip.c: res_pjsip: Documentation improvement for
	  Endpoint and AOR mailbox options. Making the help text for both
	  more explicit regarding the format of mailbox identifiers. i.e.
	  clarifying the format for app_voicemail mailboxes vs mailboxes
	  from external MWI sources through modules such as
	  res_external_mwi. ........ Merged revisions 406133 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-21 21:08 +0000 [r406082]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/manager.c, /, configs/manager.conf.sample: manager: Clarify
	  eventfilter documentation. Textual changes only. Review:
	  https://reviewboard.asterisk.org/r/3133/ ........ Merged
	  revisions 406079 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406080 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406081 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-21 20:28 +0000 [r406006-406078]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_mgcp.c, /: chan_mgcp: Enforce locking for oseq This
	  restricts direct usage of global oseq so that all accesses are
	  locked and threads are not racing to get oseq values that they
	  did not claim. This also fixes a build error in res_pktccops
	  under dev mode. (closes issue ASTERISK-23100) Reported by:
	  adomjan Patch by: adomjan ........ Merged revisions 406037 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 406038 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 406049 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_outbound_registration.c, res/res_pjsip.c: PJSIP:
	  Handle headers in a list appropriately The PJSIP header parsing
	  function (pjsip_parse_hdr) can generate more than one header
	  instance from a single header field. These header instances exist
	  as a list attached to the returned header and must be handled
	  appropriately when they are added to a message or else only the
	  first header instance will be used. This changes the linked list
	  functions used in outbound proxy code to merge the lists
	  properly. ........ Merged revisions 406020 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/ari/resource_sounds.h, res/ari/resource_bridges.h,
	  res/ari/resource_device_states.h, res/ari/resource_mailboxes.h,
	  res/ari/resource_asterisk.h, rest-api/api-docs/channels.json,
	  res/ari/resource_applications.h, res/ari/resource_channels.c,
	  res/res_ari_playbacks.c, res/res_ari_sounds.c,
	  rest-api-templates/asterisk_processor.py,
	  res/ari/resource_channels.h, res/res_ari_bridges.c, /,
	  res/res_ari_device_states.c,
	  rest-api-templates/ari_resource.h.mustache,
	  res/res_ari_mailboxes.c, res/res_ari_asterisk.c,
	  res/res_ari_applications.c,
	  rest-api-templates/res_ari_resource.c.mustache,
	  rest-api-templates/body_parsing.mustache (added),
	  res/res_ari_channels.c, res/ari/resource_playbacks.h,
	  rest-api-templates/param_parsing.mustache: ARI: Support channel
	  variables in originate This adds back in support for specifying
	  channel variables during an originate without compromising the
	  ability to specify query parameters in the JSON body. This was
	  accomplished by generating the body-parsing code in a separate
	  function instead of being integrated with the URI query parameter
	  parsing code such that it could be called by paths with body
	  parameters. This is transparent to the user of the API and
	  prevents manual duplication of code or data structures. (closes
	  issue ASTERISK-23051) Review:
	  https://reviewboard.asterisk.org/r/3122/ Reported by: Matt Jordan
	  ........ Merged revisions 406003 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-20 23:25 +0000 [r405985]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Skinny: fix up handling of fragmented
	  packets. Bad offset in reading second or more fragment of skinny
	  packets. Fixed to offset by char (single byte) rather than size
	  of req. ........ Merged revisions 405982 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-20 22:23 +0000 [r405947]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, /: chan_dahdi/PRI: Suppress CONNECTED_LINE
	  updates when nothing in the udpate is valid. * Also simplified
	  some subddress handling code. (closes issue ASTERISK-23008)
	  Reported by: Michael Cargile ........ Merged revisions 405926
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 405927 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405928 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-20 21:56 +0000 [r405925]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Skinny: fix up session logging.
	  Logging from the skinny session loop was providing some incorrect
	  reasons for exiting the loop. Cleaned up messages and handling so
	  correct reason displayed. ........ Merged revisions 405924 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-20 18:18 +0000 [r405910]  Jonathan Rose <jrose@digium.com>

	* channels/chan_pjsip.c, /: chan_pjsip: Provide a means for
	  tracking device state when holding/unholding Previously PJSIP did
	  not track hold/unhold and it would always simply be 'inuse'. This
	  patch fixes that. review:
	  https://reviewboard.asterisk.org/r/3129/ ........ Merged
	  revisions 405908 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-19 00:01 +0000 [r405894]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Skinny: fix reversed device reset from
	  CLI. Existing code would do a full device restart when "skinny
	  reset device" was entered at the CLI and do a reset when "skinny
	  reset device restart" entered. ........ Merged revisions 405893
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-17 22:09 +0000 [r405878]  Sean Bright <sean@malleable.com>

	* /, channels/chan_sip.c: Make sure the maxptime attribute is added
	  to the correct offers. ........ Merged revisions 405877 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-17 21:33 +0000 [r405862-405876]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/format_pref.c, main/sorcery.c, main/frame.c, /,
	  include/asterisk/format_pref.h, res/res_pjsip_sdp_rtp.c: pjsip:
	  fix support for allow=all This change adds improvements to
	  support for allow=all in pjsip.conf so that it functions as
	  intended. Previously, the allow/disallow socery configuration
	  would set & clear codecs from the media.codecs and media.prefs
	  list, but if all was specified the prefs list was not updated.
	  Then a call would fail when create_outgoing_sdp_stream() created
	  an SDP with no audio codecs. A new function
	  ast_codec_pref_append_all() is provided to add all codecs to the
	  prefs list - only those not already on the list. This enables the
	  configuration to specify a codec preference, but still add all
	  codecs, and even then remove some codecs, as shown in this
	  example: allow = ulaw, alaw, all, !g729, !g723 Also, the display
	  order of allow in cli output is updated to match the
	  configuration by using prefs instead of caps when generating a
	  human readable string. Finally, a change to
	  create_outgoing_sdp_stream() skips a codec when it does not have
	  a payload code instead of the call failing. (closes issue
	  ASTERISK-23018) Reported by: xrobau Review:
	  https://reviewboard.asterisk.org/r/3131/ ........ Merged
	  revisions 405875 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/http.c: http: supported chunked Transfer-Encoding This
	  change implements support for HTTP Transfer-Encoding chunked in
	  both JSON and Form (post vars) body content. A new function
	  ast_http_get_contents() handles both regular and chunked mode
	  body, returning after the entire body is received. (closes issue
	  ASTERISK-23068) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3125/ ........ Merged
	  revisions 405861 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-17 18:55 +0000 [r405778-405844]  Rusty Newton <rnewton@digium.com>

	* res/res_pjsip.c, /: Fixing some XML syntax issues with my
	  previous commit at r405777 for ASTERISK-23071 ........ Merged
	  revisions 405843 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c, doc/asterisk.8, main/features.c,
	  configs/sip.conf.sample, apps/app_queue.c, apps/app_transfer.c,
	  channels/chan_iax2.c: Documentation: doc fixes across various
	  parts of the code for ASTERISK issues 23061,23028,23046,23027
	  Fixes typos of "transfered" instead of "transferred" in various
	  code. Fixes incorrect gosub param help text for app_queue. Fixes
	  Asterisk man pages containing unquoted minus signs. Adds note
	  about the "textsupport" option in sip.conf.sample. (issue
	  ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046)
	  (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes
	  issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue
	  ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis
	  Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine
	  (license 6561) hyphen.patch uploaded by Jeremy Laine (license
	  6561) sip.conf.sample.patch uploaded by Eugene (license 6360)
	  ........ Merged revisions 405791 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 405792 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405829 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip.c, /: res_pjsip: enhance documentation for
	  mailboxes options, for both endpoints and aors Made documentation
	  more explicit as to the use of the both options. (issue
	  ASTERISK-23071) (closes issue ASTERISK-23071) Reported by: Matt
	  Jordan ........ Merged revisions 405777 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-17 14:17 +0000 [r405766]  Walter Doekes <walter+asterisk@wjd.nu>

	* res/res_musiconhold.c, CHANGES: Enable wide band audio in
	  musiconhold streams. Review:
	  https://reviewboard.asterisk.org/r/3112/

2014-01-16 20:06 +0000 [r405747-405749]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip/pjsip_options.c, /: res_pjsip: AOR option
	  qualify_frequency not respected on startup If an endpoint had
	  previously dynamically registered a contact and the contact
	  information was successfully stored in astdb then upon restart
	  the qualify notifications would not be sent out if the
	  qualify_frequency was set. This was due to the fact that only
	  permanent contacts were being checked and scheduled for qualifies
	  on startup. Modified the code to check and schedule all
	  registered contacts at startup. (closes issue ASTERISK-23062)
	  Reported by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/3124/ ........ Merged
	  revisions 405748 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/manager.c, /: manager: Originate doesn't abort on failed
	  format_cap allocation action_originate responds to the remote
	  system with an error when cap==NULL, but doesn't return (abort
	  the originate). Patched to return. (closes issue ASTERISK-23034)
	  Reported by: Corey Farrell Patches: ASTERISK-23034.patch uploaded
	  by coreyfarrell (license 5909) ........ Merged revisions 405745
	  from http://svn.asterisk.org/svn/asterisk/branches/11 ........
	  Merged revisions 405746 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-16 19:33 +0000 [r405744]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_pjsip.c: PJSIP: Fix outbound OPTIONS support When path
	  support was added and contacts were made available during request
	  creation and transmission, the code path used by outbound qualify
	  support was not modified correctly and was causing request
	  creation to fail. This ensures that outbound request creation
	  with only a contact and no dialog, endpoint, or uri can succeed
	  which restores qualify support. Reported by: gtjoseph Reported
	  by: kharwell ........ Merged revisions 405743 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-16 19:13 +0000 [r405644-405695]  Kevin Harwell <kharwell@digium.com>

	* /, res/res_fax.c, configs/res_fax.conf.sample: res_fax:
	  check_modem_rate() returned incorrect rate for V.27 According to
	  the new standard for V.27 and V.32 they are able to transmit at a
	  bit rate of 4,800 or 9,600. The check_mode_rate function needed
	  to be updated to reflect this. Also, because of this change the
	  default 'minrate' value was updated to be 4800. (closes issue
	  ASTERISK-22790) Reported by: Paolo Compagnini Patches:
	  res_fax.txt uploaded by looserouting (license 6548) ........
	  Merged revisions 405656 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 405693 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405694 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_pjsip.c: chan_pjsip: initial device state on
	  endpoints is INVALID When endpoints get loaded their device state
	  gets set to 'INVALID' because the channel driver has not been
	  loaded yet. Fixed by updating the device state for every endpoint
	  upon load of the channel driver. (closes issue ASTERISK-23065)
	  Reported by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/3123/ ........ Merged
	  revisions 405643 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-15 16:51 +0000 [r405586-405589]  Jonathan Rose <jrose@digium.com>

	* CHANGES: Make 12 - 12.1 CHANGES log the same as in 12

	* CHANGES, /: Include CHANGES info for r405553 ........ Merged
	  revisions 405585 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-15 16:36 +0000 [r405584]  Joshua Colp <jcolp@digium.com>

	* /, cel/cel_manager.c: cel_manager: Don't crash if configuration
	  file is invalid. The cel_manager module did not properly handle
	  the case where the configuration file was invalid. The module
	  will now output a warning message and disable itself if this
	  occurs. Reported by: Bryan Walters ........ Merged revisions
	  405581 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 405582 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405583 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-15 13:16 +0000 [r405566]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c,
	  res/res_pjsip_path.c (added), res/res_pjsip_mwi.c,
	  res/res_pjsip/pjsip_distributor.c, res/res_pjsip_diversion.c,
	  channels/chan_pjsip.c, res/res_pjsip_registrar.c,
	  res/res_pjsip_refer.c, include/asterisk/res_pjsip.h,
	  include/asterisk/res_pjsip_session.h, res/res_pjsip_notify.c, /,
	  res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c,
	  res/res_pjsip_t38.c, res/res_pjsip.c,
	  res/res_pjsip/pjsip_options.c, res/res_pjsip_nat.c,
	  res/res_pjsip_session.c,
	  contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py
	  (added), res/res_pjsip_header_funcs.c: PJSIP: Add Path header
	  support This adds Path support to chan_pjsip in res_pjsip_path.c
	  with minimal additions in res_pjsip_registrar.c to store the path
	  and additions in res_pjsip_outbound_registration.c to enable
	  advertisement of path support to registrars and intervening
	  proxies. Path information is stored on contacts and is enabled
	  via Address of Record (AoRs) and Registration configuration
	  sections. While adding path support, it became necessary to be
	  able to add SIP supplements that handled messages outside of
	  sessions, so a framework for handling these types of hooks was
	  added in parallel to the already-existing session supplements and
	  several senders of out-of-dialog requests were refactored as a
	  result. (closes issue ASTERISK-21084) Review:
	  https://reviewboard.asterisk.org/r/3050/ ........ Merged
	  revisions 405565 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-14 23:44 +0000 [r405554]  Jonathan Rose <jrose@digium.com>

	* res/res_stasis_mailbox.exports.in (added),
	  res/ari/ari_model_validators.h, rest-api/api-docs/mailboxes.json
	  (added), include/asterisk/stasis_app_mailbox.h (added),
	  res/ari/resource_mailboxes.c (added), /, res/ari.make,
	  res/res_ari_mailboxes.c (added), res/ari/resource_mailboxes.h
	  (added), res/res_stasis_mailbox.c (added),
	  rest-api/resources.json, res/ari/ari_model_validators.c: ARI: Add
	  mailboxes resource for controlling and polling external MWI Adds
	  the following AMI commands: PUT mailboxes/mailboxName modifies
	  mailbox state and implicitly creates new mailboxes GET
	  mailboxes/mailboxName retrieves a JSON representation of a single
	  mailbox if it exists GET mailboxes retrieves a JSON array of all
	  mailboxes DELETE mailbox/mailboxName deletes a mailbox Note that
	  res_mwi_external must be loaded for these functions to actually
	  do anything. Review: https://reviewboard.asterisk.org/r/3117/
	  ........ Merged revisions 405553 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-14 21:46 +0000 [r405542]  Richard Mudgett <rmudgett@digium.com>

	* main/strings.c, /: string container: Remove unnecessary RAII_VAR
	  usage and string object lock. ........ Merged revisions 405541
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-14 18:15 +0000 [r405437]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, channels/chan_sip.c: chan_sip: fix Local From tag on outbound
	  register regression In ASTERISK-12117, an improvement to insure
	  consistant local from tags on outbound registrations resulted in
	  an undesirable behavior - caused by leftover unexpired sip_pvt
	  dialogs (with the previous cseq number), resulting in many
	  uncessary REGISTER requests. Instead of significant rework of
	  transmit_register(), this change deletes the dialogs after a 200
	  OK response indiciating a successful registration, keeping the
	  old dialogs from interfering with normal operation. (closes issue
	  ASTERISK-22946) Reported by: Stephan Eisvogel Review:
	  https://reviewboard.asterisk.org/r/3109/ ........ Merged
	  revisions 405433 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 405434 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405435 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-14 18:14 +0000 [r405436]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_verbose.c, main/asterisk.c, configs/logger.conf.sample,
	  main/cli.c, include/asterisk/logger.h, main/pbx.c,
	  main/manager.c, /, funcs/func_timeout.c, apps/app_dumpchan.c,
	  main/logger.c, UPGRADE.txt: verbosity: Fix performance of console
	  verbose messages. The per console verbose level feature as
	  previously implemented caused a large performance penalty. The
	  fix required some minor incompatibilities if the new rasterisk is
	  used to connect to an earlier version. If the new rasterisk
	  connects to an older Asterisk version then the root console
	  verbose level is always affected by the "core set verbose"
	  command of the remote console even though it may appear to only
	  affect the current console. If an older version of rasterisk
	  connects to the new version then the "core set verbose" command
	  will have no effect. * Fixed the verbose performance by not
	  generating a verbose message if nothing is going to use it and
	  then filtered any generated verbose messages before actually
	  sending them to the remote consoles. * Split the "core set debug"
	  and "core set verbose" CLI commands to remove the per module
	  verbose support that cannot work with the per console verbose
	  level. * Added a silent option to the "core set verbose" command.
	  * Fixed "core set debug off" tab completion. * Made "core show
	  settings" list the current console verbosity in addition to the
	  root console verbosity. * Changed the default verbose level of
	  the 'verbose' setting in the logger.conf [logfiles] section. The
	  default is now to once again follow the current root console
	  level. As a result, using the AMI Command action with "core set
	  verbose" could again set the root console verbose level and
	  affect the verbose level logged. (closes issue AST-1252) Reported
	  by: Guenther Kelleter Review:
	  https://reviewboard.asterisk.org/r/3114/ ........ Merged
	  revisions 405431 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405432 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-14 16:43 +0000 [r405420]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip/pjsip_distributor.c: Fix erroneous behavior when
	  sending auth rejection to artificial endpoint. We were not
	  including an authentication challenge when sending a 401 response
	  to unmatched endpoints. This was due to the conversion to use a
	  vector for authentication section names on an endpoint. The
	  vector for artificial endpoints was empty, resulting in the
	  challenge being sent back containing no challenges. This is
	  worked around by placing a bogus value in the artificial
	  endpoint's auth vector. This value is never looked up by
	  anything, since they instead will directly call
	  ast_sip_get_artificial_auth().

2014-01-14 03:27 +0000 [r405369]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Skinny: do not add call to missed
	  calls list if answered elsewhere. Patch updates skinny devices
	  with a SKINNY_CONNECTED callstate if an inbound ringing or
	  callwaiting call is answered elsewhere. ........ Merged revisions
	  405367 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-13 13:34 +0000 [r405339]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_pjsip/pjsip_cli.c: res_pjsip: Fix CLI tab completion
	  issues This fixes several issues with the new res_pjsip CLI tab
	  completion such as output of headers during tab completion and
	  being able to tab-complete more items than the code actually
	  handled (further items would simply be ignored). (closes issue
	  ASTERISK-23081) Review: https://reviewboard.asterisk.org/r/3115/
	  Reported by: xrobau ........ Merged revisions 405338 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-12 22:24 +0000 [r405326]  Joshua Colp <jcolp@digium.com>

	* res/ari/resource_playbacks.c, res/ari/resource_channels.c,
	  include/asterisk/ari.h, res/ari/resource_bridges.c,
	  res/ari/resource_recordings.c, res/ari/resource_device_states.c,
	  res/res_ari.c, res/ari/resource_endpoints.c, /,
	  res/ari/resource_applications.c: res_ari: Fix various memory
	  leaks. This change fixes a few memory leaks that were found based
	  on a mailing list post. 1. Some JSON response messages were never
	  freed. This was caused by the documentation stating that message
	  references were stolen when in reality they were not. The code
	  now follows the documentation and usage has been updated. 2. HTTP
	  response headers were never freed. 3. The variable list for
	  wildcards paths was never freed. (closes issue ASTERISK-23128)
	  Reported by: Kenneth Watson (on list) Review:
	  https://reviewboard.asterisk.org/r/3119/ ........ Merged
	  revisions 405325 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-12 22:13 +0000 [r405313-405314]  Matthew Jordan <mjordan@digium.com>

	* apps/app_forkcdr.c, /, funcs/func_cdr.c, include/asterisk/cdr.h,
	  apps/app_cdr.c, main/cdr.c: CDRs: Synchronize dialplan
	  applications that manipulate CDRs with the engine In
	  https://reviewboard.asterisk.org/r/3057/, applications and
	  functions that manipulate CDRs were made to interact over Stasis.
	  This was done to synchronize manipulations of CDRs from the
	  dialplan with the updates the engine itself receives over the
	  message bus. This change rested on a faulty premise: that
	  messages published to the CDR topic or to a topic that forwards
	  to the CDR topic are synchronized with the messages handled by
	  the CDR topic subscription in the CDR engine. This is not the
	  case. There is no ordering guaranteed for two messages published
	  to the same topic; ordering is only guaranteed if a message is
	  published to the same subscriber. Stasis was modified in r405311
	  to allow a publisher to synchronize on the subscriber. This patch
	  uses that API to synchronize the CDR publishers with the CDR
	  engine message router, which maintains the overall topic
	  subscription. (closes issue ASTERISK-22884) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........
	  Merged revisions 405312 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis.c, main/stasis_message_router.c, /,
	  include/asterisk/stasis.h,
	  include/asterisk/stasis_message_router.h, tests/test_stasis.c:
	  stasis: Add methods to allow for synchronous publishing to
	  subscriber This patch adds an API call to Stasis that allows a
	  publisher to publish a stasis message that will not return until
	  a specific subscriber handles the message. Since a subscriber can
	  have their own forwarding topic which orders messages from many
	  topics, this allows a publisher who knows of that subscriber to
	  synchronize to that subscriber regardless of the forwarding
	  relationships between topics. This is of particular use for
	  dialplan applications that need to synchronize on a particular
	  subscriber's handling of a message. (issue ASTERISK-22884)
	  Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3099/ ........ Merged
	  revisions 405311 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-10 20:00 +0000 [r405299]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip/security_events.c: Print "<unknown>" for
	  artificial endpoint in PJSIP security events. Previously, this
	  printed a UUID, which was not very clear when dealing with an
	  artificial endpoint. Review:
	  https://reviewboard.asterisk.org/r/3113 ........ Merged revisions
	  405298 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-10 18:17 +0000 [r405284]  Richard Mudgett <rmudgett@digium.com>

	* /, main/logger.c: Logging callid: Fix some sizeof() references
	  per coding guidelines. ........ Merged revisions 405281 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405282 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-09 23:52 +0000 [r405270]  Jonathan Rose <jrose@digium.com>

	* res/res_pjsip_session.c: PJSIP: Add unhold on reinvite without
	  SDP behavior Review: https://reviewboard.asterisk.org/r/3106/

2014-01-09 23:50 +0000 [r405269]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_dahdi.c, /: Fix chan_dahdi copile issue in
	  dev-mode. Error "unused variable i in dahdi_create_channel_range"
	  when compiling in dev-mode. Small restructure to
	  dahdi_create_channel_range to move the for(x) loop and int i,x to
	  a block within the IFDEF. ........ Merged revisions 405268 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-09 23:39 +0000 [r405267]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip.c, /, res/res_pjsip_messaging.c:
	  res_pjsip_messaging: potential for field values in from/to
	  headers to be missing Added in ability to specify display name
	  format ("name" <sip:name@ipaddr:port>) for a given URI and made
	  sure it was fully propagated to the outgoing message. Also made
	  it so outoing messages in res_pjsip always send as "sip:".
	  (closes issue ASTERISK-22924) Reported by: Anthony Messina
	  Review: https://reviewboard.asterisk.org/r/3094/ ........ Merged
	  revisions 405266 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-09 20:34 +0000 [r405254]  Kinsey Moore <kmoore@digium.com>

	* main/astobj2.c, res/res_pjsip_session.c, /,
	  include/asterisk/astobj2.h: astobj2: Correct ao2_iterator opacity
	  violations This corrects the ao2_iterator opacity violations in
	  res_pjsip_session.c by adding a global function to get the number
	  of elements inside the container hidden behind the iterator.
	  (closes issue ASTERISK-23053) Review:
	  https://reviewboard.asterisk.org/r/3111/ Reported by: Richard
	  Mudgett ........ Merged revisions 405253 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-09 16:52 +0000 [r405236]  Kevin Harwell <kharwell@digium.com>

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fails to resume
	  WebRTC call from hold In ast_rtp_ice_start if the ice session
	  create check list failed, start check was never initiated and
	  ice_started was never set to true. Upon re-entering the function
	  (for instance, [un]hold) it would try to create the check list
	  again with duplicate remote candidates. Fixed so that if the
	  create check list fails the necessary data structures are
	  properly re-initialized for any subsequent retries. Note, it was
	  decided to not stop ice support (by calling ast_rtp_ice_stop) on
	  a check list failure because it possible things might still work.
	  However, a debug message was added to help with any future
	  troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis
	  Valentinavičius Patches: works_on_my_machine.patch uploaded by
	  xytis (license 6558) ........ Merged revisions 405234 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405235 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-09 15:50 +0000 [r405217]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_confbridge.c,
	  apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
	  crash caused when waitmarked/marked users leave together When
	  waitmarked users join a ConfBridge, the conference state is
	  transitioned from EMPTY -> INACTIVE. In this state, the users are
	  maintined in a waiting users list. When a marked user joins, the
	  ConfBridge conference transitions from INACTIVE -> MULTI_MARKED,
	  and all users are put onto the active list of users. This process
	  works correctly. When the marked user leaves, if they are the
	  last marked user, the MULTI_MARKED state does the following: (1)
	  It plays back a message to the bridge stating that the leader has
	  left the conference. This requires an unlocking of the bridge.
	  (2) It moves waitmarked users back to the waiting list (3) It
	  transitions to the appropriate state: in this case, INACTIVE
	  However, because it plays the prompt back to the bridge before
	  moving the users and before finishing the state transition, this
	  creates a race condition: with the bridge unlocked, waitmarked
	  users who leave the conference (or are kicked from it) can cause
	  a state transition of the bridge to another state before the
	  conference is transitioned to the INACTIVE state. This causes the
	  state machine to get a bit wonky, often leading to a crash when
	  the MULTI_MARKED state attempts to conclude its processing. This
	  patch fixes this problem: (1) It prevents kicked users from being
	  kicked again. That's just a nicety. (2) More importantly, it
	  fixes the race condition by only playing the prompt once the
	  state has transitioned correctly to INACTIVE. If waitmarked users
	  sneak out during the prompt being played, no harm no foul.
	  Review: https://reviewboard.asterisk.org/r/3108/ Note that the
	  patch committed here is essentially the same as uploaded by Simon
	  Moxon on ASTERISK-22740, with the addition of the double kick
	  prevention. (closes issue AST-1258) Reported by: Steve Pitts
	  (closes issue ASTERISK-22740) Reported by: Simon Moxon patches:
	  ASTERISK-22740.diff uploaded by Simon Moxon (license 6546)
	  ........ Merged revisions 405215 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405216 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-09 14:15 +0000 [r405163]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, apps/app_dumpchan.c: "Minimun" typo. ........ Merged revisions
	  405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 405161 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405162 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-08 17:23 +0000 [r405144]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip/security_events.c: Use proper case for checking
	  if digest authentication is used. ........ Merged revisions
	  405131 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-08 16:34 +0000 [r405129-405130]  Kinsey Moore <kmoore@digium.com>

	* /, configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support
	  for Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is
	  available on newer operating systems. (closes issue
	  ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/
	  Reported by: George Joseph Patch by: George Joseph ........
	  Merged revisions 405090 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 405091 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405124 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: Add the missing part of r400140 When the
	  patch to add retry-on-forbidden-response was committed, part of
	  the patch for chan_sip was not committed which caused the feature
	  to be entirely nonfunctional. This corrects the code in question.
	  (closes issue ASTERISK-17138) Review:
	  https://reviewboard.asterisk.org/r/2874 ........ Merged revisions
	  405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 405081 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 405083 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-07 19:56 +0000 [r405020-405035]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_acl.c: res_pjsip_acl: Fix another case of
	  assuming a contact will always contain a URI. ........ Merged
	  revisions 405034 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_nat.c: res_pjsip_nat: Don't assume a Contact
	  header will always contain a URI. If the 'rewrite_contact' option
	  was enabled and a Contact header was received which contained a
	  '*' a crash would occur. This change makes the res_pjsip_nat
	  module ignore the Contact header if it contains only a '*'.
	  (closes issue ASTERISK-23101) Reported by: Matt Jordan ........
	  Merged revisions 405019 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-06 21:55 +0000 [r404953-405007]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_voicemail.c, /: app_voicemail: Explicitly set
	  defaultenabled=yes ........ Merged revisions 405006 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_mwi_external_ami.c (added): External MWI AMI support.
	  The external MWI AMI interface provides a thin wrapper around the
	  core external MWI resource. The resource adds the following AMI
	  actions: MWIGet, MWIDelete, and MWIUpdate. (closes issue AFS-46)
	  Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged
	  revisions 404954 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_mwi_external.c (added), configs/sorcery.conf.sample,
	  include/asterisk/res_mwi_external.h (added),
	  res/res_mwi_external.exports.in (added), apps/app_voicemail.c:
	  External MWI core support. * The core external MWI resource
	  provides for MWI message counts persistence using sorcery. With
	  sorcery, the user is able to configure which sorcery wizzard
	  backend to use if the default astdb is not desired. * The core
	  external MWI resoruce provides some debugging CLI commands
	  enabled by defining MWI_DEBUG_CLI. The debugging CLI commands
	  are: "mwi delete all", "mwi delete like <regex>", "mwi delete
	  mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi
	  show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new>
	  [<old>]]". (closes issue AFS-43) Review:
	  https://reviewboard.asterisk.org/r/3061/ ........ Merged
	  revisions 404952 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-05 16:01 +0000 [r404924-404936]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_outbound_registration.c:
	  res_pjsip_outbound_registration: Don't assume that a registration
	  client will always exist. ........ Merged revisions 404935 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_outbound_registration.c:
	  res_pjsip_outbound_registration: Create registration client in pj
	  thread. Depending on which threading was loading the outbound
	  registration it was possible for the registration client to be
	  allocated outside of a pj thread. This change moves the creation
	  inside the synchronous task where it is guaranteed it will occur
	  in a pj thread. Reported by: Rob Thomas ........ Merged revisions
	  404923 from http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-04 10:52 +0000 [r404912]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/asterisk.c, /: asterisk.c: suppress live_dangerously warning
	  on rasterisk Even since the fixes of AST-2013-007, Asterisk
	  prints the following warning on startup if the user decided to
	  live dangerously: Privilege escalation protection disabled! See
	  https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
	  message is intended for the logs and interactive startup. No need
	  for it to appear on a remote console. This commit removes it from
	  there. (closes issue ASTERISK-23084) Review:
	  https://reviewboard.asterisk.org/r/3101/ ........ Merged
	  revisions 404861 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404888 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404911 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-03 22:00 +0000 [r404860]  Kevin Harwell <kharwell@digium.com>

	* cel/cel_pgsql.c, /: cel_pgsql: module not correctly reloading
	  Upon reload the module unconditionally "unloaded" the module
	  (freeing memory and setting pointers to NULL) and then when
	  attempting a "load" if the config file had not changed then
	  nothing would be reinitialized. By moving the "unload" to occur
	  conditionally (reload only) after an attempted configuration
	  load, but before module "loading" alleviates the issue. The
	  module now loads/unloads/reloads correctly. (closes issue
	  ASTERISK-22871) Reported by: Matteo ........ Merged revisions
	  404857 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 404858 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404859 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-03 21:45 +0000 [r404844-404856]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_pjsip_logger.c: res_pjsip_logger: Add the
	  ASTERISK_FILE_VERSION macro Registering yourself with the
	  Asterisk core is the nice thing to do, even when you're a logging
	  module. ........ Merged revisions 404855 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_authenticator_digest.c, tests/test_utils.c:
	  res_pjsip_authenticator_digest: Fix md5 hash buffer An md5 hash
	  is 32 bytes long. The char buffer must be at least 33 bytes to
	  avoid clobbering of the stack. This patch also fixes a potential
	  clobbering in test_utils.c. Thanks to Andrew Nagy for reporting
	  and testing this out in #asterisk-dev Reported by: Andrew Nagy
	  Tested by: Andrew Nagy ........ Merged revisions 404843 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-03 20:02 +0000 [r404787-404832]  Kevin Harwell <kharwell@digium.com>

	* main/manager.c: manager: UserEvent including action on output AMI
	  action UserEvent event response would include the action header
	  in its keyvalue pairs list. Adjusted the start of the header loop
	  to skip over the action part. (closes issue ASTERISK-22899)
	  Reported by: outtolunc Patches:
	  svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license
	  5198)

	* channels/chan_dahdi.c, /: chan_dahdi: dahdi show channels slices
	  PRI channel dnid on output dahdi show channels output slices the
	  callerid (which is dnid copied over on PRI channels). If the
	  channel naming structures look like: 'DAHDI/i1/1408409XXXX-6'
	  then the output slices 1408409XXXX down to 1408409XXX. This patch
	  just opens it up to 15 chars so you can see the whole thing.
	  (closes issue ASTERISK-22918) Reported by: outtolunc Patches:
	  svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc
	  (license 5198) ........ Merged revisions 404784 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404785 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404786 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-03 18:33 +0000 [r404783]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_stasis.c, /: test_stasis.c: Fix ref leak in normal
	  execution path. ........ Merged revisions 404764 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-03 18:31 +0000 [r404782]  Kevin Harwell <kharwell@digium.com>

	* /, apps/app_meetme.c: app_meetme: compiler warning Fixed a
	  compiler warning (errors in 'dev-mode') given by gcc version
	  4.8.1. The one in app_meetme involved the
	  'sizeof-pointer-memaccess' (see:
	  http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so it
	  would no longer issue a warning and can compile again in
	  'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/
	  ........ Merged revisions 404742 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404773 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404781 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-03 17:27 +0000 [r404726-404738]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip/pjsip_configuration.c, /, res/res_pjsip/location.c:
	  res_pjsip: Ensure more URI validation happens in pj threads.
	  ........ Merged revisions 404737 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_outbound_registration.c:
	  res_pjsip_outbound_registration: Ensure URI validation happens in
	  a pjlib thread. This change moves outbound registration URI
	  validation into the task executed within a pjlib thread. Reported
	  by: Andrew Nagy ........ Merged revisions 404725 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-02 19:38 +0000 [r404677]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, funcs/func_strings.c: func_strings: use memmove to prevent
	  overlapping memory on strcpy When calling REPLACE() with an empty
	  replace-char argument, strcpy is used to overwrite the the
	  matching <find-char>. However as the src and dest arguments to
	  strcpy must not overlap, it causes other parts of the string to
	  be overwritten with adjacent characters and the result is
	  mangled. Patch replaces call to strcpy with memmove and adds a
	  test suite case for REPLACE. (closes issue ASTERISK-22910)
	  Reported by: Gareth Palmer Review:
	  https://reviewboard.asterisk.org/r/3083/ Patches:
	  func_strings.patch uploaded by Gareth Palmer (license 5169)
	  ........ Merged revisions 404674 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404675 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404676 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2014-01-02 19:08 +0000 [r404664]  Kevin Harwell <kharwell@digium.com>

	* channels/chan_pjsip.c, include/asterisk/res_pjsip.h, /,
	  configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c,
	  CHANGES, res/res_pjsip.c: res_pjsip: add 'set_var' support on
	  endpoints Added a new 'set_var' option for ast_sip_endpoint(s).
	  For each variable specified that variable gets set upon creation
	  of a pjsip channel involving the endpoint. (closes issue
	  ASTERISK-22868) Reported by: Joshua Colp Review:
	  https://reviewboard.asterisk.org/r/3095/ ........ Merged
	  revisions 404663 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-31 22:51 +0000 [r404620-404653]  Joshua Colp <jcolp@digium.com>

	* channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip:
	  Handle hanging up before calling. Channel creation in Asterisk is
	  broken up into two steps: requesting and calling. In some cases a
	  channel may be requested but never called. This happens in the
	  ChanIsAvail dialplan application for determining if something is
	  reachable or not. The PJSIP channel driver did not take this
	  situation into account and attempted to end a session that was
	  never called out on. The code now checks the session state to
	  determine if the session has been called out on and if not
	  terminates it instead of ending it. (closes issue ASTERISK-23074)
	  Reported by: Kilburn ........ Merged revisions 404652 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_endpoint_identifier_ip.c:
	  res_pjsip_endpoint_identifier_ip: Accept hostnames in the 'match'
	  field. Hostnames specified in the 'match' field will be resolved
	  and all addresses returned. Each address will be added to the
	  endpoint identifier for the matching process. Reported by: Rob
	  Thomas ........ Merged revisions 404613 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-31 21:39 +0000 [r404606]  Kevin Harwell <kharwell@digium.com>

	* cel/cel_pgsql.c, /: cel_pgsql: deadlock on unload and
	  core_event_dispatcher A deadlock can happen between a thread
	  unloading or reloading the cel_pgsql module and the
	  core_event_dispatcher taskprocessor thread. Description of what
	  is happening: Thread 1 (for example, a netconsole thread): a
	  "module reload cel_pgsql" is launched the thread enter the
	  "my_unload_module" function (cel_pgsql.c) the thread acquire the
	  write lock on psql_columns the thread enter the
	  "ast_event_unsubscribe" function (event.c) the thread try to
	  acquire the write lock on ast_event_subs[sub->type] Thread 2
	  (core_event_dispatcher taskprocessor thread): the taskprocessor
	  pop a CEL event the thread enter the "handle_event" function
	  (event.c) the thread acquire the read lock on
	  ast_event_subs[sub->type] the thread callback the "pgsql_log"
	  function (cel_pgsql.c), since it's a subscriber of CEL events the
	  thread try to acquire a read lock on psql_columns (closes issue
	  ASTERISK-22854) Reported by: Etienne Lessard Patches:
	  cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license
	  6394) ........ Merged revisions 404603 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404604 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404605 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-31 20:27 +0000 [r404593]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_outbound_registration.c, /:
	  res_pjsip_outbound_registration: Add validation for 'server_uri'
	  and 'client_uri'. When applying configuration for outbound
	  registrations the 'server_uri' and 'client_uri' fields were not
	  validated. The code will now confirm that they exist and that
	  they contain parseable SIP URIs. Reported by: Andrew Nagy
	  ........ Merged revisions 404592 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-30 23:25 +0000 [r404582]  Kevin Harwell <kharwell@digium.com>

	* main/channel.c, /: channels.c: core show channeltypes slicing
	  'core show channeltypes' type column is being sliced, resulting
	  in incomplete type names. (closes issue ASTERISK-22919) Reported
	  by: outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded
	  by outtolunc (license 5198) ........ Merged revisions 404579 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404581 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-24 17:12 +0000 [r404567-404569]  David M. Lee <dlee@digium.com>

	* UPGRADE-12.txt, /: Added note to UPGRADE.txt about the default
	  value of live_dangerously changing ........ Merged revisions
	  404568 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/http.c: http: Properly reject requests with
	  Transfer-Encoding set Asterisk does not support any of the
	  transfer encodings specified in HTTP/1.1, other than the default
	  "identity" encoding. According to RFC 2616: A server which
	  receives an entity-body with a transfer-coding it does not
	  understand SHOULD return 501 (Unimplemented), and close the
	  connection. A server MUST NOT send transfer-codings to an
	  HTTP/1.0 client. This patch adds the 501 Unimplemented response,
	  instead of the hard work of actually implementing other
	  recordings. This behavior is especially problematic for Node.js
	  clients, which use chunked encoding by default. (closes issue
	  ASTERISK-22486) Review: https://reviewboard.asterisk.org/r/3092/
	  ........ Merged revisions 404565 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-24 02:20 +0000 [r404554]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Ensure dialog
	  manipulation happens on proper thread. When destroying a
	  subscription we remove the serializer from its dialog and
	  decrease its reference count. Depending on which thread dropped
	  the subscription reference count to 0 it was possible for this to
	  occur in a thread where it is not possible. (closes issue
	  ASTERISK-22952) Reported by: Matt Jordan ........ Merged
	  revisions 404553 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-23 16:38 +0000 [r404542]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
	  UPGRADE-12.txt: chan_dahdi: enable ignore_failed_channels by
	  default If ignore_failed_channels is set to "true" for a channel,
	  the channel will continue to be configured even if configuring it
	  has failed. This allows Asterisk to start before all the DAHDI
	  initialization is done and thus not force the starting order
	  dahdi -> asterisk. Review:
	  https://reviewboard.asterisk.org/r/3063/

2013-12-21 03:35 +0000 [r404532]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_pjsip/pjsip_cli.c: res_pjsip/pjsip_cli: fix
	  compilation error caused by passing ast_free When wanting to pass
	  *free as a function pointer, ast_free_ptr has to be used instead
	  of ast_free. This allows it to be compiled with MALLOC_DEBUG
	  enabled. ........ Merged revisions 404531 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 22:04 +0000 [r404511-404512]  David M. Lee <dlee@digium.com>

	* rest-api/api-docs/channels.json, res/ari/resource_channels.c,
	  res/res_ari_channels.c, res/ari/resource_channels.h, /,
	  rest-api/api-docs/applications.json: ari: Remove support for
	  specifying channel vars during origination. When we added support
	  for specifying channel variables for an origination, we didn't
	  consider how that would interact with another feature, namely
	  specifying request parameters in a JSON request body. The method
	  of specifying channel variables (as a flat JSON object passed in
	  the JSON body) interferes with parsing parameters out of the
	  request body. Unfortunately, fixing this would be a backward
	  incompatible change. In the interest of keeping the API sane and
	  keeping our release schedule, we're dropping the feature for
	  specifying channel variables in the origination request. We will
	  bring the feature back soon, as a backward compatible addition to
	  the API. (closes issue ASTERISK-23051) Review:
	  https://reviewboard.asterisk.org/r/3088 ........ Merged revisions
	  404509 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /: Remove automerge properties ........ Merged revisions 404488
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 21:32 +0000 [r404507]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/config.h, main/config.c, main/channel.c,
	  res/res_pjsip/location.c, include/asterisk/res_pjsip_cli.h
	  (added), res/res_pjsip/pjsip_cli.c (added),
	  include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip/include/res_pjsip_private.h,
	  res/res_pjsip_registrar.c, main/sorcery.c,
	  include/asterisk/res_pjsip.h, CREDITS,
	  res/res_pjsip/config_auth.c, /,
	  res/res_pjsip_endpoint_identifier_ip.c: res_pjsip: Add PJSIP CLI
	  commands Implements the following cli commands: pjsip list aors
	  pjsip list auths pjsip list channels pjsip list contacts pjsip
	  list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show
	  channels pjsip show endpoint(s) Also... Minor modifications made
	  to the AMI command implementations to facilitate reuse. New
	  function ast_variable_list_sort added to config.c and config.h to
	  implement variable list sorting. (issue ASTERISK-22610) patches:
	  pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
	  ........ Merged revisions 404480 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 21:18 +0000 [r404461]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, main/say.c: say.c: correct time for polish In
	  ast_say_date_with_format_pl(), change ast_say_number() to use
	  tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported
	  by: Robert Mordec Review:
	  https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch
	  uploaded by veilen (license 6555) ........ Merged revisions
	  404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 404457 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404458 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 20:28 +0000 [r404452]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip_refer.c: Fix issue where PJSIP blind transferer
	  dialog may not complete as planned. When transferring to a
	  dialplan extension that will not place any outbound calls, the
	  only control frames that the PJSIP REFER framehook will receive
	  are inconsequential (such as unhold or srcchange). As such, we
	  shouldn't allow for the reception of those types of frames
	  prevent us from signaling to the transferring party that the
	  transfer has completed successfully once voice frames are read.
	  Thanks to Jonathan Rose for pointing this out. ........ Merged
	  revisions 404439 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 20:05 +0000 [r404438]  Matthew Jordan <mjordan@digium.com>

	* /, res/ari/resource_applications.h,
	  res/res_stasis_device_state.c: res_stasis_device_state: Set
	  resource type for subscriptions to deviceState The documentation
	  for ARI already specifies that the device state resource when
	  used for subscribing for events is "deviceState", not
	  "device_state". The code, however, used "device_state"; although
	  this was inconsistent as well in doxygen comments in
	  resource_applications. Because the actual resource being
	  subscribed to is /deviceStates/{device}/, it makes sense for the
	  resource type specifier to be deviceState. Note that the key
	  value in the events is still "device_state". ........ Merged
	  revisions 404437 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 20:00 +0000 [r404436]  Richard Mudgett <rmudgett@digium.com>

	* res/ari/resource_channels.c, tests/test_scoped_lock.c,
	  tests/test_stasis.c, res/parking/parking_manager.c,
	  res/ari/resource_bridges.c, res/ari/resource_endpoints.c, /,
	  res/res_pjsip/location.c, tests/test_cel.c: ao2_iterator:
	  Mini-audit of the ao2_iterator loops in the new code files. *
	  Fixed several places where ao2_iterator_destroy() was not called.
	  * Fixed several iterator loop object variable reference problems.
	  * Fixed res_parking AMI actions returning non-zero. Only the AMI
	  logoff action can return non-zero. Review:
	  https://reviewboard.asterisk.org/r/3087/ ........ Merged
	  revisions 404434 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 19:25 +0000 [r404433]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/manager.h, /: manager: bump version to 2.0.0 AMI
	  has received substantial updates over the past year. Not only has
	  the syntax been vastly improved and made consistent (which
	  entails many event changes), but the underlying things that those
	  events convey have changed substantially as well. After some
	  conversation in #asterisk-dev, it was agreed that this is a good
	  time to jump to 2. At the same time, since ARI will most likely
	  use semantic versioning, we might as well use that for AMI as
	  well. That also affords us greater meaning for the AMI version.
	  ........ Merged revisions 404421 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 19:06 +0000 [r404420]  Richard Mudgett <rmudgett@digium.com>

	* /, main/sounds_index.c: Whitespace fixes. ........ Merged
	  revisions 404419 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-20 17:22 +0000 [r404406]  Rusty Newton <rnewton@digium.com>

	* /, configs/pjsip.conf.sample: Documentation: Updates for info
	  about NAT-related settings and fixes for pjsip.conf.sample Added
	  another NAT example to pjsip.conf.sample. We had a few mentions
	  of NAT configuration throughout the sample, but I added another
	  for a little bit more clarity. Additionally many pjsip options
	  were affected by the change to snake case, so I fixed any
	  instances of those options in pjsip.conf. I regenerated the
	  config option list (at the bottom of the file) from a new xml
	  config doc dump, so all the snake case changes should be
	  reflected there, as well as any other changes to those options.
	  (issue ASTERISK-23004) (closes issue ASTERISK-23004) Reported by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/3086/
	  ........ Merged revisions 404405 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 20:48 +0000 [r404387]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/security_events.c: security_events: log events with
	  descriptive names This patch updates the log messages to include
	  descriptive names for event types. This is an improvement over
	  having only cryptic type numbers. (closes issue ASTERISK-22909)
	  Reported by: outtolunc Review:
	  https://reviewboard.asterisk.org/r/3081/ Patches:
	  svn_security_events.c.names.diff.txt uploaded by outtolunc
	  (license 5198)

2013-12-19 18:16 +0000 [r404376]  Richard Mudgett <rmudgett@digium.com>

	* /, CHANGES: Put notice in CHANGES as well as UPGRADE.txt.
	  ........ Merged revisions 404375 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 18:00 +0000 [r404370-404372]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip/pjsip_outbound_auth.c, /: res_pjsip: Ignore 401/407
	  responses for transactions and dialogs we don't know about. Under
	  normal conditions it is unlikely we will ever receive a response
	  for a transaction or dialog we don't know about but if any are
	  received ignore them. ........ Merged revisions 404371 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_session.c: res_pjsip_session: Fix SDP
	  negotiation when resending an INVITE with authentication. The
	  process for resending an INVITE with authentication involves
	  restarting the UAC session. We were incorrectly passing in that a
	  new offer is being sent, causing the SDP negotiation to get into
	  a (technically speaking) funky state. ........ Merged revisions
	  404369 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 17:45 +0000 [r404368]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/channel.h, res/res_pjsip.c, main/channel.c, /,
	  include/asterisk/autochan.h: Fix a deadlock that occurred due to
	  a conflict of masquerades. For the explanation, here is a
	  copy-paste of the review board explanation: Initially, it was
	  discovered that performing an attended transfer of a multiparty
	  bridge with a PJSIP channel would cause a deadlock. A PBX thread
	  started a masquerade and reached the point where it was calling
	  the fixup() callback on the "original" channel. For chan_pjsip,
	  this involves pushing a synchronous task to the session's
	  serializer. The problem was that a task ahead of the fixup task
	  was also attempting to perform a channel masquerade. However,
	  since masquerades are designed in a way to only allow for one to
	  occur at a time, the task ahead of the fixup could not continue
	  until the masquerade already in progress had completed. And of
	  course, the masquerade in progress could not complete until the
	  task ahead of the fixup task had completed. Deadlock. The initial
	  fix was to change the fixup task to be asynchronous. While this
	  prevented the deadlock from occurring, it had the frightful side
	  effect of potentially allowing for tasks in the session's
	  serializer to operate on a zombie channel. Taking a step back
	  from this particular deadlock, it became clear that the problem
	  was not really this one particular issue but that masquerades
	  themselves needed to be addressed. A PJSIP attended transfer
	  operation calls ast_channel_move(), which attempts to both set up
	  and execute a masquerade. The problem was that after it had set
	  up the masquerade, the PBX thread had swooped in and tried to
	  actually perform the masquerade. Looking at changes that had been
	  made to Asterisk 12, it became clear that there never is any time
	  now that anyone ever wants to set up a masquerade and allow for
	  the channel thread to actually perform the masquerade. Everyone
	  always is calling ast_channel_move(), performs the masquerade
	  itself before returning. In this patch, I have removed all blocks
	  of code from channel.c that will attempt to perform a masquerade
	  if ast_channel_masq() returns true. Now, there is no distinction
	  between setting up a masquerade and performing the masquerade. It
	  is one operation. The only remaining checks for
	  ast_channel_masq() and ast_channel_masqr() are in ast_hangup()
	  since we do not want to interrupt a masquerade by hanging up the
	  channel. Instead, now ast_hangup() will wait for a masquerade to
	  complete before moving forward with its operation. The
	  ast_channel_move() function has been modified to basically
	  in-line the logic that used to be in ast_channel_masquerade().
	  ast_channel_masquerade() has been killed off for real.
	  ast_channel_move() now has a lock associated with it that is used
	  to prevent any simultaneous moves from occurring at once. This
	  means there is no need to make sure that ast_channel_masq() or
	  ast_channel_masqr() are already set on a channel when
	  ast_channel_move() is called. It also means the channel container
	  lock is not pulling double duty by both keeping the container
	  locked and preventing multiple masquerades from occurring
	  simultaneously. The ast_do_masquerade() function has been renamed
	  to do_channel_masquerade() and is now internal to channel.c. The
	  function now takes explicit arguments of which channels are
	  involved in the masquerade instead of a single channel. While it
	  probably is possible to do some further refactoring of this
	  method, I feel that I would be treading dangerously. Instead, all
	  I did was change some comments that no longer are true after this
	  changeset. The other more minor change introduced in this patch
	  is to res_pjsip.c to make ast_sip_push_task_synchronous() run the
	  task in-place if we are already a SIP servant thread. This is
	  related to this patch because even when we isolate the channel
	  masquerade to only running in the SIP servant thread, we would
	  still deadlock when the fixup() callback is reached since we
	  would essentially be waiting forever for ourselves to finish
	  before actually running the fixup. This makes it so the fixup is
	  run without having to push a task into a serializer at all.
	  (closes issue ASTERISK-22936) Reported by Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/3069 ........ Merged revisions
	  404356 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 17:13 +0000 [r404355]  Richard Mudgett <rmudgett@digium.com>

	* main/udptl.c, addons/chan_ooh323.c, /, channels/chan_sip.c,
	  include/asterisk/udptl.h: udptl: Dead code elimination.
	  ast_udptl_bridge was not used. Removing dead code starting with
	  ast_udptl_bridge() eliminated the code in this change. Note: This
	  code has actually been dead since Asterisk v1.4 when it was first
	  put in. Review: https://reviewboard.asterisk.org/r/3079/ ........
	  Merged revisions 404354 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 17:03 +0000 [r404353]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, res/res_fax.c: res_fax.c: crash on framehook with no dsp in
	  fax detect In fax_detect_framehook() a null pointer reference can
	  occur where a voice frame is processed but no dsp is attached to
	  the fax detection structure. The code block that rejects frames
	  that detection cannot be processed on is checking for dsp but
	  falls through when it should instead return, as this change
	  implements. (closes issue ASTERISK-22942) Reported by: adomjan
	  Review: https://reviewboard.asterisk.org/r/3076/ ........ Merged
	  revisions 404351 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404352 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 16:52 +0000 [r404350]  Richard Mudgett <rmudgett@digium.com>

	* configs/skinny.conf.sample, res/res_xmpp.c, res/res_jabber.c,
	  CHANGES, channels/chan_iax2.c, channels/sig_pri.c,
	  channels/h323/chan_h323.h, configs/iax.conf.sample,
	  channels/sig_pri.h, channels/chan_dahdi.c,
	  include/asterisk/app.h, channels/chan_skinny.c,
	  channels/chan_dahdi.h, channels/chan_h323.c, main/app.c,
	  UPGRADE-12.txt, configs/sip.conf.sample,
	  channels/sip/include/sip.h, channels/chan_mgcp.c,
	  apps/app_voicemail.c, channels/chan_unistim.c,
	  configs/chan_dahdi.conf.sample, /, channels/chan_sip.c,
	  configs/voicemail.conf.sample, funcs/func_vmcount.c: Voicemail:
	  Remove mailbox identifier format (box@context) assumptions in the
	  system. This change is in preparation for external MWI support.
	  Removed code from the system for normal mailbox handling that
	  appends @default to the mailbox identifier if it does not have a
	  context. The only exception is the legacy hasvoicemail users.conf
	  option. The legacy option will only work for app_voicemail
	  mailboxes. The system cannot make any assumptions about the
	  format of the mailbox identifer used by app_voicemail. chan_sip
	  and chan_dahdi/sig_pri had the most changes because they both
	  tried to interpret the mailbox identifier. chan_sip just stored
	  and compared the two components. chan_dahdi actually used the box
	  information. The ISDN MWI support configuration options had to be
	  reworked because chan_dahdi was parsing the box@context format to
	  get the box number. As a result the mwi_vm_boxes chan_dahdi.conf
	  option was added and is documented in the chan_dahdi.conf.sample
	  file. Review: https://reviewboard.asterisk.org/r/3072/ ........
	  Merged revisions 404348 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 16:33 +0000 [r404346]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/db.c, /: astdb: crash in sqlite3 during shutdown When
	  Asterisk is shut down, the astdb_atexit() function releases
	  (finalize) the previously initiated (prepared) SQL statements in
	  sqlite3. Another thread making a subsequent request can cause a
	  crash in sqlite3. This patch eliminates that issue by resetting
	  the statement pointer after it is released/cleared. The sqlite3
	  code detects the null pointer, and aborts the operation cleanly.
	  (closes issue AST-1265) Reported by: Alexander Hömig (closes
	  issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter
	  Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged
	  revisions 404344 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404345 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 12:18 +0000 [r404333]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: channel: Add a missing ast_channel_unlock when
	  allocating a Surrogate channel. ........ Merged revisions 404332
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 08:35 +0000 [r404321]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooGkClient.c,
	  addons/chan_ooh323.c, /, addons/ooh323c/src/ooGkClient.h: Handle
	  temporary failures on gk registration Introduce new 'stopped'
	  state for gk client and restart gk client on failures Remove
	  ooh323 stack command lock as it is not need now. (closes issue
	  ASTERISK-21960) Reported by: Dmitry Melekhov Patches:
	  ASTERISK-21960.patch ASTERISK-21960-stacklockup-2.patch Tested
	  by: Dmitry Melekhov ........ Merged revisions 404318 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404320 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 02:59 +0000 [r404307]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Fixup some skinny bugs causing Fracks
	  and ao2 cleanup issues. Moved channel locking into setsubstate so
	  that a process can complete working on a sub before another
	  starts changing it. The existing code was causing some Fracks
	  with schedule deletion. Removed multiple rtp cleanup. Now only
	  cleansup up once, fixing ao2 object cleanup issues. ........
	  Merged revisions 404306 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 00:50 +0000 [r404295]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/cdr.h, CHANGES, apps/app_cdr.c, main/cdr.c,
	  apps/app_forkcdr.c, main/pbx.c, /, funcs/func_cdr.c,
	  apps/app_disa.c, UPGRADE-12.txt: app_cdr,app_forkcdr,func_cdr:
	  Synchronize with engine when manipulating state When doing the
	  rework of the CDR engine that pushed all of the logic into cdr.c
	  and made it respond to changes in channel state over Stasis, we
	  knew that accessing the CDR engine from the dialplan would be
	  "slightly" non-deterministic. Dialplan threads would be accessing
	  CDRs while Stasis threads would be updating the state of said
	  CDRs - whereas in the past, everything happened on the dialplan
	  threads. Tests have shown that "slightly" is in reality "very".
	  This patch synchronizes things by making the dialplan
	  applications/functions that manipulate CDRs do so over Stasis.
	  ForkCDR, NoCDR, ResetCDR, CDR, and CDR_PROP now all use Stasis to
	  send their requests over to the CDR engine, and synchronize on
	  the channel Stasis topic via a subscription so that they return
	  their values/control to the dialplan at the appropriate time.
	  While going through this, the following changes were also made: *
	  DISA, which can reset the CDR when a user successfully
	  authenticates, now just uses the ResetCDR app to do this. This
	  prevents having to duplicate the same Stasis synchronization
	  logic in that application. * Answer no longer disables CDRs. It
	  actually didn't work anyway - calling DISABLE on the channel's
	  CDR doesn't stop the CDR from getting the Answer time - it just
	  kills all CDRs on that channel, which isn't what the caller would
	  intend. (closes issue ASTERISK-22884) (closes issue
	  ASTERISK-22886) Review: https://reviewboard.asterisk.org/r/3057/
	  ........ Merged revisions 404294 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-19 00:32 +0000 [r404293]  Damien Wedhorn <voip@facts.com.au>

	* /, channels/chan_skinny.c: Fixup skinny registration following
	  network issues. On session registration, if device is already
	  reporting that it is connected to a device, an innocuous packet
	  (update time) is sent to the already connected device. If the tcp
	  connection is down, the device will be unregistered and the new
	  connection allowed. Without this patch, network issues can see a
	  situation where a device can not reregister until after
	  3*timeout. ........ Merged revisions 404292 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 23:00 +0000 [r404280]  Jason Parker <jparker@digium.com>

	* main/manager.c, /: Add AMI event for presence state. Review:
	  https://reviewboard.asterisk.org/r/3039/ ........ Merged
	  revisions 404275 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404279 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 21:12 +0000 [r404264]  Richard Mudgett <rmudgett@digium.com>

	* addons/ooh323c/src/ooTimer.c, /: ooh323c: Fix gcc 4.6.3 compiler
	  warnings. ........ Merged revisions 404212 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404219 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404263 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 20:48 +0000 [r404260-404262]  Kevin Harwell <kharwell@digium.com>

	* channels/chan_oss.c, /: chan_oss.c: channel being locked twice
	  and unlocked once Removed channel lock as it is now being down in
	  ast_channel_alloc ........ Merged revisions 404261 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c,
	  addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c,
	  channels/chan_pjsip.c, res/parking/parking_manager.c,
	  channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c,
	  funcs/func_timeout.c, /, apps/app_meetme.c, main/bridge.c,
	  tests/test_stasis_channels.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, channels/sig_pri.c, apps/app_queue.c,
	  main/cel.c, main/stasis_bridges.c, channels/chan_jingle.c,
	  channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c,
	  channels/sig_analog.c, include/asterisk/stasis_channels.h,
	  res/res_agi.c, channels/chan_motif.c, tests/test_cel.c,
	  apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c,
	  apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc,
	  addons/chan_ooh323.c, main/pickup.c, include/asterisk/aoc.h,
	  include/asterisk/stasis_bridges.h, apps/app_userevent.c,
	  apps/app_disa.c, channels/chan_console.c,
	  include/asterisk/channelstate.h, main/core_local.c,
	  channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c,
	  res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
	  main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c:
	  channel locking: Add locking for channel snapshot creation
	  Original commit message by mmichelson (asterisk 12 r403311):
	  "This adds channel locks around calls to create channel snapshots
	  as well as other functions which operate on a channel and then
	  end up creating a channel snapshot. Functions that expect the
	  channel to be locked prior to being called have had their
	  documentation updated to indicate such." The above was initially
	  committed and then reverted at r403398. The problem was found to
	  be in core_local.c in the publish_local_bridge_message function.
	  The ast_unreal_lock_all function locks and adds a reference to
	  the returned channels and while they were being unlocked they
	  were not being unreffed when no longer needed. Fixed by unreffing
	  the channels. Also in bridge.c a lock was obtained on
	  "other->chan", but then an attempt was made to unlock "other" and
	  not the previously locked channel. Fixed by unlocking
	  "other->chan" (closes issue ASTERISK-22709) Reported by: John
	  Bigelow ........ Merged revisions 404237 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 19:36 +0000 [r404211]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, configs/ooh323.conf.sample: Introduce new
	  config option 'aniasdni'. If yes then asterisk set dialed number
	  as own id back to the caller on incoming h.323 calls. Option can
	  be set globally or per user section. (closes issue
	  ASTERISK-22020) Reported by: Ross Beer

2013-12-18 19:28 +0000 [r404210]  Joshua Colp <jcolp@digium.com>

	* channels/chan_mgcp.c, main/pbx.c, channels/chan_sip.c,
	  apps/confbridge/conf_chan_record.c, tests/test_app.c,
	  tests/test_stasis_channels.c, main/core_unreal.c,
	  include/asterisk/channel.h, channels/chan_console.c,
	  channels/chan_oss.c, channels/chan_jingle.c,
	  channels/chan_misdn.c, channels/chan_h323.c, tests/test_cel.c,
	  channels/chan_nbs.c, channels/chan_pjsip.c, res/res_calendar.c,
	  apps/app_voicemail.c, channels/chan_unistim.c,
	  tests/test_substitution.c, channels/chan_vpb.cc,
	  addons/chan_ooh323.c, channels/chan_multicast_rtp.c, /,
	  apps/app_meetme.c, res/res_stasis_snoop.c, channels/chan_gtalk.c,
	  channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
	  channels/chan_phone.c, channels/chan_skinny.c,
	  res/parking/parking_tests.c, channels/chan_motif.c,
	  tests/test_voicemail_api.c, channels/chan_alsa.c, main/message.c,
	  addons/chan_mobile.c, tests/test_cdr.c: channels: Return
	  allocated channels locked. This change makes ast_channel_alloc
	  return allocated channels locked. By doing so no other thread can
	  acquire, lock, and manipulate the channel before it is completely
	  set up. (closes issue AST-1256) Review:
	  https://reviewboard.asterisk.org/r/3067/ ........ Merged
	  revisions 404204 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 19:10 +0000 [r404198]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: Implement module reload command for
	  chan_ooh323 (close issue ASTERISK-22817) Patches:
	  ooh323_module_reload.patch

2013-12-18 12:46 +0000 [r404185]  Matthew Jordan <mjordan@digium.com>

	* rest-api/api-docs/applications.json,
	  rest-api/api-docs/playbacks.json,
	  rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
	  rest-api/resources.json, rest-api/api-docs/bridges.json,
	  rest-api/api-docs/recordings.json,
	  rest-api/api-docs/deviceStates.json,
	  rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
	  /, rest-api/api-docs/asterisk.json: ari: Bump the version of ARI
	  to 1.0.0 (closes issue ASTERISK-23007) ........ Merged revisions
	  404184 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 12:01 +0000 [r404138]  Joshua Colp <jcolp@digium.com>

	* res/res_calendar.c, /: res_calendar: Protect channel when adding
	  datastore. This change adds a missing channel lock when adding a
	  datastore to a channel. ........ Merged revisions 404135 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404136 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404137 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 00:36 +0000 [r404100]  Rusty Newton <rnewton@digium.com>

	* /, funcs/func_strings.c: func_strings: Documentation fix for
	  QUOTE() Example output was inaccurate. (issue ASTERISK-22970)
	  (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches:
	  func_strings.patch uploaded by Gareth Palmer (license 5169)
	  ........ Merged revisions 404081 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 404087 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 404099 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-18 00:17 +0000 [r404051]  Matthew Jordan <mjordan@digium.com>

	* /, LICENSE: LICENSE: Update language to include ARI ........
	  Merged revisions 404050 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-17 23:57 +0000 [r404049]  Jonathan Rose <jrose@digium.com>

	* /, tests/test_cel.c, tests/test_cdr.c: tests: fix
	  ast_bridge_base_new calls not using the additional arguments
	  r404042 gave ast_bridge_base_new two new arguments for setting a
	  bridge creator and name. Unfortunately since a couple test
	  modules aren't compiled by default, I missed the fact that this
	  change impacted those tests and caused compilation failures
	  against them. ........ Merged revisions 404048 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-17 23:38 +0000 [r404047]  Rusty Newton <rnewton@digium.com>

	* include/asterisk/test.h, main/channel.c, main/rtp_engine.c, /,
	  channels/chan_iax2.c, apps/app_chanspy.c, apps/app_mixmonitor.c:
	  Several components: fixing Typos in comments and code,
	  "avaliable" instead of "available" (issue ASTERISK-23021) (closes
	  issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty
	  Newton Patches: available.patch uploaded by Jeremy Lainé (license
	  6561) ........ Merged revisions 404046 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-17 23:25 +0000 [r404043]  Jonathan Rose <jrose@digium.com>

	* apps/app_bridgewait.c, res/ari/ari_model_validators.c,
	  doc/appdocsxml.xslt, main/stasis_bridges.c,
	  rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
	  apps/app_agent_pool.c, res/parking/parking_bridge.c,
	  res/ari/ari_model_validators.h, main/manager_bridges.c,
	  res/ari/resource_bridges.h, include/asterisk/bridge_internal.h,
	  apps/app_confbridge.c, res/res_stasis.c,
	  include/asterisk/bridge.h, res/res_ari_bridges.c, /,
	  main/bridge.c, main/bridge_basic.c,
	  include/asterisk/stasis_bridges.h, include/asterisk/stasis_app.h:
	  bridging: Give bridges a name and a known creator Bridges have
	  two new optional properties, a creator and a name. Certain
	  consumers of bridges will automatically provide bridges that they
	  create with these properties. Examples include app_bridgewait,
	  res_parking, app_confbridge, and app_agent_pool. In addition, a
	  name may now be provided as an argument to the POST function for
	  creating new bridges via ARI. (closes issue AFS-47) Review:
	  https://reviewboard.asterisk.org/r/3070/ ........ Merged
	  revisions 404042 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-17 18:35 +0000 [r404028-404030]  Joshua Colp <jcolp@digium.com>

	* res/res_sorcery_config.c, /: res_sorcery_config: Output an error
	  message when an object can't be created. If object creation fails
	  an error message will now be output with the id, type, and
	  configuration file. ........ Merged revisions 404029 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/framehook.c: framehooks: Re-iterate if framehook provides
	  different frame. Framehooks can be used in a reactive manner to
	  execute specific logic when a frame is received with a certain
	  type and payload. Since it is possible for framehooks to provide
	  frames it was possible for this reactive framehook to be unaware
	  of frames it is looking for. This change makes it so that when
	  framehooks return a modified frame the code will now re-iterate
	  (from the beginning) and call any previous framehooks that have
	  not provided a modified frame themselves. Review:
	  https://reviewboard.asterisk.org/r/3046/ ........ Merged
	  revisions 404027 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-17 14:41 +0000 [r404008-404009]  David M. Lee <dlee@digium.com>

	* /, configs/asterisk.conf.sample, main/asterisk.c: Changed the
	  default for live_dangerously to no ........ Merged revisions
	  404006 from http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/pjsip, /: Setting svn:ignore ........ Merged revisions
	  403748 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-17 12:59 +0000 [r403994]  Matthew Jordan <mjordan@digium.com>

	* /, res/ari/resource_channels.c: ari/resource_channels: When
	  creating a channel, specify a default format (SLIN) When creating
	  channels via ARI, the current code fails to provide any default
	  format capabilities. For non-virtual channels this isn't really a
	  problem - the channels typically receive their capabilities as a
	  result of the underlying channel driver configuration. For
	  virtual channels (such as Local channels), the lack of any format
	  capabilities causes the Asterisk core to make some 'odd' choices
	  with respect to the translation paths. The issue reporter had
	  some paths that had 3 hops on each channel leg, causing multiple
	  transcodings and some really crappy audio/performance. By
	  specifying a baseline of SLIN, we prevent that from occurring.
	  Note that this is what AMI does when it performs an Originate, as
	  does res_clioriginate. Review:
	  https://reviewboard.asterisk.org/r/3068/ (issue ASTERISK-22962)
	  Reported by: Matt DiMeo ........ Merged revisions 403993 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-16 19:11 +0000 [r403960]  David M. Lee <dlee@digium.com>

	* include/asterisk/pbx.h, main/asterisk.c, funcs/func_realtime.c,
	  main/pbx.c, main/tcptls.c, funcs/func_db.c, /,
	  README-SERIOUSLY.bestpractices.txt, configs/asterisk.conf.sample,
	  funcs/func_shell.c, funcs/func_env.c, funcs/func_lock.c,
	  UPGRADE-12.txt: security: Inhibit execution of privilege
	  escalating functions This patch allows individual dialplan
	  functions to be marked as 'dangerous', to inhibit their execution
	  from external sources. A 'dangerous' function is one which
	  results in a privilege escalation. For example, if one were to
	  read the channel variable SHELL(rm -rf /) Bad Things(TM) could
	  happen; even if the external source has only read permissions.
	  Execution from external sources may be enabled by setting
	  'live_dangerously' to 'yes' in the [options] section of
	  asterisk.conf. Although doing so is not recommended. Also, the
	  ABI was changed to something more reasonable, since Asterisk 12
	  does not yet have a public release. (closes issue ASTERISK-22905)
	  Review: http://reviewboard.digium.internal/r/432/ ........ Merged
	  revisions 403913 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 403917 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 403959 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-16 18:31 +0000 [r403958]  Jonathan Rose <jrose@digium.com>

	* /, main/bridge.c: transfers: Fix bug setting both BLINDTRANSFER
	  and ATTENDEDTRANSFER The ast_bridge_set_transfer_variables
	  function is supposed to wipe whichever variable isn't being set.
	  Instead it was setting both to the new value. Oops. (issue
	  AFS-24) ........ Merged revisions 403957 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-16 16:12 +0000 [r403857-403865]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to
	  prevent memory corruption During dialplan execution in
	  pbx_extension_helper(), the contexts global read lock prevents
	  link list corruption, but was released with a pointer to the
	  ast_exten and data later used in variable substitution. Instead,
	  this patch removes pbx_substitute_variables() and locates a copy
	  of the ast_exten data on the stack before releasing the lock,
	  where ast_exten could get free'd by another thread performing a
	  module reload. (issue AST-1179) Reported by: Thomas Arimont
	  (issue AST-1246) Reported by: Alexander Hömig Review:
	  https://reviewboard.asterisk.org/r/3055/ ........ Merged
	  revisions 403862 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 403863 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 403864 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, apps/app_sms.c: app_sms: BufferOverflow when receiving odd
	  length 16 bit message This patch prevents an infinite loop
	  overwriting memory when a message is received into the
	  unpacksms16() function, where the length of the message is an odd
	  number of bytes. (closes issue ASTERISK-22590) Reported by: Jan
	  Juergens Tested by: Jan Juergens ........ Merged revisions 403856
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-15 01:39 +0000 [r403824]  Matthew Jordan <mjordan@digium.com>

	* channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions:
	  Use the right buffer length when printing URIs While
	  entertaining, sizeof(buflen) is not the same as buflen. Doh.
	  ........ Merged revisions 403823 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-14 17:28 +0000 [r403810-403812]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c,
	  res/res_pjsip/pjsip_options.c, res/res_pjsip.c: res_pjsip: Apply
	  outbound proxy to all SIP requests. Objects which are involved in
	  SIP request creation and sending now allow an outbound proxy to
	  be specified. For cases where an endpoint is used the outbound
	  proxy specified there will be applied. (closes issue
	  ASTERISK-22673) Reported by: Antti Yrjola Review:
	  https://reviewboard.asterisk.org/r/3022/ ........ Merged
	  revisions 403811 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis_channels.c, apps/app_queue.c,
	  res/ari/ari_model_validators.c, apps/app_dial.c,
	  res/ari/ari_model_validators.h, main/dial.c,
	  include/asterisk/stasis_channels.h,
	  rest-api/api-docs/events.json, /, res/stasis/app.c: res_stasis:
	  Expose event for call forwarding and follow forwarded channel.
	  This change adds an event for when an originated call is
	  redirected to another target. This event contains the original
	  channel and the newly created channel. If a stasis subscription
	  exists on the original originated channel for a stasis
	  application then a new subscription will also be created on the
	  stasis application to the redirected channel. This allows the
	  application to follow the call path completely. (closes issue
	  ASTERISK-22719) Reported by: Joshua Colp Review:
	  https://reviewboard.asterisk.org/r/3054/ ........ Merged
	  revisions 403808 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-13 21:35 +0000 [r403797]  Jonathan Rose <jrose@digium.com>

	* /, res/res_pjsip_messaging.c, main/message.c: documentation: Add
	  PJSIP technology to messaging documentation ........ Merged
	  revisions 403796 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-13 20:17 +0000 [r403784]  Richard Mudgett <rmudgett@digium.com>

	* /, main/test.c: test.c: Fix too sticky unit test failed status.
	  Rerunning a failed unit test after loading any required modules
	  should allow the test to report a pass status if it now passes.
	  ........ Merged revisions 403782 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-13 20:13 +0000 [r403783]  Jonathan Rose <jrose@digium.com>

	* /, main/bridge.c, main/bridge_basic.c, include/asterisk/bridge.h,
	  res/parking/parking_bridge_features.c,
	  res/parking/parking_manager.c: Transfers: Make Asterisk set
	  ATTENDEDTRANSFER/BLINDTRANSFER more reliably There were still a
	  few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be
	  set on channels involved with blind and attended transfers. This
	  would happen with features that were initialized by channel
	  driver specific mechanisms in multiparty calls. This patch
	  resolves those cases while attempted to keep the behavior for
	  setting those variables as consistent as possible. (closes issue
	  AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ ........
	  Merged revisions 403781 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-13 18:33 +0000 [r403750-403768]  Kevin Harwell <kharwell@digium.com>

	* main/channel.c, /, channels/chan_sip.c,
	  include/asterisk/channel.h, bridges/bridge_native_rtp.c,
	  channels/chan_pjsip.c: bridge_native_rtp: Deadlock during 4-way
	  conference creation The change contains a slightly adjusted patch
	  that was on the issue (submitted by kmoore). A fix was made by
	  adding in a bridge lock while calling bridge_start/stop from the
	  framehook callback. Since the framehook callback is not called
	  from the bridging core the bridge is not locked, but needs to be
	  before calling bridge_start. (closes issue ASTERISK-22749)
	  Reported by: Kinsey Moore Review:
	  https://reviewboard.asterisk.org/r/3066/ Patches:
	  lock_inversion.diff uploaded by kmoore (license 6273) ........
	  Merged revisions 403767 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* rest-api/api-docs/channels.json, res/ari/resource_channels.c,
	  res/res_ari_channels.c, res/ari/resource_channels.h, /,
	  main/http.c: ARI: Allow specifying channel variables during a
	  POST /channels Added the ability to specify channel variables
	  when creating/originating a channel in ARI. The variables are
	  sent in the body of the request and should be formatted as a
	  single level JSON object. No nested objects allowed. For example:
	  {"variable1": "foo", "variable2": "bar"}. (closes issue
	  ASTERISK-22872) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3052/ ........ Merged
	  revisions 403752 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis_answer.c, rest-api/api-docs/bridges.json,
	  res/ari/resource_bridges.c, res/res_ari_bridges.c,
	  res/stasis/command.c, res/res_stasis_playback.c, /,
	  res/stasis/control.c, res/stasis/command.h,
	  include/asterisk/stasis_app.h,
	  include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c:
	  ARI: Adding a channel to a bridge while a live recording is
	  active blocks Added the ability to have rules that are checked
	  when adding and/or removing channels to/from a bridge. In this
	  case, if a channel is currently recording and someone attempts to
	  add it to a bridge an "is recording" rule is checked, fails, and
	  a 409 conflict is returned. Also command functions now return an
	  integer value that can be descriptive of what kind of problems,
	  if any, occurred before or during execution. (closes issue
	  ASTERISK-22624) Reported by: Joshua Colp Review:
	  https://reviewboard.asterisk.org/r/2947/ ........ Merged
	  revisions 403749 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-13 05:00 +0000 [r403737]  Matthew Jordan <mjordan@digium.com>

	* /, channels/Makefile: channels/Makefile: clean pjsip directory
	  ........ Merged revisions 403736 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-13 00:40 +0000 [r403726]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/app.h, tests/test_voicemail_api.c, main/app.c:
	  test_voicemail_api: Add check for a registered voicemail provider
	  before tests. It is much nicer diagnosing a test failure if
	  app_voicemail is actually loaded.

2013-12-12 19:46 +0000 [r403714]  Scott Griepentrog <sgriepentrog@digium.com>

	* contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py
	  (added), /: realtime: Create extensions in alembic ast-db-manage
	  contribution When the alembic scripts were written for creating
	  Asterisk realtime databases the extensions table for dialplan
	  wasn't included. This update creates the extensions table.
	  (closes issue ASTERISK-22815) Reported by: Zone Conkle Review:
	  https://reviewboard.asterisk.org/r/3064/ ........ Merged
	  revisions 403713 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-12 19:18 +0000 [r403707]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_pjsip.c: chan_pjsip: Revert r403587 This patch
	  was intended to eliminate a deadlock that occurs when masquerades
	  occur in pjsip channels, but has some potential side effects.
	  Mark Michelson is currently working on addressing this problem
	  from another angle. (issue ASTERISK-22936) Reported by: Jonathan
	  Rose ........ Merged revisions 403705 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-11 20:24 +0000 [r403687]  Kevin Harwell <kharwell@digium.com>

	* include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, /,
	  configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip_messaging.c,
	  res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c:
	  res_pjsip_messaging: send message to a default outbound endpoint
	  In some cases messages need to be sent to a direct URI (sip:<ip
	  address>). This patch adds in that support by using a default
	  outbound endpoint. When sending messages, if no endpoint can be
	  found then the default one is used. To facilitate this a new
	  default_outbound_endpoint option was added to the globals section
	  for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/
	  ........ Merged revisions 403680 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-11 19:22 +0000 [r403652]  Russell Bryant <russell@russellbryant.com>

	* /, channels/chan_sip.c: Reset peer outboundproxy on sip.conf
	  reload If you set a peer's outboundproxy and then removed it from
	  the config, this would not get picked up in a config reload. This
	  patch fixes that by resetting it in set_peer_defaults(). Closes
	  ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/
	  ........ Merged revisions 403634 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 403635 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 403639 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-11 19:19 +0000 [r403643]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_voicemail.c, include/asterisk/app.h,
	  include/asterisk/doxyref.h, main/app.c: app_voicemail: Voicemail
	  callback registration/unregistration function improvements. * The
	  voicemail registration/unregistration functions now take a struct
	  of callbacks instead of a lengthy parameter list of callbacks. *
	  The voicemail registration/unregistration functions now prevent a
	  competing module from interfering with an already registered
	  callback supplying module.

2013-12-11 13:06 +0000 [r403617-403619]  Matthew Jordan <mjordan@digium.com>

	* channels/pjsip/dialplan_functions.c,
	  include/asterisk/res_pjsip_session.h, channels/pjsip (added), /,
	  funcs/func_channel.c, channels/pjsip/include,
	  channels/pjsip/include/dialplan_functions.h, res/res_pjsip_t38.c,
	  channels/pjsip/include/chan_pjsip.h, channels/Makefile,
	  channels/chan_pjsip.c, main/xmldoc.c: func_channel, chan_pjsip:
	  Add CHANNEL read function support for chan_pjsip This patch adds
	  CHANNEL read support for chan_pjsip. This allows the dialplan to
	  use the CHANNEL function on a chan_pjsip channel to obtain
	  run-time information about the channel from the PJSIP channel
	  driver and the PJSIP stack. This includes: * RTP information,
	  including source/destination media addresses, whether or not the
	  media is secure, held, and other properties. * RTCP information.
	  This includes sets of parseable information, as well as
	  individual statistic attriutes. * PJSIP information. This
	  includes URIs, local/remote signalling addresses, whether or not
	  the signalling is secure, and other properties. * The endpoint
	  name. This can be used in conjunction with the PJSIP_ENDPOINT
	  function to obtain more detailed endpoint information. Review:
	  https://reviewboard.asterisk.org/r/3038/ ........ Merged
	  revisions 403618 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* Makefile, funcs/func_pjsip_endpoint.c (added), doc/snapshots.xslt
	  (removed), /, doc/appdocsxml.xslt (added), doc/appdocsxml.dtd,
	  main/sorcery.c: func_pjsip_endpoint: Add PJSIP_ENDPOINT function
	  for querying endpoint details This patch adds a new function,
	  PJSIP_ENDPOINT, which lets the dialplan query, for any endpoint,
	  any property configured on an endpoint. This function is a
	  companion to the CHANNEL function, which can be used to extract
	  the endpoint name for a channel. Review:
	  https://reviewboard.asterisk.org/r/3035 ........ Merged revisions
	  403616 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-10 15:15 +0000 [r403605]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_authenticator_digest.c: Fix correct authentication
	  behavior for artificial endpoint. When switching to using a
	  vector for authentication, I initialized the vector for the
	  artificial endpoint to be of size 1. However, this does not
	  result in AST_VECTOR_SIZE() returning 1 since there isn't
	  actually anything in the vector. Rather than trifle with the
	  vector by putting unnecessary elements in, I simply changed the
	  callback in res_pjsip_authenticator_digest.c to explicitly report
	  that the artificial endpoint requires authentication. Thanks to
	  Joshua Colp for pointing this out.

2013-12-09 22:59 +0000 [r403576-403588]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_pjsip.c: chan_pjsip: Fix a sticking channel lock
	  caused by channel masquerades (closes issue ASTERISK-22936)
	  Reported by: Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/3042/ ........ Merged
	  revisions 403587 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* CHANGES, main/dial.c, apps/app_page.c, include/asterisk/dial.h:
	  app_page: Add predial handlers for app_page. (closes issue
	  AFS-14) Review: https://reviewboard.asterisk.org/r/3045/

2013-12-09 19:24 +0000 [r403544-403560]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_sorcery_astdb.c: Reverting regex part of -r403545 at
	  request of file. res_sorcery_astdb.c: Fix get multiple records by
	  regex. * Fix sorcery_astdb_retrieve_regex() pattern matching. Let
	  the regexec() function match the stored key values instead of
	  having astdb prefilter them. Previoiusly you could only use a
	  simple regex pattern when the pattern began with '^'. ........
	  Merged revisions 403559 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix get multiple
	  records by regex. * Fix sorcery_astdb_retrieve_regex() pattern
	  matching. Let the regexec() function match the stored key values
	  instead of having astdb prefilter them. Previoiusly you could
	  only use a simple regex pattern when the pattern began with '^'.
	  * Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
	  ........ Merged revisions 403545 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/sorcery.c, /: sorcery: Eliminate shadowing a varaible that
	  caused confusion. * Eliminated shadowing of the
	  __ast_sorcery_apply_config() name parameter causing confusion. *
	  Fix potential crash from sorcery.conf user input in
	  __ast_sorcery_apply_config() if the user supplied a malformed
	  config line that is missing the sorcery object type name. *
	  Remove redundant test in __ast_sorcery_apply_config(). !config
	  and config == CONFIGS_STATUS_FILEMISSING are identical. ........
	  Merged revisions 403541 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-09 18:32 +0000 [r403543]  Joshua Colp <jcolp@digium.com>

	* /, main/endpoints.c: endpoints: Keep a reference to channel ids
	  when creating snapshot. The snapshot process for endpoints uses
	  the channel ids present on the endpoint itself. Without keeping a
	  reference it was possible for the strings to be freed underneath
	  any consumer of an endpoint snapshot. A reference is now held by
	  the snapshot to the channel ids and released when the snapshot is
	  destroyed. (issue ASTERISK-22801) Reported by: Matt Jordan
	  ........ Merged revisions 403542 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-09 18:14 +0000 [r403528]  Richard Mudgett <rmudgett@digium.com>

	* main/sorcery.c, /: sorcery: Whitespace You would think that a new
	  file would start off without any whitespace oddities. ........
	  Merged revisions 403527 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-09 17:29 +0000 [r403512-403526]  Mark Michelson <mmichelson@digium.com>

	* apps/app_confbridge.c, CHANGES,
	  apps/confbridge/conf_state_multi_marked.c: Add a
	  CONFBRIDGE_RESULT channel variable to discern why a channel left
	  a ConfBridge. Review: https://reviewboard.asterisk.org/r/3009

	* CHANGES, apps/app_mixmonitor.c: Create function for retrieving
	  Mixmonitor instance data. For the time, this is only useful for
	  retrieving the filename. The purpose of this function is to
	  better facilitate multiple mixmonitors per channel. Setting a
	  MIXMONITOR_FILENAME channel variable is not conducive to such
	  behavior, so allowing finer grained access to individual
	  mixmonitor properties improves the situation. The
	  MIXMONITOR_FILENAME channel variable is still set, though, so
	  there is no worry about backwards compatibility. Review:
	  https://reviewboard.asterisk.org/r/3023

2013-12-09 16:41 +0000 [r403511]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_nat.c, /: res_pjsip_nat: Add NAT module to session
	  dialogs. Due to the way pjproject internally works it was
	  possible for the NAT module to not be invoked on messages with-in
	  a session dialog. This means that the various parts of the
	  message would not get rewritten with the source IP address and
	  port. This change uses a session supplement to add the NAT module
	  to the dialog on the first incoming or outgoing INVITE. (closes
	  issue ASTERISK-22941) Reported by: Leif Madsen ........ Merged
	  revisions 403510 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-09 16:10 +0000 [r403499]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip/config_auth.c,
	  res/res_pjsip_outbound_authenticator_digest.c,
	  res/res_pjsip_authenticator_digest.c,
	  res/res_pjsip_outbound_registration.c,
	  res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip/pjsip_distributor.c, res/res_pjsip.c,
	  include/asterisk/res_pjsip.h: Switch PJSIP auth to use a vector.
	  Since Asterisk has a vector API now, places where arrays are
	  manually resized don't really make sense any more. Since the auth
	  work in PJSIP was freshly-written, it was easy to reform it to
	  use a vector. Review: https://reviewboard.asterisk.org/r/3044

2013-12-09 03:21 +0000 [r403436-403466]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_fax_spandsp.c: res_fax_spandsp: Always init T.38
	  session to avoid crashes during state change Prior to this patch,
	  res_fax_spandsp was conservative with how it initialized the
	  spandsp T.38 context. It would only initialize it if the driver
	  thought the current state was a T.38 fax. While this works fine
	  in nominal situations, in certain off nominal situations,
	  res_fax_spandsp can believe that a T.38 fax will not occur when
	  in fact one has started. In particular, this was discovered when
	  res_fax would fall back to audio after timing out on a T.38
	  upgrade. The SIP channel driver would continue to retry the
	  re-INVITE and - if the remote end responded after res_fax timed
	  out with a 200 OK - a T.38 frame would be delivered to the
	  res_fax stack when it no longer expected it. As it turns out,
	  there does not appear to be any downside to always initializing
	  the T.38 context, other than the actual memory allocation. Since
	  that avoids this off nominal situation (and others which are
	  equally likely hard to predict), this is the safest way to avoid
	  this problem. Much thanks to Torrey as well for providing a
	  scenario that reproduces this issue. (closes issue
	  ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey
	  Searle patches: always-init-t38.patch uploaded by awinters
	  (License 6477) A_PARTY.xml uploaded by tsearle (License 5334)
	  ........ Merged revisions 403449 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 403450 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 403458 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_config_sqlite.c: res_config_sqlite: Check for CDR
	  unregistration failures If the CDR unregistration fails due to an
	  inflight CDR, the res_config_sqlite module needs to bail on
	  unloading itself. Otherwise, the config could be unloaded
	  (including the CDR table name) while the CDR engine posts a CDR
	  to the still registered backend, resulting in a crash. ........
	  Merged revisions 403435 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-05 23:40 +0000 [r403414]  Jonathan Rose <jrose@digium.com>

	* apps/app_record.c: app_record: Add an option that allows DTMF '0'
	  to act as an additional terminator Using this terminator when
	  active results in ${RECORD_STATUS} being set to 'OPERATOR'
	  instead of 'DTMF' (closes issue AFS-7) Review:
	  https://reviewboard.asterisk.org/r/3041/

2013-12-05 22:10 +0000 [r403402-403404]  David M. Lee <dlee@digium.com>

	* addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c,
	  channels/chan_pjsip.c, res/parking/parking_manager.c,
	  channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c, /,
	  apps/app_meetme.c, funcs/func_timeout.c, main/bridge.c,
	  tests/test_stasis_channels.c, main/core_unreal.c,
	  include/asterisk/channel.h, channels/chan_gtalk.c, main/cel.c,
	  apps/app_queue.c, channels/sig_pri.c, main/stasis_bridges.c,
	  channels/chan_jingle.c, channels/chan_phone.c,
	  channels/chan_dahdi.c, main/dial.c, channels/sig_analog.c,
	  include/asterisk/stasis_channels.h, res/res_agi.c,
	  channels/chan_motif.c, channels/chan_h323.c, tests/test_cel.c,
	  apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c,
	  apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc,
	  addons/chan_ooh323.c, channels/chan_sip.c, main/pickup.c,
	  include/asterisk/aoc.h, include/asterisk/stasis_bridges.h,
	  apps/app_userevent.c, apps/app_disa.c, main/core_local.c,
	  include/asterisk/channelstate.h, channels/chan_console.c,
	  channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c,
	  res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
	  main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
	  pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c,
	  channels/chan_nbs.c: Reverting r403311. It's causing ARI tests to
	  hang. ........ Merged revisions 403398 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/stasis/control.c: ari: Fix deadlock problem with functions
	  that use autoservice. The code for getting channel variables from
	  ARI assumed that you needed to lock the channel in order to
	  properly execute functions and read channel variables.
	  Apparently, this is not the case, since any dialplan function
	  that puts the channel into autoservice deadlocks when attempting
	  to remove the channel from autoservice. ........ Merged revisions
	  403342 from http://svn.asterisk.org/svn/asterisk/branches/12

	* /: Multiple revisions 403304,403310 ........ r403304 | dlee |
	  2013-12-02 12:34:50 -0600 (Mon, 02 Dec 2013) | 1 line Fixed the
	  filename for the ari.conf docs ........ r403310 | file |
	  2013-12-03 10:32:12 -0600 (Tue, 03 Dec 2013) | 5 lines Revert
	  revision 403304: Fixed the filename for the ari.conf docs The
	  changed value refers to the name of the module. The name of the
	  configuration file is specified in the configFile section.
	  ........ Merged revisions 403304,403310 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-04 21:42 +0000 [r403378]  Kevin Harwell <kharwell@digium.com>

	* /, res/res_pjsip_registrar.c: res_pjsip_registrar: undefined
	  function pointer symbol Used a static wrapper around the
	  offending function to alleviate the issue. Reported by: rmudgett
	  ........ Merged revisions 403377 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-04 20:54 +0000 [r403365]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_t38.c, /: res_pjsip_t38: Don't pass T.38 control
	  frames through to other hooks. This crept up during gateway
	  testing where the gateway would receive the request to negotiate
	  and assume it came from the remote side, causing the gateway
	  state machine to go a little, to a use a technical term, "wonky".
	  ........ Merged revisions 403364 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-04 18:41 +0000 [r403350]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip.c: Initialize the hash value argument to
	  pj_hash_get() to 0. Passing a non-zero value causes PJLIB to use
	  the given input as the hash value. Passing zero causes the
	  parameter to become an output parameter that receives the hash
	  value that was computed based on the given key. This change
	  essentially makes ast_sip_dict_get() properly retrieve the
	  desired value. ........ Merged revisions 403349 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-03 18:01 +0000 [r403330]  Joshua Colp <jcolp@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  res/res_pjsip_session.c: res_pjsip_session: Add support for
	  PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag. Newer versions of PJSIP
	  have changed to using a flag for the
	  PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds
	  a configure check to detect the presence of the flag and use it
	  if found. ........ Merged revisions 403329 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-03 17:35 +0000 [r403327]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip_registrar_expire.c, res/res_pjsip/pjsip_options.c,
	  tests/test_sorcery.c, include/asterisk/bucket.h, main/sorcery.c,
	  /, main/bucket.c: sorcery, bucket: Change observer remove calls
	  to take const callbacks struct. * Make
	  ast_sorcery_observer_remove() accept a const callbacks struct. *
	  Make ast_sorcery_observer_remove() tolerant of the sorcery
	  parameter being NULL. Now it can be called within a module unload
	  routine if the sorcery initialization fails. * Fix
	  ast_sorcery_observer_add() to fail if the container link fails.
	  ........ Merged revisions 403324 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-03 17:07 +0000 [r403314]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_nbs.c, main/bridge_channel.c, res/res_stasis.c,
	  channels/chan_pjsip.c, res/parking/parking_manager.c,
	  apps/app_voicemail.c, channels/chan_unistim.c,
	  channels/chan_vpb.cc, addons/chan_ooh323.c, /,
	  include/asterisk/aoc.h, apps/app_meetme.c, main/bridge.c,
	  apps/app_userevent.c, channels/chan_gtalk.c,
	  channels/chan_iax2.c, main/endpoints.c, main/stasis_bridges.c,
	  main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
	  main/dial.c, channels/sig_analog.c, channels/chan_skinny.c,
	  res/res_agi.c, channels/chan_motif.c, pbx/pbx_realtime.c,
	  channels/chan_alsa.c, main/stasis_channels.c,
	  apps/app_confbridge.c, addons/chan_mobile.c, tests/test_cdr.c,
	  res/res_pjsip_refer.c, channels/chan_mgcp.c, apps/app_dial.c,
	  main/pbx.c, channels/chan_sip.c, main/pickup.c,
	  funcs/func_timeout.c, tests/test_stasis_channels.c,
	  main/core_unreal.c, include/asterisk/stasis_bridges.h,
	  apps/app_disa.c, include/asterisk/channel.h, main/core_local.c,
	  include/asterisk/channelstate.h, channels/chan_console.c,
	  main/cel.c, apps/app_queue.c, channels/sig_pri.c,
	  channels/chan_oss.c, res/parking/parking_bridge_features.c,
	  apps/app_agent_pool.c, channels/chan_jingle.c,
	  channels/chan_misdn.c, include/asterisk/stasis_channels.h,
	  channels/chan_h323.c, tests/test_cel.c: Add channel locking for
	  channel snapshot creation. This adds channel locks around calls
	  to create channel snapshots as well as other functions which
	  operate on a channel and then end up creating a channel snapshot.
	  Functions that expect the channel to be locked prior to being
	  called have had their documentation updated to indicate such.
	  ........ Merged revisions 403311 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-03 16:39 +0000 [r403313]  Joshua Colp <jcolp@digium.com>

	* main/media_index.c, /: media_index: Make media indexing tolerable
	  of bad symlinks. Media indexing will now skip over files and
	  directories that stat will not return information about. This can
	  occur under normal conditions when a symbolic link points to a
	  location that no longer exists. ........ Merged revisions 403312
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-02 18:12 +0000 [r403292]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: Check and reject non-digits e164 values
	  on peers and general sections in ooh323.conf Regenerate e164
	  endpoint list on reload ooh323 (issue ASTERISK-22901) Reported
	  by: Cyril CONSTANTIN Patches: ASTERISK-22901.patch ........
	  Merged revisions 403288 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 403290 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-12-01 21:13 +0000 [r403257-403272]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_session.c: res_pjsip_session: Apply fromuser and
	  fromdomain to all requests as documented. ........ Merged
	  revisions 403271 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_t38.c, /: res_pjsip_t38: Add the framehook to the
	  channel only on first INVITE. The check for determining whether
	  the T.38 framehook should be added to the channel or not has now
	  been changed to guarantee adding only occurs on the first
	  incoming or outgoing INVITE. ........ Merged revisions 403258
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip/security_events.c, res/res_pjsip/pjsip_options.c,
	  res/res_pjsip.c, res/res_pjsip_transport_websocket.c,
	  include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c:
	  res_pjsip_transport_websocket: Fix security events and simplify
	  implementation. Transport type determination for security events
	  has been simplified to use the type present on the message itself
	  instead of searching through configured transports to find the
	  transport used. The actual WebSocket transport has also been
	  simplified. It now leverages the existing PJSIP transport manager
	  for finding the active WebSocket transport for outgoing messages.
	  This removes the need for res_pjsip_transport_websocket to store
	  a mapping itself. (closes issue ASTERISK-22897) Reported by: Max
	  E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/
	  ........ Merged revisions 403256 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-30 14:12 +0000 [r403241]  Joshua Colp <jcolp@digium.com>

	* res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /,
	  res/ari/ari_model_validators.c: res_ari: Add Recording events to
	  the validator. ........ Merged revisions 403240 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-28 02:12 +0000 [r403208-403224]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't produce an
	  invalid media stream with no formats. Depending on configuration
	  it was possible for a media stream to be created without any
	  media formats. The produced SDP would fail internal validation
	  and cause a crash. The code will now no longer add media streams
	  with no formats to the SDP, allowing it to pass validation and
	  work. (closes issue ASTERISK-22858) Reported by: Anthony Messina
	  ........ Merged revisions 403223 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_header_funcs.c, /: res_pjsip_header_funcs: Don't
	  add headers to re-INVITEs. When sending a re-INVITE to an
	  endpoint it was possible for received headers to be added as well
	  (since they are stored for retrieval using the PJSIP_HEADER
	  dialplan function). This caused a broken (and potentially large)
	  SIP INVITE to be produced and sent. This changes the module so it
	  will no longer add headers to re-INVITEs. (closes issue
	  ASTERISK-22882) Reported by: David M. Lee ........ Merged
	  revisions 403221 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis_playback.c, /: res_stasis_playback: Add 'number',
	  'digits', and 'characters' URI scheme implementations. This
	  change adds new URI scheme implementations for playing numbers,
	  digits, and characters. This is done as part of the normal
	  playback mechanism and can be used with queueing to create a
	  combined sentence. Review:
	  https://reviewboard.asterisk.org/r/3028/ ........ Merged
	  revisions 403209 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c,
	  res/res_pjsip_session.c, include/asterisk/res_pjsip.h:
	  res_pjsip_session: Add configurable behavior for redirects. The
	  action taken when a redirect occurs is now configurable on a
	  per-endpoint basis. The redirect can either be treated as a
	  redirect to a local extension, to a URI that is dialed through
	  the Asterisk core, or to a URI that is dialed within PJSIP
	  itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/2963/ ........ Merged
	  revisions 403207 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-27 17:32 +0000 [r403192]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astdb.h: astdb: Tweak some doxygen comments.

2013-11-27 16:12 +0000 [r403180]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix crash when
	  reloading certain configurations. Certain options available that
	  specify a SIP URI perform validation on the provided URI using
	  the PJSIP URI parser. This operation requires that the thread
	  executing it be registered with the PJLIB library. During reloads
	  this was done on a thread which was NOT registered with it. This
	  fixes the problem by creating a task which reloads the
	  configuration on a PJSIP thread. (closes issue ASTERISK-22923)
	  Reported by: Anthony Messina ........ Merged revisions 403179
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-27 15:48 +0000 [r403177]  David M. Lee <dlee@digium.com>

	* res/res_ari_channels.c, include/asterisk/ari.h,
	  rest-api-templates/param_parsing.mustache,
	  include/asterisk/http.h, res/res_ari_recordings.c,
	  res/res_ari_endpoints.c, main/http.c,
	  rest-api-templates/swagger_model.py, res/res_ari_playbacks.c,
	  res/res_ari_sounds.c, rest-api-templates/asterisk_processor.py,
	  res/res_ari_bridges.c, tests/test_ari.c, res/res_ari.c, /,
	  res/res_ari_device_states.c, res/res_ari_asterisk.c,
	  rest-api-templates/res_ari_resource.c.mustache,
	  res/res_ari_applications.c: ari:Add application/json parameter
	  support The patch allows ARI to parse request parameters from an
	  incoming JSON request body, instead of requiring the request to
	  come in as query parameters (which is just weird for POST and
	  DELETE) or form parameters (which is okay, but a bit asymmetric
	  given that all of our responses are JSON). For any operation that
	  does _not_ have a parameter defined of type body (i.e.
	  "paramType": "body" in the API declaration), if a request
	  provides a request body with a Content type of
	  "application/json", the provided JSON document is parsed and
	  searched for parameters. The expected fields in the provided JSON
	  document should match the query parameters defined for the
	  operation. If the parameter has 'allowMultiple' set, then the
	  field in the JSON document may optionally be an array of values.
	  (closes issue ASTERISK-22685) Review:
	  https://reviewboard.asterisk.org/r/2994/

2013-11-27 15:31 +0000 [r403161-403174]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Update
	  handling of some options to work with new option names. Some
	  options (such as call_group and pickup_group) share the same
	  configuration handler and decide what logic to use based on the
	  name of the option. These handlers were not updated to check for
	  the new option names and were treating the options as invalid.
	  This change simply updates the handlers with the proper names of
	  the options. (closes issue ASTERISK-22922) Reported by: Anthony
	  Messina ........ Merged revisions 403173 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac: Fix
	  a configure issue with PJSIP transaction group lock detection.
	  The configure check did not use the provided paths for pjproject
	  if provided when looking for transaction group lock support.
	  ........ Merged revisions 403160 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-23 17:48 +0000 [r403133-403135]  Kevin Harwell <kharwell@digium.com>

	* res/ari.make, rest-api/api-docs/applications.json,
	  res/ari/resource_device_states.h (added),
	  include/asterisk/stasis_app_device_state.h (added),
	  res/ari/resource_applications.h, res/res_stasis.c,
	  include/asterisk/devicestate.h, rest-api/api-docs/events.json,
	  res/res_stasis_device_state.exports.in (added), res/stasis/app.c,
	  res/res_ari_device_states.c (added), /,
	  include/asterisk/stasis_app.h, main/devicestate.c,
	  res/stasis/app.h, rest-api/resources.json,
	  res/res_stasis_device_state.c (added),
	  res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
	  res/ari/resource_device_states.c (added),
	  rest-api/api-docs/deviceStates.json (added),
	  rest-api-templates/ari.make.mustache: ARI: Implement device state
	  API Created a data model and implemented functionality for an ARI
	  device state resource. The following operations have been added
	  that allow a user to manipulate an ARI controlled device:
	  Create/Change the state of an ARI controlled device PUT
	  /deviceStates/{deviceName}&{deviceState} Retrieve all ARI
	  controlled devices GET /deviceStates Retrieve the current state
	  of a device GET /deviceStates/{deviceName} Destroy a device-state
	  controlled by ARI DELETE /deviceStates/{deviceName} The ARI
	  controlled device must begin with 'Stasis:'. An example
	  controlled device name would be Stasis:Example. A
	  'DeviceStateChanged' event has also been added so that an
	  application can subscribe and receive device change events. Any
	  device state, ARI controlled or not, can be subscribed to. While
	  adding the event, the underlying subscription control mechanism
	  was refactored so that all current and future resource
	  subscriptions would be the same. Each event resource must now
	  register itself in order to be able to properly handle
	  [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/3025/ ........ Merged
	  revisions 403134 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_registrar.c, main/sorcery.c,
	  include/asterisk/res_pjsip.h, include/asterisk/acl.h,
	  res/res_pjsip/config_auth.c, include/asterisk/utils.h,
	  res/res_pjsip.exports.in, /,
	  res/res_pjsip_endpoint_identifier_ip.c, main/acl.c, main/utils.c,
	  res/res_pjsip.c, res/res_pjsip_exten_state.c,
	  include/asterisk/res_pjsip_pubsub.h, res/res_pjsip/location.c,
	  res/res_pjsip_outbound_registration.c, res/res_pjsip_mwi.c,
	  res/res_pjsip/pjsip_configuration.c, include/asterisk/sorcery.h,
	  include/asterisk/strings.h,
	  res/res_pjsip/include/res_pjsip_private.h,
	  res/res_pjsip_pubsub.c, res/res_pjsip/config_transport.c:
	  res_pjsip: AMI commands and events. Created the following AMI
	  commands and corresponding events for res_pjsip:
	  PJSIPShowEndpoints - Provides a listing of all pjsip endpoints
	  and a few select attributes on each. Events: EndpointList - for
	  each endpoint a few attributes. EndpointlistComplete - after all
	  endpoints have been listed. PJSIPShowEndpoint - Provides a detail
	  list of attributes for a specified endpoint. Events:
	  EndpointDetail - attributes on an endpoint. AorDetail - raised
	  for each AOR on an endpoint. AuthDetail - raised for each
	  associated inbound and outbound auth TransportDetail - transport
	  attributes. IdentifyDetail - attributes for the identify object
	  associated with the endpoint. EndpointDetailComplete - last event
	  raised after all detail events. PJSIPShowRegistrationsInbound -
	  Provides a detail listing of all inbound registrations. Events:
	  InboundRegistrationDetail - inbound registration attributes for
	  each registration. InboundRegistrationDetailComplete - raised
	  after all detail records have been listed.
	  PJSIPShowRegistrationsOutbound - Provides a detail listing of all
	  outbound registrations. Events: OutboundRegistrationDetail -
	  outbound registration attributes for each registration.
	  OutboundRegistrationDetailComplete - raised after all detail
	  records have been listed. PJSIPShowSubscriptionsInbound - A
	  detail listing of all inbound subscriptions and their attributes.
	  Events: SubscriptionDetail - on each subscription detailed
	  attributes SubscriptionDetailComplete - raised after all detail
	  records have been listed. PJSIPShowSubscriptionsOutbound - A
	  detail listing of all outboundbound subscriptions and their
	  attributes. Events: SubscriptionDetail - on each subscription
	  detailed attributes SubscriptionDetailComplete - raised after all
	  detail records have been listed. (issue ASTERISK-22609) Reported
	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/
	  ........ Merged revisions 403131 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-23 12:52 +0000 [r403118-403120]  Joshua Colp <jcolp@digium.com>

	* res/res_stasis_playback.c, rest-api/api-docs/events.json, /,
	  res/res_stasis_recording.c, res/ari/ari_model_validators.c,
	  rest-api/api-docs/recordings.json,
	  res/ari/ari_model_validators.h: ari: Add events for playback and
	  recording. While there were events defined for playback and
	  recording these were not actually sent. This change implements
	  the to_json handlers which produces them. (closes issue
	  ASTERISK-22710) Reported by: Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/3026/ ........ Merged
	  revisions 403119 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis_snoop.exports.in (added), /,
	  include/asterisk/stasis_app_snoop.h (added),
	  rest-api/api-docs/channels.json, res/res_stasis_snoop.c (added),
	  main/audiohook.c, res/ari/resource_channels.c,
	  res/res_ari_channels.c, res/ari/resource_channels.h: ari: Add
	  Snoop operation for spying/whispering on channels. The Snoop
	  operation can be invoked on a channel to spy or whisper on it. It
	  returns a channel that any channel operations can then be invoked
	  on (such as record to do monitoring). (closes issue
	  ASTERISK-22780) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/3003/ ........ Merged
	  revisions 403117 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-23 00:22 +0000 [r403106]  Rusty Newton <rnewton@digium.com>

	* apps/app_voicemail.c: app_voicemail: when forwarding a message,
	  play vm-msgforwarded instead of vm-msgsaved In the last release
	  of sounds, 1.4.25 we added a vm-msgforwarded prompt for various
	  core languages. Now we use that prompt. (issue ASTERISK-21413)
	  (closes issue ASTERISK-21413) Reported by: netwrkr Tested by:
	  newtonr

2013-11-22 23:57 +0000 [r403095]  Kinsey Moore <kmoore@digium.com>

	* tests/test_stasis.c, /, tests/test_stasis_channels.c: Make sure
	  unit tests compile This fixes the unit tests that were broken by
	  r403069 and several functions requiring a new parameter for
	  sanitization of JSON messages generated from object snapshots.
	  ........ Merged revisions 403094 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-22 22:37 +0000 [r403083]  Kevin Harwell <kharwell@digium.com>

	* /,
	  contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py,
	  res/res_pjsip/pjsip_configuration.c: res_pjsip: convert
	  configuration settings names to snake case some more Updated the
	  alembic script for pjsip. Also, the dtls config parsing stuff was
	  expecting strings with no underscores, so removed the underscores
	  from the option name before passing it to the parser. ........
	  Merged revisions 403082 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-22 20:10 +0000 [r403070]  Kinsey Moore <kmoore@digium.com>

	* res/res_stasis.c, main/stasis_endpoints.c,
	  res/ari/resource_endpoints.c, main/rtp_engine.c, /,
	  res/stasis/app.c, include/asterisk/stasis_bridges.h,
	  include/asterisk/stasis.h, include/asterisk/stasis_app.h,
	  main/stasis_bridges.c, res/ari/resource_bridges.c, main/json.c,
	  main/stasis_message.c, include/asterisk/stasis_channels.h,
	  main/stasis_channels.c, res/ari/resource_channels.c,
	  include/asterisk/stasis_endpoints.h: ARI: Don't leak
	  implementation details This change prevents channels used as
	  implementation details from leaking out to ARI. It does this by
	  preventing creation of JSON blobs of channel snapshots created
	  from those channels and sanitizing JSON blobs of bridge snapshots
	  as they are created. This introduces a framework for excluding
	  information from output targeted at Stasis applications on a
	  consumer-by-consumer basis using channel sanitization callbacks
	  which could be extended to bridges or endpoints if necessary.
	  This prevents unhelpful error messages from being generated by
	  ast_json_pack. This also corrects a bug where BridgeCreated
	  events would not be created. (closes issue ASTERISK-22744)
	  Review: https://reviewboard.asterisk.org/r/2987/ Reported by:
	  David M. Lee ........ Merged revisions 403069 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-22 17:27 +0000 [r403051]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_acl.c, res/res_pjsip.c,
	  res/res_pjsip/config_transport.c, res/res_pjsip/config_global.c,
	  /, configs/pjsip.conf.sample, res/res_pjsip/config_system.c,
	  contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
	  res/res_pjsip/pjsip_configuration.c: res_pjsip: convert
	  configuration settings names to snake case Renamed, where
	  appropriate, the configuration options for chan/res_pjsip to use
	  snake case (compound words separated by an underscore). For
	  example, faxdetect will become fax_detect, recordofffeature will
	  become record_off_feature, etc... Review:
	  https://reviewboard.asterisk.org/r/3002/ ........ Merged
	  revisions 403022 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-22 17:12 +0000 [r403017]  Joshua Colp <jcolp@digium.com>

	* /, main/translate.c: translate: Move freeing of frame to after it
	  is used. When translating from one format to another it is
	  possible to inform the translation function that the source frame
	  should be freed. This was previously done immediately but shortly
	  afterwards the frame that was freed was accessed and used again.
	  This change moves code around a bit so that the frame is now
	  freed after it has been completely used. (closes issue
	  ASTERISK-22788) Reported by: Corey Farrell Patches:
	  translate-access-after-free-11up.patch uploaded by coreyfarrell
	  (license 5909) translate-access-after-free-1.8.patch uploaded by
	  coreyfarrell (license 5909) ........ Merged revisions 403014 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 403015 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 403016 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-22 16:43 +0000 [r403013]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_directed_pickup.c, CHANGES: PickupChan: Add ability to
	  specify channel uniqueids as well as channel names. * Made
	  PickupChan() search by channel uniqueids if the search could not
	  find a channel by name. * Ensured PickupChan() never considers
	  the picking channel for pickup. * Made PickupChan() option p use
	  a common search by name routine. The original search was
	  erroneously case sensitive. (issue AFS-42) Review:
	  https://reviewboard.asterisk.org/r/3017/

2013-11-21 22:38 +0000 [r402995]  Jonathan Rose <jrose@digium.com>

	* CHANGES, apps/app_directory.c: app_directory: Set variable
	  indicating reason directory exited By the time the directory
	  application exits, a channel variable DIRECTORY_RESULT will be
	  set for the channel that invoked it which can be used to
	  determine the reason for exit. The changes log and the
	  app_directory documentation contain specific details about each
	  of the possible values for DIRECTORY_RESULT. Review:
	  https://reviewboard.asterisk.org/r/3016/

2013-11-21 22:36 +0000 [r402982-402994]  David M. Lee <dlee@digium.com>

	* rest-api-templates/ari_resource.c.mustache, /,
	  rest-api-templates/res_ari_resource.c.mustache: ari: Fix #include
	  to match generated headers for snakeCase resource files ........
	  Merged revisions 402993 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* rest-api-templates/make_ari_stubs.py, /: ari: Fix generators for
	  resources with camelCase names. For the new deviceState resource,
	  we need to properly generate device_state.[ch] files. ........
	  Merged revisions 402981 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-21 19:22 +0000 [r402969]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip_session.c, /: res_pjsip_session: Fix memory leak of
	  direct media format capabilities The direct media format
	  capabilities are always allocated in ast_sip_session_alloc and
	  were not freed in the session destructor. Whoops. (This being the
	  third whoops caught by Scott and Nitesh's valgrind work for the
	  Asterisk Test Suite. Nifty!) ........ Merged revisions 402968
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-21 19:09 +0000 [r402945-402957]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/app.h, /: voicemail: Fixup some doxygen
	  comments. ........ Merged revisions 402956 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/bucket.c: bucket: Fix scheme ref leak in
	  __ast_bucket_scheme_register(). ........ Merged revisions 402944
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-21 17:53 +0000 [r402942-402943]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix use of
	  uninitialized value in PJSIP In PJMEDIA,
	  pjmedia_sdp_rtpmap_to_attr will attempt to use the string
	  rtpmap.param regardless of its length value. Simply setting the
	  length to 0 does not prevent the garbage on the stack in
	  rtpmap.param.ptr from being formatted in a sprintf call. This
	  patch initializes the string to NULL so that at the very least,
	  something is provided to the function that is predictable.
	  ........ Merged revisions 402941 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_mwi.c: res_pjsip_mwi: Fix memory leak of MWI
	  subscriptions container This patch fixes a reference counting
	  memory leak on the ao2_container created as part of
	  create_mwi_subscriptions. When we create the container in this
	  routine, the intent is to hand lifetime ownership over to the
	  global container unsolicited_mwi. When
	  ao2_global_obj_replace_unref is called, the reference count on
	  mwi_subscriptions (the container) will be bumped by 1; however,
	  the function does not decrement the reference count on
	  mwi_subscriptions when this occurs. This will prevent the
	  container from being fully disposed of when Asterisk exits (or on
	  any subsequent call to this operation, such as during a reload).
	  ........ Merged revisions 402940 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-21 15:57 +0000 [r402928-402929]  David M. Lee <dlee@digium.com>

	* res/res_stasis.c, /: stasis: Fixed scoping problem with bridge
	  tracking. ........ Merged revisions 402817 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/ari/resource_channels.c, res/res_ari_channels.c,
	  res/ari/resource_channels.h, /, res/stasis/control.c,
	  include/asterisk/stasis_app.h, rest-api/api-docs/channels.json:
	  ari: Add silence generator controls This patch adds the ability
	  to start a silence generator on a channel via ARI. This generator
	  will play silence on the channel (avoiding audio timeouts on the
	  peer) until it is stopped, or some other media operation is
	  started (like playing media, starting music on hold, etc.).
	  (closes issue ASTERISK-22514) Review:
	  https://reviewboard.asterisk.org/r/3019/ ........ Merged
	  revisions 402926 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-19 23:17 +0000 [r402892]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_caller_id.c: res_pjsip_caller_id: Don't
	  overwrite user portion of the From header when fromuser is set.
	  The fromuser option is used to explicitly set the user within the
	  From header. The res_pjsip_caller_id module did not take this
	  setting into account when determining if the From header could be
	  modified or not. (closes issue ASTERISK-22866) Reported by:
	  Anthony Messina ........ Merged revisions 402891 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-16 13:51 +0000 [r402865]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip/pjsip_distributor.c, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: res_pjsip: Add
	  support for building against pjproject with SIP transaction group
	  lock support. SIP transaction group lock support has been
	  backported into our pjproject. Since the code now internally uses
	  a group lock the code is now changed to unlock it if present.
	  Note that the act of finding the transaction is what actually
	  returns it locked. For further information about group locks
	  check out the wiki page at:
	  http://trac.pjsip.org/repos/wiki/Group_Lock (issue
	  ASTERISK-22818) Reported by: Matt Jordan ........ Merged
	  revisions 402864 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-15 22:38 +0000 [r402854]  Jonathan Rose <jrose@digium.com>

	* apps/app_confbridge.c, CHANGES,
	  apps/confbridge/conf_config_parser.c,
	  configs/confbridge.conf.sample,
	  apps/confbridge/include/confbridge.h: Confbridge: Add option to
	  review the recording similar to announce_join_leave Review:
	  https://reviewboard.asterisk.org/r/3008/

2013-11-15 14:37 +0000 [r402839]  Kinsey Moore <kmoore@digium.com>

	* /, main/cel.c: CEL: Fix crash when using CELGenUserEvent This
	  fixes a crash when CELGenUserEvent is called from the dialplan
	  while CEL is disabled. Currently, CEL does not create its topics
	  and forwards if it is not enabled and external entities may
	  depend on these topics blindly since they should always be
	  available. This patch breaks up route creation and topic/forward
	  creation such that the CEL topics and forwards will always exist
	  while the router and its associated routes will be torn down and
	  recreated as necessary. (closes issue ASTERISK-22799) Review:
	  https://reviewboard.asterisk.org/r/3010/ Reported by: Matt Jordan
	  ........ Merged revisions 402838 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-14 21:36 +0000 [r402820-402829]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_directed_pickup.c: Pickup: Pickup() and PickupChan()
	  parameter parsing improvements. * Made Pickup() and PickupChan()
	  tollerate empty pickup values. i.e., You can now have
	  Pickup(&&exten@context). * Made PickupChan() use the standard
	  option flag parsing code.

	* apps/app_directed_pickup.c: Pickup: Ensure using PICKUPMARK never
	  considers the picking channel.

2013-11-14 20:32 +0000 [r402819]  Jonathan Rose <jrose@digium.com>

	* CHANGES, main/pbx.c, apps/app_sayunixtime.c: Say: If
	  SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF
	  Similar to how background works, if a say application is called
	  with this variable set to 'true', 'yes', 'on', etc. then using
	  DTMF while the say action is in progress will result in the
	  channel jumping to that extension in the dialplan. Review:
	  https://reviewboard.asterisk.org/r/3011/

2013-11-13 23:11 +0000 [r402805]  Joshua Colp <jcolp@digium.com>

	* rest-api/api-docs/channels.json, res/ari/resource_channels.c,
	  res/res_ari_channels.c, res/ari/resource_channels.h, /,
	  res/stasis/control.c, include/asterisk/stasis_app.h:
	  res_ari_channels: Add the ability to stop locally generated
	  ringing on a channel. Using the 'ring' operation it is possible
	  to start locally generated ringback if the channel is answered.
	  This change adds the ability to stop it by using DELETE. ........
	  Merged revisions 402804 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-12 23:17 +0000 [r402788-402795]  Kevin Harwell <kharwell@digium.com>

	* res/ari/resource_endpoints.c, /: ari endpoints: GET
	  /ari/endpoints/{invalid-tech} should return a 404 Was returning a
	  404 on a valid technology with an empty list of endpoints. Now
	  checking against the channel tech to make sure the tech itself is
	  valid and not just an empty list of endpoints. (issue
	  ASTERISK-22803) Reported by: David M. Lee ........ Merged
	  revisions 402793 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* rest-api/api-docs/endpoints.json, res/ari/resource_endpoints.c,
	  /, res/res_ari_endpoints.c: ari endpoints: GET
	  /ari/endpoints/{invalid-tech} should return a 404 Implementation
	  listing endpoints by technology returned an empty array if no
	  matching endpoints were found. Fixed so a "404 Not Found" will be
	  returned instead. (closes issue ASTERISK-22803) Reported by:
	  David M. Lee ........ Merged revisions 402787 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-12 19:38 +0000 [r402768-402778]  Mark Michelson <mmichelson@digium.com>

	* /, main/channel.c: Switch to a scoped lock to avoid missing
	  unlocks in failure returns. ........ Merged revisions 402769 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/channel.c, /: Move a NULL check to a place that makes more
	  sense. Two variables were being checked for NULLity immediately
	  after being declared NULL. I moved the NULL check until after the
	  variables are allocated. This allows for the "channelvars" option
	  in manager.conf to work as intended again. ........ Merged
	  revisions 402767 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-12 16:49 +0000 [r402758]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_messaging.c, res/res_pjsip_header_funcs.c, /:
	  pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer
	  dereferences Both res_pjsip_messaging and res_pjsip_header_funcs
	  were causing asterisk to crash because they were trying to
	  dereference a NULL pointer. In the case of res_pjsip_messaging it
	  was attempting to "print" a contact header that did not exist. In
	  fact contact headers should not be part of a SIP MESSAGE, so the
	  offending code was simply removed. In the case of
	  res_pjsip_header_funcs a null private channel tech was being
	  passed to the function and then later dereferenced. Added null
	  checks (and error logging) to the read/write function handlers to
	  guard against crashing. (closes issue ASTERISK-22821) Reported
	  by: Anthony Messina ........ Merged revisions 402757 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-12 16:34 +0000 [r402756]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_celgenuserevent.c: CELGenUserEvent: Fix error message
	  from ast_json_pack This prevents NULL from being passed into an
	  ast_json_pack call when no extra information is passed to the
	  application which prevents an error message about NULL arguments
	  from being generated. ........ Merged revisions 402755 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-12 15:27 +0000 [r402741]  David M. Lee <dlee@digium.com>

	* res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /:
	  Fixed a typ. ........ Merged revisions 402738 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-12 15:03 +0000 [r402711]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_dahdi.c, /: chan_dahdi: Fix crash during caller ID
	  read Asterisk will sometimes core dump during caller id read on
	  analog channels due to a negative return value from the read() in
	  my_get_callerid that slips through as a negative length argument
	  to callerid_feed() if the errno returned by DAHDI is ELAST. This
	  change ensures that the negative return is treated properly even
	  when it is ELAST. (closes issue ASTERISK-22746) Reported by:
	  Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch
	  uploaded by Michael Walton (License 6502) ........ Merged
	  revisions 402708 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 402709 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402710 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-11 20:28 +0000 [r402698]  Jonathan Rose <jrose@digium.com>

	* apps/app_confbridge.c: Confbridge: add test events for dynamic
	  menus test Adds a couple of test events for conference menu
	  actions so that it's easy to discern when those menu actions have
	  been triggered. (issue ASTERISK-22760) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/2999/

2013-11-11 19:31 +0000 [r402688]  Mark Michelson <mmichelson@digium.com>

	* apps/app_confbridge.c, /: Get rid of some inaccurate comments.
	  I'm doing some unrelated work in app_confbridge and finding these
	  "invalid pin" comments to be annoying. Get out! ........ Merged
	  revisions 402686 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402687 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-11 15:37 +0000 [r402648]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_queue.c: app_queue: Honor penalty limits of 0 In the
	  current app_queue code from 1.8 up to trunk the upper and lower
	  penalties can be set to 0 but the value is interpreted to be
	  disabled instead of actually setting limits. This is especially
	  evident if min and max limits are set to 0 and members with
	  penalties of 0 and 1 are in the queue since the member with
	  penalty 1 will still receive calls. This patch adjusts the
	  special disabled value to be INT_MAX instead of 0. (closes issue
	  ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/
	  Reported by: Schmooze Com ........ Merged revisions 402645 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 402646 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402647 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 23:07 +0000 [r402607]  Scott Griepentrog <sgriepentrog@digium.com>

	* /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip:
	  keep same local (from) tag for outgoing register requests For
	  outbound register requests the tag on the From line was updated
	  every 20 seconds prior to a successful registration and also once
	  for each registration renewal. That behavior can possibly cause
	  the registration to be denied because of the different tag, and
	  is not aligned with the intention of RFC 3261 8.1.3.5 "...
	  request constitutes a new transaction and SHOULD have the same
	  value of the Call-ID, To, and From of the previous request...".
	  This updates chan_sip to have a field to keep the local tag in
	  the registration structure and use that tag for registration
	  requests where the callid is also unchanged. (closes issue
	  ASTERISK-12117) Reported by: Pawel Pierscionek Review:
	  https://reviewboard.asterisk.org/r/2988/ ........ Merged
	  revisions 402604 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 402605 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402606 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 20:37 +0000 [r402595]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_stasis.c: res_stasis.c: Fix locking issues with the
	  app_bridge_moh container. * Fix unlinking from the
	  app_bridges_moh container in remove_bridge_moh() without a lock
	  under normal circumstances. * Made check
	  ast_bridge_set_after_callback() return value in
	  bridge_moh_create() to handle failure. * Fixed SCOPED_AO2LOCK()
	  locking over too much scope in stasis_app_bridge_moh_channel()
	  and stasis_app_bridge_moh_stop(). * Fixed unusual usage of
	  ao2_unlink_flag() in control_unlink(). * Fixed orphaned bridge
	  from off nominal path in stasis_app_bridge_create(). * Fixed
	  strange construct in stasis_app_unsubscribe(). From a bad merge?
	  * Made load_module() cleanup on failure. Review:
	  https://reviewboard.asterisk.org/r/2962/ ........ Merged
	  revisions 402593 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 19:33 +0000 [r402585]  Jonathan Rose <jrose@digium.com>

	* /, main/security_events.c, configs/manager.conf.sample, CHANGES,
	  include/asterisk/manager.h, main/manager.c: security_events: Push
	  out security events over AMI events Security Events will now be
	  written to any listener of the new 'security' class Review:
	  https://reviewboard.asterisk.org/r/2998/ ........ Merged
	  revisions 402584 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 19:22 +0000 [r402583]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip.c, /: Clarify an ambiguous error message. ........
	  Merged revisions 402582 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 18:53 +0000 [r402571-402572]  David M. Lee <dlee@digium.com>

	* /, res/res_pjsip/config_system.c: res_pjsip: Print a helpful
	  error message if sorcery registration fails ........ Merged
	  revisions 402570 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/ari/resource_playbacks.h, /: Changes from make ari-stubs
	  after r402560 ........ Merged revisions 402561 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 17:59 +0000 [r402562]  Kevin Harwell <kharwell@digium.com>

	* rest-api/resources.json, res/ari/resource_playback.h (removed),
	  res/res_ari_playbacks.c (added), res/ari/resource_playbacks.h
	  (added), /, res/ari.make, rest-api/api-docs/playback.json
	  (removed), res/ari/resource_playback.c (removed),
	  res/res_ari_playback.c (removed),
	  rest-api/api-docs/playbacks.json (added),
	  res/ari/resource_playbacks.c (added): ARI playback: Rename ARI
	  Playback to Playbacks Before playback was the only non plural
	  resource. It has been renamed to playbacks for consistency.
	  (closes issue ASTERISK-22737) Reported by: Paul Belanger ........
	  Merged revisions 402560 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 17:29 +0000 [r402557]  David M. Lee <dlee@digium.com>

	* res/res_ari.c, main/manager.c, /, main/http.c: ari: Add
	  application/x-www-form-urlencoded parameter support ARI POST
	  calls only accept parameters via the URL's query string. While
	  this works, it's atypical for HTTP API's in general, and
	  specifically frowned upon with RESTful API's. This patch adds
	  parsing for application/x-www-form-urlencoded request bodies if
	  they are sent in with the request. Any variables parsed this way
	  are prepended to the variable list supplied by the query string.
	  (closes issue ASTERISK-22743) Review:
	  https://reviewboard.asterisk.org/r/2986/ ........ Merged
	  revisions 402555 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-08 14:58 +0000 [r402546]  Kevin Harwell <kharwell@digium.com>

	* apps/app_dahdiras.c, utils/extconf.c, main/asterisk.c:
	  app_dahdiras: Use waitpid instead of wait4. Several places in the
	  code were using wait4 while other places were using waitpid. This
	  change makes all places use waitpid in order to make things more
	  consistent and since the 'rusage' object passed in/out of wait4
	  was never used. (closes issue ASTERISK-22557) Reported by:
	  YvesGael Patches: asterisk-11.5.1-wait4.patch uploaded by hurdman
	  (license 6537)

2013-11-07 23:42 +0000 [r402538]  Jonathan Rose <jrose@digium.com>

	* res/res_pjsip_authenticator_digest.c, /: PJSIP: Improve error
	  handling in digest authenticator Previously, regardless of
	  whether failure to authenticate was due to lacking any
	  authentication or actually failing authentication, the Digest
	  Authenticator would simply return that a challenge was still
	  needed. It will continue to do that when no authentication
	  information is in the received SIP digest, but when
	  authentication information is present and does not pass
	  authentication, that will be treated as an authentication error.
	  This is to ensure that PJSIP will issue security events indicated
	  failed auths. ........ Merged revisions 402537 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-07 21:10 +0000 [r402529]  David M. Lee <dlee@digium.com>

	* res/ari/resource_applications.c, res/ari/resource_playback.c,
	  rest-api/api-docs/channels.json, res/ari/resource_applications.h,
	  res/ari/resource_channels.c, res/ari/resource_playback.h,
	  rest-api/api-docs/recordings.json, res/ari/resource_recordings.c,
	  rest-api-templates/ari_resource.c.mustache,
	  rest-api-templates/asterisk_processor.py,
	  res/ari/resource_channels.h, rest-api/api-docs/endpoints.json,
	  res/ari/resource_endpoints.c, res/ari/resource_recordings.h,
	  res/ari/resource_events.c, res/res_ari_playback.c,
	  res/res_ari_applications.c, res/ari/resource_endpoints.h,
	  res/ari/resource_events.h, rest-api/api-docs/sounds.json,
	  res/ari/resource_sounds.c, res/res_ari_channels.c,
	  rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
	  res/ari/resource_sounds.h, res/res_ari_recordings.c,
	  res/ari/resource_bridges.h, rest-api/api-docs/asterisk.json,
	  res/ari/resource_asterisk.c, res/res_ari_endpoints.c,
	  rest-api/api-docs/applications.json,
	  rest-api/api-docs/playback.json, res/res_ari_events.c,
	  res/ari/resource_asterisk.h, rest-api-templates/swagger_model.py,
	  res/res_ari_sounds.c, res/res_ari_bridges.c, /,
	  rest-api-templates/ari_resource.h.mustache,
	  rest-api-templates/rest_handler.mustache, res/res_ari_asterisk.c,
	  rest-api-templates/res_ari_resource.c.mustache: ari: User better
	  nicknames for ARI operations While working on building client
	  libraries from the Swagger API, I noticed a problem with the
	  nicknames. channel.deleteChannel() channel.answerChannel()
	  channel.muteChannel() Etc. We put the object name in the nickname
	  (since we were generating C code), but it makes OO generators
	  redundant. This patch makes the nicknames more OO friendly. This
	  resulted in a lot of name changing within the res_ari_*.so
	  modules, but not much else. There were a couple of other fixed I
	  made in the process. * When reversible operations (POST /hold,
	  POST /unhold) were made more RESTful (POST /hold, DELETE
	  /unhold), the path for the second operation was left in the API
	  declaration. This worked, but really the two operations should
	  have been on the same API. * The POST /unmute operation had still
	  not been REST-ified. Review:
	  https://reviewboard.asterisk.org/r/2940/ ........ Merged
	  revisions 402528 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-06 21:58 +0000 [r402518]  Kevin Harwell <kharwell@digium.com>

	* /, apps/app_queue.c: app_queue: crash if first agent is "busy" If
	  the first agent/member (via CLI "queue show") in a queue is
	  "busy" (dnd, circuit busy, etc...) and no agents answered then
	  app_queue would crash. This occurred because while the calling of
	  agent(s) remained valid the channel on "busy" agent would be set
	  to NULL and then later dereferenced upon a second "rna" function
	  call. The original intention of the code is to have only valid
	  "call attempt" objects (channels != NULL) checked while
	  attempting to call agent(s). It does this by building a
	  "call_next" list of valid "call attempt" objects. In the case of
	  the "busy" agent subsequent builds of the valid "call attempt"
	  list would sometimes include (the case mentioned above) an
	  invalid "call attempt" object. The fix was to make sure the "call
	  attempt" list was appropriately built on every iteration. A NULL
	  sanity check was also added at the original offending spot of the
	  crash just in case another one slipped by somehow. (closes issue
	  ASTERISK-22644) Reported by: Marco Signorini Review:
	  https://reviewboard.asterisk.org/r/2983/ ........ Merged
	  revisions 402517 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-05 21:17 +0000 [r402502-402508]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: chan_sip: Use AST_AF* defined constant
	  when calling ast_get_ip While the structure passed to ast_get_ip
	  should be set memset to 0, thus initializing the ss_family member
	  to 0, explicitly setting it to AST_AF_UNSPEC is more portable.
	  ........ Merged revisions 402507 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_iax2.c, /: chan_iax2: Fix incorrect usage of
	  ast_get_ip involving uninitialized struct This started off as a
	  fix for the failing IAX2 acl_call test in the Asterisk Test
	  Suite. When inspecting why that test was failing, it became clear
	  that all attempts to bind to any local loopback address was
	  failing: [Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding
	  IAX2 to address 127.0.0.1:4569 [Nov 2 15:56:28] DEBUG[15787]
	  netsock2.c: Splitting '127.0.0.1' into... [Nov 2 15:56:28]
	  DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''. [Nov 2
	  15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1",
	  "(null)", ...): ai_family not supported [Nov 2 15:56:28]
	  WARNING[15787] acl.c: Unable to lookup '127.0.0.1' While there's
	  conceivably other ways for getaddrino to return EAI_FAMILY, the
	  most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not
	  provided as the desired family. The culprit was the call to
	  ast_get_ip, defined in acl.h. This function uses the family from
	  the passed in addr object (which it will also populate when it
	  returns!) when it eventually calls getaddrinfo. This patch fixes
	  the use of ast_get_ip that were not specifying the family in
	  chan_iax2. This prevents uninitialized use of the structure, so
	  that the addresses resolve correctly. Review:
	  https://reviewboard.asterisk.org/r/2991 ........ Merged revisions
	  402505 from http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/acl.h, /, include/asterisk/netsock2.h: netsock2:
	  Define AST_AF_* enum constants to their AF_* equivalents This
	  patch explicitly defines AST_AF_* enum constants to their
	  sys/socket.h defined equivalents. It is certainly unclear why
	  these constants actually have to exist, given that netsock2.h
	  includes sys/socket.h; however, since the code base is already
	  liberally sprinkled with the usage of AST_AF_* (as well as with
	  direct calls to AF_*), this will at least keep the semantics
	  consistent between their usage across systems. ........ Merged
	  revisions 402503 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis_channels.c, /: stasis_channels: Don't give preference
	  to ANI info in channel snapshots When publishing channel
	  snapshots, we currently compute the caller ID name and number by
	  giving preference first to ani.{name|number}, then to
	  id.{name|number}. However, when a channel driver (such as
	  chan_sip) updates the caller ID, it typically only updates the
	  caller ID stored in id.{name|number}. This means that we are
	  currently giving preference to stale information. When looking at
	  the rest of the code base, the only other place where we appear
	  to use this same logic is in app_amd. Everywhere else, we treat
	  the party information in ani as being separate to the party
	  information in id. This patch publishes only the caller ID name
	  and number in the snapshot field for caller_name and caller_num.
	  Note that the information in ANI is still available in
	  caller_ani. Review: https://reviewboard.asterisk.org/r/2992/
	  ........ Merged revisions 402501 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-04 21:02 +0000 [r402453]  Kevin Harwell <kharwell@digium.com>

	* /, channels/chan_sip.c: chan_sip: notify dialog info ignores
	  presentation indicator in callerid The presentation indicator in
	  a callerid (e.g. set by dialplan function
	  Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
	  Info Notifies are generated during extension monitoring. Added a
	  check to make sure the name and/or number presentations on the
	  callee (remote identity) are set to allow. If they are restricted
	  then "anonymous" is used instead. (closes issue AST-1175)
	  Reported by: Thomas Arimont Review:
	  https://reviewboard.asterisk.org/r/2976/ ........ Merged
	  revisions 402450 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402452 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-02 04:30 +0000 [r402406-402439]  Richard Mudgett <rmudgett@digium.com>

	* main/stasis.c, main/stasis_message_router.c, /,
	  include/asterisk/vector.h: vector: Uppercase API to follow C
	  convention. C does not support templates like C++. ........
	  Merged revisions 402438 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/lock.h, main/stasis.c,
	  main/stasis_message_router.c, /, include/asterisk/vector.h:
	  vector: Update API to be more flexible. Made the vector macro API
	  be more like linked lists. 1) Added a name parameter to
	  ast_vector() to name the vector struct. 2) Made the API take a
	  pointer to the vector struct instead of the struct itself. 3)
	  Added an element cleanup macro/function parameter when removing
	  an element from the vector for ast_vector_remove_cmp_unordered()
	  and ast_vector_remove_elem_unordered(). 4) Added
	  ast_vector_get_addr() in case the vector element is not a simple
	  pointer. * Converted an inline vector usage in
	  stasis_message_router to use the vector API. It needed the API
	  improvements so it could be converted. * Fixed topic reference
	  leak in router_dtor() when the stasis_message_router is
	  destroyed. * Fixed deadlock potential in stasis_forward_all() and
	  stasis_forward_cancel(). Locking two topics at the same time
	  requires deadlock avoidance. * Made internal_stasis_subscribe()
	  tolerant of a NULL topic. * Made stasis_message_router_add(),
	  stasis_message_router_add_cache_update(),
	  stasis_message_router_remove(), and
	  stasis_message_router_remove_cache_update() tolerant of a NULL
	  message_type. * Promoted a LOG_DEBUG message to LOG_ERROR as
	  intended in dispatch_message(). Review:
	  https://reviewboard.asterisk.org/r/2903/ ........ Merged
	  revisions 402429 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/confbridge/conf_state_single.c,
	  apps/confbridge/conf_state_inactive.c,
	  apps/confbridge/conf_state_single_marked.c, /,
	  apps/confbridge/include/confbridge.h,
	  apps/confbridge/conf_state_multi.c, apps/app_confbridge.c,
	  apps/confbridge/conf_state_multi_marked.c,
	  apps/confbridge/conf_state.c: confbridge: Separate user muting
	  from system muting overrides. The system overrides the user
	  muting requests when MOH is playing or a waitmarked user is
	  waiting for a marked user to join. System muting overrides
	  interfere with what the user may wish the muting to be when the
	  system override ends. * User muting requests are now independent
	  of the system muting overrides. The effective muting is now the
	  logical or of the user request and system override. * Added a
	  Muted flag to the CLI "confbridge list <conference>" command. *
	  Added a Muted header to the AMI ConfbridgeList action
	  ConfbridgeList event. (closes issue AST-1102) Reported by: John
	  Bigelow Review: https://reviewboard.asterisk.org/r/2960/ ........
	  Merged revisions 402425 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402427 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/config.c, apps/confbridge/conf_config_parser.c,
	  configs/confbridge.conf.sample, /: config: Allow ConfBridge DTMF
	  menus to have '#' as the first digit. ConfBridge allows custom
	  DTMF menus to be created in the confbridge.conf file by assigning
	  a DTMF key sequence to a sequence of actions as follows:
	  DTMF-sequence = action,action... Unfortunately, the normal config
	  file processing code interprets an initial '#' character as
	  starting a directive such as #include. * Add the ability to
	  escape the first non-blank character in a config line so the '#'
	  character can be used without triggering the directive processing
	  code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported
	  by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch
	  (license #5621) patch uploaded by rmudgett (modified) Review:
	  https://reviewboard.asterisk.org/r/2969/ ........ Merged
	  revisions 402407 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402416 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/app.h, /, main/app.c: voicemail: Simplify
	  callback pointer declarations and add doxygen. * Typedefed and
	  added doxegen for the voicemail callback functions. * Simplified
	  the prototypes for ast_install_vm_functions() and
	  ast_install_vm_test_functions() to use the new function typedefs.
	  * Simplified the voicemail callback function pointer variable
	  declarations to use the new function typedefs. ........ Merged
	  revisions 402398 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-01 22:48 +0000 [r402397]  Jonathan Rose <jrose@digium.com>

	* apps/confbridge/conf_config_parser.c,
	  apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
	  CHANGES: app_confbridge: Make the CONFBRIDGE function be able to
	  create dynamic menus Also adds the ability to clear all profile
	  items and makes behavior more consistent with documentation as
	  when choosing whether to use CONFBRIDGE datastore profiles or the
	  application arguments to the confbridge application. (closes
	  issue ASTERISK-22760) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2971/

2013-11-01 21:51 +0000 [r402388]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/manager_bridges.c, /, main/bridge.c,
	  include/asterisk/bridge.h: Manager: Add equivalent AMI actions
	  for the bridge CLI commands. Adds the following AMI events,
	  closely following their CLI counterparts: BridgeDestroy
	  BridgeKick BridgeTechnologyList BridgeTechnologySuspend
	  BridgeTechnologyUnsuspend BridgeDestroy kicks an entire bridge,
	  where BridgeKick kicks just one channel off the bridge. When
	  kicking a channel, specifying the bridge also (optional) insures
	  it is not removed from the wrong bridge. The BridgeTechnology
	  events allow viewing and changing suspension status, which
	  affects only subsequent not active bridging. (closes
	  ASTERISK-22356) Reported by: Richard Mudgett Review:
	  https://reviewboard.asterisk.org/r/2973/ ........ Merged
	  revisions 402387 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-01 16:31 +0000 [r402368]  David M. Lee <dlee@digium.com>

	* /, rest-api-templates/api.wiki.mustache: ari wiki docs: add notes
	  about allowMultiple parameters. This patch adds a note to any
	  parameter that has 'allowMultiple' set in the Swagger
	  documentation. (closes issue ASTERISK-22704) ........ Merged
	  revisions 402367 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-01 14:38 +0000 [r402359]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/stasis_app.h, rest-api/api-docs/channels.json,
	  res/ari/resource_channels.c, res/res_ari_channels.c,
	  res/ari/resource_channels.h, res/res_stasis_playback.c, /,
	  res/stasis/control.c: res_ari_channels: Add ring operation, dtmf
	  operation, hangup reasons, and tweak early media. The ring
	  operation sends ringing to the specified channel it is invoked
	  on. The dtmf operation can be used to send DTMF digits to the
	  specified channel of a specific length with a wait time in
	  between. Finally hangup reasons allow you to specify why a
	  channel is being hung up (busy, congestion). Early media behavior
	  has also been tweaked slightly. When playing media to a channel
	  it will no longer automatically answer. If it has not been
	  answered a progress indication is sent instead. (closes issue
	  ASTERISK-22701) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2916/ ........ Merged
	  revisions 402358 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-01 12:40 +0000 [r402349]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_asterisk.c, /, channels/chan_sip.c,
	  include/asterisk/rtp_engine.h: chan_sip: Fix RTCP port for SRFLX
	  ICE candidates This corrects one-way audio between Asterisk and
	  Chrome/jssip as a result of Asterisk inserting the incorrect RTCP
	  port into RTCP SRFLX ICE candidates. This also exposes an ICE
	  component enumeration to extract further details from candidates.
	  (closes issue ASTERISK-21383) Reported by: Shaun Clark Review:
	  https://reviewboard.asterisk.org/r/2967/ ........ Merged
	  revisions 402345 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402348 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-11-01 12:33 +0000 [r402337-402347]  Joshua Colp <jcolp@digium.com>

	* /, include/asterisk/stasis_app.h, res/ari/resource_channels.c:
	  res_ari_channels: Fix a deadlock when originating multiple
	  channels close to eachother. If a Stasis application is specified
	  an implicit subscription is done on the originated channel. This
	  was previously done with the channel lock held which is dangerous
	  as the underlying code locks the container and iterates items.
	  This change releases the lock on the originated channel before
	  subscribing occurs. (closes issue ASTERISK-22768) Reported by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/2979/
	  ........ Merged revisions 402346 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/stasis/control.c: res_stasis: Ensure the channel is always
	  departed from the bridge when it leaves. This change adds a
	  command to the command queue to explicitly depart the channel
	  from the bridge when it is told it has left. If the channel has
	  already been departed or has entered a different bridge this
	  command will become a no-op. (closes issue ASTERISK-22703)
	  Reported by: John Bigelow (closes issue ASTERISK-22634) Reported
	  by: Kevin Harwell Review:
	  https://reviewboard.asterisk.org/r/2965/ ........ Merged
	  revisions 402336 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-31 22:09 +0000 [r402328]  Mark Michelson <mmichelson@digium.com>

	* /, contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
	  contrib/scripts/sip_to_res_sip (removed),
	  contrib/scripts/sip_to_pjsip (added),
	  contrib/scripts/sip_to_pjsip/astconfigparser.py,
	  contrib/scripts/sip_to_pjsip/astdicts.py: Update the conversion
	  script from sip.conf to pjsip.conf (closes issue ASTERISK-22374)
	  Reported by Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2846 ........ Merged revisions
	  402327 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-31 16:06 +0000 [r402286-402290]  Matthew Jordan <mjordan@digium.com>

	* main/loader.c, /: core/loader: Don't call dlclose in a while loop
	  For awhile now, we've noticed continuous integration builds
	  hanging on CentOS 6 64-bit build agents. After resolving a number
	  of problems with symbols, strange locks, and other shenanigans,
	  the problem has persisted. In all cases, gdb shows the Asterisk
	  process stuck in loader.c on one of the infinite while loops that
	  calls dlclose repeatedly until success. The documentation of
	  dlclose states that it returns 0 on success; any other value on
	  error. It does not state that repeatedly calling it will
	  eventually clear those errors. Most likely, the repeated calls to
	  dlclose was to force a close by exhausting the references on the
	  library; however, that will never succeed if: (a) There is some
	  fundamental error at work in the loaded library that precludes
	  unloading it (b) Some other loaded module is referencing a symbol
	  in the currently loaded module This results in Asterisk sitting
	  forever. Since we have matching pairs of dlopen/dlclose, this
	  patch opts to only call dlclose once, and log out as an ERROR if
	  dlclose fails to return success. If nothing else, this might help
	  to determine why on the CentOS 6 64-bit build agent things are
	  not closing successfully. Review:
	  https://reviewboard.asterisk.org/r/2970 ........ Merged revisions
	  402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 402288 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402289 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/media_index.c, /: medix_index: Display errors when library
	  calls fail Based on feedback from ipengineer in #asterisk, when
	  the media indexer cannot access a sound file on the system (or
	  otherwise fails) Asterisk displays a "Cannot frob file" error but
	  fails to tell you why. This is especially problematic as the
	  media_indexer failing will rpevent Asterisk from starting, as it
	  is in the core. We now display the errno error messages so folks
	  can figure out what they've done wrong. ........ Merged revisions
	  402285 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-31 14:45 +0000 [r402277]  David M. Lee <dlee@digium.com>

	* /, res/stasis/app.c: stasis: add functions embarrassingly missing
	  from r400522 I neglected to implement two of the endpoint
	  subscription functions when I did the work. Normally, you'll only
	  hit that when you unsubscribe from a specific endpoint. ........
	  Merged revisions 402276 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-30 17:54 +0000 [r402266]  Kevin Harwell <kharwell@digium.com>

	* channels/chan_pjsip.c, /, res/res_pjsip_messaging.c:
	  pjsip_messaging: Added debug for in dialog messaging (issue
	  ASTERISK-22777) Reported by: Matt Jordan ........ Merged
	  revisions 402265 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-29 23:43 +0000 [r402227]  Rusty Newton <rnewton@digium.com>

	* /, sounds/Makefile: Updates for 1.4.25 core sounds and 1.4.14
	  extra sounds, plus new en_GB language set The new sound packages
	  relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413,
	  ASTERISK-20782 Modified sounds/Makefile for the new sound
	  versions and to account for the new en_GB language set. (issue
	  ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue
	  ASTERISK-22411) (closes issue ASTERISK-22544) ........ Merged
	  revisions 402224 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 402225 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402226 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-29 12:57 +0000 [r402155]  Matthew Jordan <mjordan@digium.com>

	* main/xmldoc.c, main/channel.c, main/pbx.c, /, main/translate.c:
	  Remove some spammy debug messages; improve clarity of others
	  Debug messages aren't free. Even when the debug level is
	  sufficiently low such that the messages are never evaluated,
	  there is a cost to having to parse Asterisk logs that contain
	  debug messages that (a) fail to convey sufficient information or
	  (b) occur so frequently as to be next to meaningless. Based on
	  having to stare at lots of DEBUG messages, this patch makes the
	  following changes: * channel.c: When copying variables from a
	  parent channel to a child channel, specify the channels involved.
	  Do not log anything for a variable that is not inherited; the
	  fact that it doesn't have an _ or __ already signifies that it
	  won't be inherited. * pbx.c: Specify what function evaluation has
	  occurred that created the result. * translate.c: Bump up the
	  translator path messages to 10. I've never once had to use these
	  debug messages, and for each format that is registered (on
	  startup) and unregistered (on shutdown) the entire f^2 matrix is
	  logged out. For short tests in the Asterisk Test Suite, this
	  should make finding the actual test much easier. * xmldoc.c: The
	  debug message that 'blah' is not found in the tree is expected.
	  Often, description elements - which are not required - are not
	  provided. This debug message adds no additional value, as it is
	  not indicative of an error or helpful in debugging which element
	  did not contain a 'blah' element as a child. If an element is
	  supposed to contain a child element, then that XML tree should
	  have failed validation in the first place. Review:
	  https://reviewboard.asterisk.org/r/2966/ ........ Merged
	  revisions 402150 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 402151 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402154 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-29 12:51 +0000 [r402149-402153]  Kinsey Moore <kmoore@digium.com>

	* rest-api/api-docs/channels.json, res/ari/resource_channels.c,
	  res/res_ari_channels.c, res/ari/resource_channels.h, /: ARI:
	  Remove channels/{channelId}/dial This removes the
	  /ari/channels/{channelId}/dial URI since it is redundant, overly
	  complex, is likely to become more externally complex over time,
	  and is too high-level compared with other ARI operations. See the
	  following for further information:
	  http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html
	  (closes issue ASTERISK-22784) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2968/ ........ Merged
	  revisions 402152 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* bridges/bridge_native_rtp.c, /: bridge_native_rtp: Ensure bridge
	  is torn down When a bridge transitions away from one tech to
	  another, the tech going away is provided a dummy bridge with no
	  channels in it to tear down. Currently this means that the
	  teardown code exits prematurely and does not tear anything down.
	  This change tears down RTP bridging for the channel provided in
	  the leave bridge tech callback. This also reverts the majority of
	  r400403 since it is now redundant. (closes issue ASTERISK-22628)
	  (closes issue ASTERISK-22676) Reported by: John Bigelow Reported
	  by: Kevin Harwell Tested by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/2905/ Patches:
	  native_rtp_fix.diff uploaded by Kinsey Moore (License 6273)
	  ........ Merged revisions 402148 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-29 11:15 +0000 [r402140]  Joshua Colp <jcolp@digium.com>

	* /, rest-api/api-docs/playback.json, res/res_ari_playback.c:
	  res_ari_playback: Add missing 404 error response for GET and
	  DELETE. (closes issue ASTERISK-22722) Reported by: Richard
	  Mudgett ........ Merged revisions 402139 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-28 22:10 +0000 [r402128-402130]  David M. Lee <dlee@digium.com>

	* /, doc: Ignore full docs ........ Merged revisions 402127 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /: Put back several merge revisions that were lost in r402054

	* /: Put back several merge revisions that were lost in r401962

2013-10-28 15:08 +0000 [r402113-402117]  Michael L. Young <elgueromexicano@gmail.com>

	* /, UPGRADE-11.txt, UPGRADE-12.txt: Fix UPGRADE.txt Due To Merging
	  From Branch 11 When merging in the patch for ASTERISK-22728, the
	  UPGRADE.txt file was changed incorrectly. That change should have
	  gone into ASTERISK-11.txt. This commit is to fix that. Also,
	  another comment in the UPGRADE-11.txt was missing and this commit
	  adds that as well. ........ Merged revisions 402115 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c, UPGRADE-12.txt: chan_sip: Clarify
	  'Forcerport' Setting Displayed When Running "sip show peers"
	  While looking at ASTERISK-22236, Walter Doekes pointed out that
	  when running "sip show peers", the setting being displayed can be
	  confusing. The display of "N" used to mean NAT (i.e. yes). The
	  NAT setting has gone through many different changes resulting in
	  the display of different characters to try and convey what the
	  current setting is for 'Forcerport' (A for Auto and Forcerport is
	  currently on, a for Auto but Forcerport is off, Y for yes, and N
	  for no). During the initial code review to try and clarify these
	  settings (especially since "N" no longer meant what it used to
	  mean in prior versions of Asterisk), Mark Michelson suggested
	  using the full space available to display the settings which
	  helped to make the settings very clear. That was a great
	  suggestion. Therefore, this patch does the following: * The
	  column for 'Forcerport' now will show: Auto (Yes), Auto (No),
	  Yes, or No. * A column for the 'Comedia' setting has been added.
	  It too will display the setting in a non-cryptic way: Auto (Yes),
	  Auto (No), Yes, or No. * UPGRADE.txt has been updated to document
	  this change. (closes issue ASTERISK-22728) Reported by: Walter
	  Doekes Tested by: Michael L. Young Patches:
	  asterisk-forcerport-display-clarification_v3.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2941 ........ Merged revisions
	  402111 from http://svn.asterisk.org/svn/asterisk/branches/11
	  ........ Merged revisions 402112 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-27 23:22 +0000 [r402073-402091]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /: Filter out internal channels from dial message
	  handling Surrogate channels would pop up from time to time in
	  dial message handling. This would cause a WARNING message to
	  appear, indicating that the Surrogate channel had no CDR. This
	  patch filters out those channels that have the internal
	  implementation flag set, such that the WARNING message isn't
	  displayed. ........ Merged revisions 402090 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* cdr/cdr_sqlite3_custom.c, /, cdr/cdr_syslog.c, cdr/cdr_sqlite.c,
	  cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
	  include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
	  cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
	  cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c: Prevent CDR backends
	  from unregistering while billing data is in flight This patch
	  makes it so that CDR backends cannot be unregistered while active
	  CDR records exist. This helps to prevent billing data from being
	  lost during restarts and shutdowns. Review:
	  https://reviewboard.asterisk.org/r/2880/ ........ Merged
	  revisions 402081 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, contrib/ast-db-manage/config/env.py,
	  contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py,
	  contrib/ast-db-manage/voicemail/env.py: Update Alembic database
	  scripts for external scripting and PostgreSQL, Oracle This patch
	  does the following: 1) The env scripts have been updated to be
	  tolerant of a NULL configuration file. This occurs when
	  configuration is provided by an external script, such that the
	  actual config.ini file is not used. 2) Enum types have all been
	  given names. This is needed for PostgreSQL script generation. 3)
	  The identifier meetme_confno_starttime_endtime is greater than 30
	  characters, and hence invalid for Oracle databases. This has been
	  truncated down to meetme_confno_start_end. ........ Merged
	  revisions 400383 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-26 12:56 +0000 [r402065]  Joshua Colp <jcolp@digium.com>

	* channels/chan_pjsip.c, include/asterisk/res_pjsip_session.h, /:
	  chan_pjsip: Fix a crash when direct media is enabled and an ACK
	  is received after the channel is hung up. (closes issue
	  ASTERISK-22731) Reported by: Kinsey Moore ........ Merged
	  revisions 402064 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-26 00:36 +0000 [r402056]  Richard Mudgett <rmudgett@digium.com>

	* res/res_stasis.c, /: res_stasis.c: Made use the ao2_container
	  callback templates. * Made res_stasis.c use the OBJ_SEARCH_XXX
	  defines. ........ Merged revisions 402055 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-26 00:27 +0000 [r402054]  Scott Griepentrog <sgriepentrog@digium.com>

	* main/rtp_engine.c, /, include/asterisk/rtp_engine.h: rtp_engine:
	  fix rtp payloads copy and improve argument names In function
	  ast_rtp_instance_early _bridge_make_compatible the use of
	  instance 0/1 as arguments doesn't clearly communicate a direction
	  that the copying of payloads from the source channel to the
	  destination channel will occur, making it more probable to have
	  the arguments to ast_rtp_codecs_payloads_copy() put in the
	  reverse order. This patch renames the arguments with _dst and
	  _src suffixes and corrects the copy direction. (closes issue
	  ASTERISK-21464) Reported by: Kevin Stewart Review:
	  https://reviewboard.asterisk.org/r/2894/ ........ Merged
	  revisions 402000 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 Test shows
	  rtpmap:119 being copied per this change, but is not in sip invite
	  ........ Merged revisions 402042 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 402043 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-25 23:58 +0000 [r402004-402045]  Richard Mudgett <rmudgett@digium.com>

	* /, main/taskprocessor.c: taskprocessor: Made use pthread_equal()
	  to compare thread ids. * Removed another silly use of RAII_VAR().
	  RAII_VAR() and SCOPED_LOCK() are not silver bullets that allow
	  you to turn off your brain. ........ Merged revisions 402044 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/stasis/app.c: You'd think that new files would be free of
	  whitespace issues. But you would be wrong. ........ Merged
	  revisions 402003 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-25 22:01 +0000 [r401999-402002]  Jonathan Rose <jrose@digium.com>

	* res/ari/resource_bridges.c, res/res_ari_bridges.c, /,
	  rest-api/api-docs/channels.json, res/ari/resource_channels.c,
	  res/res_ari_channels.c, rest-api/api-docs/bridges.json: ARI:
	  channel/bridge recording errors when invalid format specified
	  Asterisk will now issue 422 if recording is requested against
	  channels or bridges with an unknown format (closes issue
	  ASTERISK-22626) Reported by: Joshua Colp Review:
	  https://reviewboard.asterisk.org/r/2939/ ........ Merged
	  revisions 402001 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis_recording.c, rest-api/api-docs/channels.json,
	  res/ari/resource_channels.c, res/ari/ari_model_validators.c,
	  res/res_ari_channels.c, rest-api/api-docs/bridges.json,
	  rest-api/api-docs/recordings.json, res/ari/resource_bridges.c,
	  res/ari/ari_model_validators.h, res/res_ari_bridges.c,
	  rest-api/api-docs/events.json, /: ARI recordings: Issue HTTP
	  failures for recording requests with file conflicts If a file
	  already exists in the recordings directory with the same name as
	  what we would record, issue a 422 instead of relying on the
	  internal failure and issuing success. (closes issue
	  ASTERISK-22623) Reported by: Joshua Colp Review:
	  https://reviewboard.asterisk.org/r/2922/ ........ Merged
	  revisions 401973 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-25 20:51 +0000 [r401962]  Scott Griepentrog <sgriepentrog@digium.com>

	* include/asterisk/pbx.h, main/pbx.c, /: pbx.c: fix confused match
	  caller id that deleted exten still in hash This fixes a bug where
	  a zero length callerid match adjacent to a no match callerid
	  extension entry would be deleted together, which then resulted in
	  hashtable references to free'd memory. A third state of the
	  matchcid value has been added to indicate match to any extension
	  which allows enforcing comparison of matchcid on/off without
	  errors. (closes issue AST-1235) Reported by: Guenther Kelleter
	  Review: https://reviewboard.asterisk.org/r/2930/ ........ Merged
	  revisions 401959 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401960 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401961 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-25 17:41 +0000 [r401898-401939]  Jonathan Rose <jrose@digium.com>

	* /, res/res_pjsip/pjsip_distributor.c,
	  res/res_pjsip_endpoint_identifier_user.c: PJSIP: Add log messages
	  when requests are received for non-existent endpoints (closes
	  issue ASTERISK-22552) Reported by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/2934/ ........ Merged
	  revisions 401938 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* utils/clicompat.c, utils/refcounter.c, /: Put clicompat-r2.patch
	  back in We've figured out how to resolve the problems this was
	  causing in 12/trunk, so this can go back in now. (issue
	  ASTERISK-22467) Reported by: Corey Farrell Patches:
	  clicompat-r2.patch uploaded by coreyfarrell (license 5909)
	  ........ Merged revisions 401914 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401935 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401936 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, utils/clicompat.c: revert clicompat-r2.patch from r401704
	  Patch caused the following build errors against testsuite
	  https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244
	  (issue ASTERISK-22467) Reported by: Corey Farrell ........ Merged
	  revisions 401895 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401896 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401897 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-25 16:09 +0000 [r401886]  Kevin Harwell <kharwell@digium.com>

	* /, channels/chan_sip.c: chan_sip: Allow a sip peer to accept both
	  AVP and AVPF calls Adapts the behaviour of avpf to only impact
	  the format of outgoing calls. For inbound calls, both AVP and
	  AVPF calls will be accepted regardless of the value of avpf in
	  the configuration. (closes issue ASTERISK-22005) Reported by:
	  Torrey Searle Patches: optional_avpf_trunk.patch uploaded by
	  tsearle (license 5334) ........ Merged revisions 401884 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401885 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-25 13:49 +0000 [r401873]  David M. Lee <dlee@digium.com>

	* tests/test_json.c, /: test_json: Fix deprecation warnings After a
	  series of upgrades over recent weeks, I've discovered that
	  test_json.c won't compile in dev mode any more for me. One of
	  gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate
	  tempnam. Which, in general, is a good thing. But for test code
	  that just needs a temporary file, it's just annoying. This patch
	  replaces usage of tempname with mkstemp, avoiding the deprecation
	  warning. It also removes the temporary files when the test is
	  complete, which apparently we weren't doing before (oops).
	  Review: https://reviewboard.asterisk.org/r/2957/ ........ Merged
	  revisions 401872 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-24 21:06 +0000 [r401836]  Kevin Harwell <kharwell@digium.com>

	* /, main/logger.c: Logging: Logging types ignored after specifying
	  a verbose level If one specified a verbose level within a logging
	  facility in logger.conf then any component after it was ignored.
	  Fixed so all values are correctly read. (closes issue
	  ASTERISK-22456) Reported by: Kevin Harwell ........ Merged
	  revisions 401833 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401835 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-24 20:48 +0000 [r401834]  David M. Lee <dlee@digium.com>

	* rest-api-templates/models.wiki.mustache,
	  rest-api/api-docs/events.json, /,
	  rest-api-templates/swagger_model.py,
	  rest-api-templates/ari_model_validators.c.mustache: The Swagger
	  1.2 specification for type extension ended up being slightly
	  different than my proposal. Instead of putting an 'extends' field
	  on the subtype, the base type has a 'subTypes' field, which is a
	  list of the subTypes. Given that its a messaging model and not an
	  object model, kinda makes sense. This patch changes the
	  events.json api-doc, and the python translators to take the new
	  format into account. Other changes that are in Swagger 1.2 were
	  not adopted, since the spec is still in flux, and could change
	  before it's finalized. A summary of changes to the Swagger-1.2
	  spec can be found at
	  https://github.com/wordnik/swagger-core/wiki/1.2-transition.
	  (closes issue ASTERISK-22440) Review:
	  https://reviewboard.asterisk.org/r/2909/ ........ Merged
	  revisions 401701 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-24 20:34 +0000 [r401622-401832]  Jonathan Rose <jrose@digium.com>

	* /, main/utils.c: utils: Fix memory leaks and missed
	  unregistration of CLI commands on shutdown Final set of patches
	  in a series of memory leak/cleanup patches by Corey Farrell
	  (closes issue ASTERISK-22467) Reported by: Corey Farrell Patches:
	  main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
	  main-utils-11.patch uploaded by coreyfarrell (license 5909)
	  main-utils-12up.patch uploaded by coreyfarrell (license 5909)
	  ........ Merged revisions 401829 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401830 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401831 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, tests/test_linkedlists.c: test_linkedlists: Fix memory leak
	  (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
	  test_linkedlists-1.8.patch uploaded by coreyfarrell (license
	  5909) test_linkedlists-11up.patch uploaded by coreyfarrell
	  (license 5909) ........ Merged revisions 401790 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401791 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401792 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer
	  reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
	  jitterbuf-jb_reset-leak-1.8.patch
	  jitterbuf-jb_reset-leak-11up.patch ........ Merged revisions
	  401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 401787 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401788 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/astobj2.c, /: astobj2: Unregister debug CLI commands at exit
	  (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
	  astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell
	  (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by
	  coreyfarrell (license 5909) ........ Merged revisions 401781 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401783 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401784 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_voicemail.c, /: app_voicemail: Memory Leaks against
	  tests (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
	  app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
	  app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
	  ........ Merged revisions 401743 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401744 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401745 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/app.c, main/asterisk.c, utils/clicompat.c,
	  channels/chan_dahdi.c, codecs/ilbc/doCPLC.c, main/data.c, /:
	  memory leaks: Memory leak cleanup patch by Corey Farrell (second
	  set) Also covers ast_app_parse_timelen-fail-zero-length.patch,
	  but the patch was replaced with one of my own. (issue
	  ASTERISK-22467) Reported by: Corey Farrell Patches:
	  chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license
	  5909) clicompat-r2.patch uploaded by coreyfarrell (license 5909)
	  codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
	  data-cleanup-test-registration.patch uploaded by coreyfarrell
	  (license 5909) main-asterisk-kill-listener.patch uploaded by
	  coreyfarrell (license 5909) ........ Merged revisions 401704 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401705 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401706 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, tests/test_dlinklists.c, funcs/func_math.c,
	  channels/sip/reqresp_parser.c, main/test.c,
	  main/editline/readline.c: memory leaks: Memory leak cleanup patch
	  by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by:
	  Corey Farrell Patches:
	  chan_sip-parse_contact_header_test-free-contacts.patch uploaded
	  by coreyfarrell (license 5909) cli-filename-completion-leak.patch
	  uploaded by coreyfarrell (license 5909) func_math.patch uploaded
	  by corefarrell (license 5909) main-test-cleanup.patch uploaded by
	  coreyfarrell (license 5909) test_dlinklists.patch uploaded by
	  coreyfarrell (license 5909) ........ Merged revisions 401660 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401661 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401662 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk:
	  Address jittery DTMF events in RTP streams (closes issue
	  ASTERISK-21170) Reported by: NITESH BANSAL Patches:
	  dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
	  Review: https://reviewboard.asterisk.org/r/2938/ ........ Merged
	  revisions 401619 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401620 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401621 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-23 16:52 +0000 [r401582]  Richard Mudgett <rmudgett@digium.com>

	* /, cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a
	  filter when the CDR value is empty. Extra CDR records are written
	  if a filtered CDR value is empty because the filter is not
	  checked. (closes issue ASTERISK-22272) Reported by: Jordi Llull
	  Chavarria ........ Merged revisions 401577 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401579 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401581 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-23 16:48 +0000 [r401580]  John Bigelow <jbigelow@digium.com>

	* /, main/bridge_channel.c: Add a test suite event to indicate when
	  the atxfer 3-way feature is detected This adds a test suite event
	  that indicates to tests when the attended transfer three-way call
	  feature is detected. Review:
	  https://reviewboard.asterisk.org/r/2912/ ........ Merged
	  revisions 401578 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-23 15:23 +0000 [r401540]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_mgcp.c, /: chan_mgcp: Properly handle malformed
	  media lines This corrects a situation in which a media line was
	  not parsed properly and resulted in a crash. (closes issue
	  ASTERISK-21190) Reported by: adomjan Patches:
	  chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
	  ........ Merged revisions 401537 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401538 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401539 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-23 11:16 +0000 [r401500]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: chan_sip: Fix an issue where an
	  incompatible audio format may be added to SDP. If preferred
	  codecs included any non-audio format the code would mistakenly
	  add the audio format, even if it was not a joint capability with
	  the remote side. (closes issue ASTERISK-21131) Reported by:
	  nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by
	  nbougues (license 6470) ........ Merged revisions 401497 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401498 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401499 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-23 02:36 +0000 [r401489]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_iax2.c, configs/iax.conf.sample, /: chan_iax2: Fix
	  Binding To Multiple Addresses Again When reworking chan_iax2 for
	  IPv6, the ability to bind to multiple addresses was removed by
	  mistake. This patch restores this functionality and adds notes
	  about IPv6 addresses in the sample config. (closes issue
	  ASTERISK-22741) Reported by: Joshua Colp Tested by: Michael L.
	  Young Patches: asterisk-22741-fix-binding-multiple-addr.diff
	  uploaded by Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2945/ ........ Merged
	  revisions 401488 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-22 23:10 +0000 [r401450]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP
	  is not available during SSRC change In r400089, a patch was put
	  in to correct erroneous RTCP statistic resets. Unfortunately,
	  ast_rtp_read can be called on an RTP instance that does not have
	  RTCP information. This patch prevents that crash by only
	  resetting the statistics if we do actually have an RTCP instance.
	  (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John
	  Bigelow ........ Merged revisions 401445 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401446 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401447 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-22 19:04 +0000 [r401421-401435]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c, /: app_queue: Fix CLI "queue remove member"
	  queue_log entry. The queue_log entry resulting from CLI "queue
	  remove member" when log_membername_as_agent is enabled is wrong.
	  It always uses the interface name instead of the member name in
	  the queue_log entry. * Get the queue member before removing it
	  from the queue so the member name is available for the queue_log
	  entry. (closes issue ASTERISK-21826) Reported by: Oscar Esteve
	  Patches: fix_membername.diff (license #6505) patch uploaded by
	  Oscar Esteve (modified to fix potential ref leak) ........ Merged
	  revisions 401433 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401434 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/bridge_channel.c,
	  include/asterisk/bridge_channel_internal.h, /, main/bridge.c:
	  Bridging: Fix orphaned bridge if neither of the joining channels
	  can join. The original issue noted that the bridge is orphaned
	  when res_parking.so is not loaded and a call uses the dial kK
	  flags. A similar issue happens when only one of the park flags is
	  used. In this case you have the bridge with one or the other
	  channel left in it. The channel and bridge will stay around until
	  the channel hangs up. * Fixed the initial bridge channel push
	  failure to act as if the channel were kicked out of the bridge.
	  The bridge then decides if it needs to be dissolved. (closes
	  issue ASTERISK-22629) Reported by: Kevin Harwell Review:
	  https://reviewboard.asterisk.org/r/2928/ ........ Merged
	  revisions 401424 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/parking/parking_bridge_features.c,
	  res/parking/parking_bridge.c, /: res_parking: Give parking
	  timeout comebacktoorigin channel DTMF features. Parking timeouts
	  did not set any DTMF features for the channel calling the parker
	  back. * Added code to set the parkedcalltransfers,
	  parkedcallreparking, parkedcallhangup, and parkedcallrecording
	  options appropriately for the channels when a parking timeout
	  occurs. The recall channel DTMF options are set using the
	  BRIDGE_FEATURES channel variable to allow the other timeout
	  options to have the DTMF features available. (closes issue
	  ASTERISK-22630) Reported by: Kevin Harwell Review:
	  https://reviewboard.asterisk.org/r/2942/ ........ Merged
	  revisions 401422 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_parking.c: res_parking: Update XML documention for
	  DTMF features after parking timeout. * Updated the XML
	  documentation to indicate that the parkedcalltransfers,
	  parkedcallreparking, parkedcallhangup, and parkedcallrecording
	  configuration options also apply to parking timeouts. (issue
	  ASTERISK-22630) Reported by: Kevin Harwell Review:
	  https://reviewboard.asterisk.org/r/2942/ ........ Merged
	  revisions 401420 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-22 15:17 +0000 [r401411]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Add an 'R' option to Dial which sends ringing
	  until early media has been received. (closes issue
	  ASTERISK-10487) Reported by: Gaspar Zoltan Patches: 10487.patch
	  uploaded by n8ideas (license 6075)

2013-10-21 21:06 +0000 [r401365]  Mark Michelson <mmichelson@digium.com>

	* /, main/bridge_channel.c: Remove a noisy debug message from
	  bridging code. This particular debug message, during a stress
	  test, was logged so often that it appeared that there may be a
	  memory leak in the logger code. In actuality, there was no memory
	  leak, but the logger thread was having a hard time keeping up
	  with the demands of the rest of the system. Since this debug
	  message has no value at all, the best way to fix the problem was
	  to just remove the message. (closes issue AST-1225) reported by
	  John Bigelow Patches: spammy_log.diff uploaded by Mark Michelson
	  (License #5049) ........ Merged revisions 401364 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-21 19:50 +0000 [r401328]  Kevin Harwell <kharwell@digium.com>

	* /, main/editline/term.c: Segfault in LIBEDIT_INTERNAL after
	  tgetstr(), when libncurses5-dev isn't installed Include the
	  appropriate declarations when not using termcap, but term+curses
	  and [n]curses do not exist. (closes issue ASTERISK-22351)
	  Reported by: A. Iglesias Patches:
	  issueA22351_libedit_internal_without_ncurses_dev.patch uploaded
	  by wdoekes (license 5674) ........ Merged revisions 401325 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401326 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401327 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-21 18:59 +0000 [r401316-401317]  David M. Lee <dlee@digium.com>

	* rest-api/api-docs/channels.json, /: Fixing r401281; the model
	  name is Channel, with a capital C ........ Merged revisions
	  401315 from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_ari.c, /: Fixed malformed Access-Control-Allow-Methods
	  header. Was causing Safari to barf on POST and DELETE. ........
	  Merged revisions 401106 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-19 21:57 +0000 [r401292]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_iax2.c: Fix IAX2 incoming call address lookups
	  This fixes address lookup for incoming calls without a peer
	  definition. The address family was unset instead of being set to
	  AST_AF_UNSPEC which was causing lookup failures on "127.0.0.1".
	  This is one of the causes of the current failure of the app_page
	  integration test. Review:
	  https://reviewboard.asterisk.org/r/2933/ ........ Merged
	  revisions 401291 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-19 14:45 +0000 [r401282]  Joshua Colp <jcolp@digium.com>

	* res/ari/resource_channels.h, main/pbx.c, /,
	  rest-api/api-docs/channels.json, res/ari/resource_channels.c,
	  res/res_ari_channels.c: Return a channel snapshot when
	  originating using ARI, and subscribe the Stasis application to
	  it. This change allows a user of ARI to know what channel it has
	  originated and also follow any progress. If a Stasis application
	  is provided it will be automatically subscribed to the originated
	  channel immediately. (closes issue ASTERISK-22485) Reported by:
	  David Lee Review: https://reviewboard.asterisk.org/r/2910/
	  ........ Merged revisions 401281 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-18 22:52 +0000 [r401272]  Richard Mudgett <rmudgett@digium.com>

	* /, res/parking/parking_controller.c: res_parking: Remove setting
	  useless flag. ........ Merged revisions 401271 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-18 21:51 +0000 [r401263]  David M. Lee <dlee@digium.com>

	* contrib/scripts/get_swagger_ui.sh (added), Makefile, /,
	  static-http: This is just a quick script for dumping swagger-ui
	  into static-http, so that it can be served by the Asterisk web
	  server. I had to change the Makefile in order to recursively
	  install content from the static-http directory, hence the code
	  review instead of just putting it in. Review:
	  https://reviewboard.asterisk.org/r/2924/ ........ Merged
	  revisions 401261 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-18 18:44 +0000 [r401249]  Mark Michelson <mmichelson@digium.com>

	* main/sorcery.c, main/cli.c, main/manager.c, /, main/bridge.c,
	  main/bucket.c: Resolve some memory leaks due to incorrect for
	  loop / ao2 ref usage. A common idiom in Asterisk is to due
	  something like: for (ao2_obj = list_beginning; ao2_obj =
	  next_item; ao2_ref(ao2_obj, -1)) { ...do stuff... } This is nice
	  because it automatically takes care of the object references for
	  you. However, there is a pitfall here. If a break statement is in
	  the for loop, then the current reference is not cleaned up. In
	  some cases, this is on purpose, but in others there is a leak.
	  This commit fixes the leak cases. ........ Merged revisions
	  401248 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-18 16:59 +0000 [r401233-401240]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_fax.c, include/asterisk/channel.h, apps/app_dial.c,
	  main/channel.c: Add channel lock protection around translation
	  path setup. Most callers of ast_channel_make_compatible() happen
	  before the channels enter a two party bridge. With the new
	  bridging framework, two party bridging technologies may also call
	  ast_channel_make_compatible() when there is more than one thread
	  involved with the two channels. * Added channel lock protection
	  in set_format() and ast_channel_make_compatible_helper() when
	  dealing with the channel's native formats while setting up a
	  translation path. * Fixed best_src_fmt and best_dst_fmt usage
	  consistency in ast_channel_make_compatible_helper(). The call to
	  ast_translator_best_choice() got them backwards. * Updated some
	  callers of ast_channel_make_compatible() and the function
	  documentation. There is actually a difference between the two
	  channels passed in. * Fixed the deadlock potential in res_fax.c
	  dealing with ast_channel_make_compatible(). The deadlock
	  potential was already there anyway because res_fax called
	  ast_channel_make_compatible() with chan locked. (closes issue
	  ASTERISK-22542) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2915/ ........ Merged
	  revisions 401239 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, include/asterisk/bridge.h: Tweak ast_bridge_depart() doxygen.
	  ........ Merged revisions 401232 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-18 16:06 +0000 [r401216-401224]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/bridge.h, /: Remove the bit about requiring
	  ast_bridge_depart() to be called before ast_bridge_destroy().
	  ........ Merged revisions 401223 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/bridge.h, /: Clarify in ast_bridge_destroy()
	  about how departable channels must be handled. ........ Merged
	  revisions 401212 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-18 15:14 +0000 [r401184]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_sip.c: Remove Port Restriction When Checking For
	  NAT When trying to determine if a peer is behind NAT, we should
	  not be using the ports when comparing addresses. This patch
	  removes the port from being checked and just useds the addresses
	  now. (closes issue ASTERISK-22729) Reported by: Michael L. Young
	  Tested by: Michael L. Young Patches:
	  asterisk-remove-using-port-for-nat-check.diff uploaded by Michael
	  L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2927/ ........ Merged
	  revisions 401182 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401183 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-18 14:50 +0000 [r401181]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/channel.c, /: Properly copy/remove the device state cache
	  flag over a masquerade. In r378303 the
	  AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the
	  devstate system to not cache states for non-real devices.
	  However, when optimizing away channels (ast_do_masquerade), that
	  flag wasn't copied. In my case, using Local devices as queue
	  members created a situation where the endpoint was considered in
	  use, but the state change of the device being available again was
	  ignored (not cached). The endpoint channel was optimized into the
	  (previously) Local channel, but kept the do-not-cache flag. The
	  end result being that the queue member apparently stayed in use
	  forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes
	  Review: https://reviewboard.asterisk.org/r/2925/ ........ Merged
	  revisions 401178 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401179 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401180 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-17 20:39 +0000 [r401169]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_sip.c: Fix Setting A chan_sip Dialog's
	  SIP_NAT_FORCE_RPORT Flag A condition was added in a commit to fix
	  ASTERISK-21374, that, if the SIP_PAGE3_NAT_AUTO_RPORT flag was
	  set, to then copy a peer's SIP_NAT_FORCE_RPORT flag to the
	  dialog. This condition should not have been there since it
	  assumed that if Asterisk is in an environment where NAT is
	  involved, that the auto_* nat settings or force_rport setting
	  would be on in the global settings. If the nat setting in the
	  global setting is set to 'nat=no' and then turned on for peers
	  (which is not quite the recommended way, although it is allowed)
	  this flag is never copied to the dialog resulting in problems
	  like, REGISTER replies going to the wrong port. This patch
	  removes this conditional check and will now always use the peer's
	  flag which by this point in the code the checks on whether the
	  peer is behind NAT or not (if using auto_force_rport) have
	  already been run. (closes issue ASTERISK-22236) Reported by:
	  Filip Frank Tested by: Michael L. Young Patches:
	  asterisk-2236-always-set-rport.diff uploaded by Michael L. Young
	  (license 5026) Review: https://reviewboard.asterisk.org/r/2919/
	  ........ Merged revisions 401167 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401168 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-17 18:25 +0000 [r401159]  Jonathan Rose <jrose@digium.com>

	* res/res_parking.c, /: res_parking: Fix bug where reloading
	  immediately wipes new parkpos extensions (closes issue
	  ASTERISK-22631) Reported by: Kevin Harwell ........ Merged
	  revisions 401158 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-17 15:41 +0000 [r401122]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_xmpp.c, res/res_jabber.c: Reduce log level of a
	  non-pubsub error message Drop an error log message to debug level
	  1 since distributed device state functions correctly when
	  receiving this message and it spams the logs. (closes issue
	  ASTERISK-22410) Reported by: abelbeck Patches:
	  asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch
	  uploaded by abelbeck (License 5903)
	  asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded
	  by abelbeck (License 5903) ........ Merged revisions 401119 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401120 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401121 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-16 21:22 +0000 [r401108]  Richard Mudgett <rmudgett@digium.com>

	* /, res/ari/resource_playback.c: ARI: Fix crash when POST
	  /playback/{id}/control does not have an operation parameter.
	  (closes issue ASTERISK-22680) Reported by: John Bigelow ........
	  Merged revisions 401107 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-16 17:01 +0000 [r401097]  David M. Lee <dlee@digium.com>

	* rest-api/resources.json, /: Oops. Leftover /stasis reference
	  ........ Merged revisions 401096 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-16 14:02 +0000 [r401088]  Kinsey Moore <kmoore@digium.com>

	* rest-api/api-docs/bridges.json, res/ari/resource_channels.h, /,
	  res/ari/resource_bridges.h, rest-api/api-docs/channels.json:
	  Clarify documentation for channel and bridge list This makes it
	  clear that the ARI API calls for listing channels and bridges
	  will list all channels or bridges in the system and not just
	  those that are in or are controlled by a Stasis application.
	  (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........
	  Merged revisions 401087 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-16 12:19 +0000 [r401079]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, apps/app_queue.c: Don't check all realtime queues when doing
	  "queue show some_queue". When using realtime queues, queues have
	  to be fetched from the database every now and then to see if any
	  info has been changed or to see if the queue has been removed.
	  When fetching info for an individual queue, the pruning of other
	  queues is unnecessarily costly. Review:
	  https://reviewboard.asterisk.org/r/2907/ ........ Merged
	  revisions 401049 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 401076 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401077 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-16 00:12 +0000 [r401041]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, rest-api/api-docs/bridges.json, res/res_ari_bridges.c: Use
	  POST / DELETE to toggle ARI bridge moh Review:
	  https://reviewboard.asterisk.org/r/2911/ ........ Merged
	  revisions 401040 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-15 23:44 +0000 [r401020-401039]  Richard Mudgett <rmudgett@digium.com>

	* main/translate.c: translate.c: Some minor code tweaks. *
	  Consistently compare format2index() return value so matrix_get()
	  cannot get passed negative values. * Optimize
	  ast_translator_best_choice() to defer initializing things until
	  needed. Also cached the matrix_get() return value rather than
	  repeatedly calling it.

	* /, channels/dahdi/bridge_native_dahdi.c: bridge_native_dahdi:
	  Return channel join failure if could not make the channels
	  compatible. ........ Merged revisions 401030 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_iax2.c: chan_iax2: Fix channel left locked in
	  off nominal code path. ........ Merged revisions 401016 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 401017 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-15 20:03 +0000 [r401019]  Kinsey Moore <kmoore@digium.com>

	* rest-api/api-docs/bridges.json, res/res_ari_bridges.c, /: Ensure
	  bridge record error responses validate This adds the list of
	  expected errors to the /bridges/{bridgeId}/record ARI
	  documentation so that outbound 4xx errors validate properly.
	  Previously, this would result in a response validation failure.
	  (closes issue ASTERISK-22627) Reported by: Joshua Colp ........
	  Merged revisions 401018 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-15 15:30 +0000 [r401007]  Paul Belanger <paul.belanger@polybeacon.com>

	* rest-api/api-docs/channels.json, res/res_ari_channels.c, /: Use
	  POST / DELETE to toggle hold / moh for ARI channels This change
	  updates how we handle toggle events, rather then create two
	  different function names, we'll just use POST / DELETE from HTTP
	  to handle it. Review: https://reviewboard.asterisk.org/r/2906/
	  ........ Merged revisions 400999 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-15 15:26 +0000 [r400998]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Prevent chan_sip from sending duplicate
	  BYEs. When a 200 OK for an initial INVITE is received, we were
	  doing the right thing by ACKing and sending an immediate BYE.
	  However, we also were doing the wrong thing and queuing an answer
	  frame, thus causing the call to be answered. This would cause the
	  call to be hung up by the channel thread, thus resulting in a
	  second BYE being sent out. In this fix, I also have set the
	  hangupcause to be correct since the initial BYE being sent by
	  Asterisk had an unknown hangup cause. I have changed to using
	  "Bearer capabilty not available" since the call was hung up due
	  to an SDP offer/answer error. (closes issue ASTERISK-22621)
	  reported by Kinsey Moore ........ Merged revisions 400970 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400971 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400984 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-15 13:44 +0000 [r400959]  David M. Lee <dlee@digium.com>

	* /, rest-api-templates/asterisk_processor.py: My doc correction in
	  r400842 had a silly bug. Because I added a wiki_description to
	  models and not their properties, the rendered wiki page had the
	  model description instead of the property descriptions, which
	  looks very silly indeed. (closes issue ASTERISK-22705) ........
	  Merged revisions 400958 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-14 22:52 +0000 [r400913-400950]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
	  channels/chan_dahdi.h: chan_dahdi: Add config support for hwgain
	  settings. * Add hwtxgain and hwrxgain config options to
	  chan_dahdi.conf with documentation in chan_dahdi.conf.sample.
	  (closes issue ASTERISK-22429) Reported by: Jaco Kroon Patches:
	  jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch
	  uploaded by rmudgett

	* channels/chan_dahdi.c, /, channels/chan_dahdi.h: chan_dahdi:
	  Reflect the set software gain in the CLI "dahdi show channel"
	  output. * Remember the swgain setting from CLI "dahdi set swgain"
	  command so the CLI "dahdi show channel" output will reflect the
	  current setting. * Updated CLI "dahdi set hwgain" and "dahdi set
	  swgain" documentation. (issue ASTERISK-22429) Reported by: Jaco
	  Kroon Patches: jira_asterisk_22429_v1.8_v2.patch (license #5621)
	  patch uploaded by rmudgett ........ Merged revisions 400907 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400909 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400911 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-14 22:03 +0000 [r400912]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: chan_sip: Do not increment the SDP
	  version between 183 and 200 responses. Bumping the SDP version
	  number can cause interoperability problems since receivers of the
	  responses will expect that a 200 SDP will be identical to a
	  previous 183 SDP. (closes issue ASTERISK-21204) reported by
	  NITESH BANSAL Patches:
	  dont-increment-session-version-in-2xx-after-183.patch uploaded by
	  NITESH BANSAL (License #6418) ........ Merged revisions 400906
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 400908 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400910 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-14 15:54 +0000 [r400891]  Kevin Harwell <kharwell@digium.com>

	* /, res/res_pjsip_outbound_registration.c: pjsip outbound
	  registration: Log message says received a 408 when we didn't If
	  the server didn't exist that we are trying to register to the log
	  message would say that a 408 was received from that server when
	  in reality one wasn't. Added log messages stating no response was
	  received if the response does not exist. (closes issue
	  ASTERISK-22554) Reported by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/2893/ ........ Merged
	  revisions 400890 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-14 15:01 +0000 [r400882]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip_mwi.c, /: Remove duplicate module info block The
	  module info block was repeated twice. Once is sufficient.
	  ........ Merged revisions 400881 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-13 15:42 +0000 [r400873]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_session.c, /: Fix a race condition in
	  res_pjsip_session with rapidly terminating the session. The
	  INVITE session state callback wrongly assumes that a session will
	  always exist, but when rapidly terminating the session this
	  assumption goes out the window. As all handler code for the
	  INVITE session state callback requires the session it will now
	  just exit immediately if no session exists. (closes issue
	  ASTERISK-22668) Reported by: John Bigelow ........ Merged
	  revisions 400872 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-12 16:53 +0000 [r400864]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_pjsip_outbound_authenticator_digest.c: Fix realm
	  comparison for outbound auth When generating the list of
	  authentication credentials to pass to PJSIP, Asterisk was using
	  the raw pointer of a pj_str_t which is not always
	  NULL-terminated. This sometimes resulted in incorrect text for
	  the realm and a failure to match the realm for authentication
	  purposes which was causing the outbound nominal auth pjsip basic
	  call test to bounce. This now uses the pj_str_t that contains the
	  realm instead of generating a new one. Thanks to John Bigelow for
	  helping to narrow this down. ........ Merged revisions 400863
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-11 17:05 +0000 [r400855]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, /: channel.h: whitespace changes.
	  ........ Merged revisions 400854 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-11 16:36 +0000 [r400851-400852]  David M. Lee <dlee@digium.com>

	* /, res/ari/resource_bridges.h, rest-api/api-docs/playback.json,
	  rest-api-templates/api.wiki.mustache, res/res_ari_playback.c,
	  rest-api/api-docs/channels.json, res/ari/resource_playback.h,
	  rest-api/api-docs/bridges.json,
	  rest-api-templates/asterisk_processor.py,
	  res/ari/resource_channels.h,
	  rest-api-templates/models.wiki.mustache: Multiple revisions
	  400508,400842-400843,400848 ........ r400508 | dlee | 2013-10-03
	  23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line Corrected response
	  class for stopPlayback ........ r400842 | dlee | 2013-10-10
	  14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line Correct some ARI wiki
	  rendering errors ........ r400843 | dlee | 2013-10-10 14:26:19
	  -0500 (Thu, 10 Oct 2013) | 1 line Updated /play resource docs.
	  The playback of http: resources isn't implemented... yet ........
	  r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5
	  lines Fix a stupid copy/paste error in ARI docs. Patches:
	  ari-doc-patch.txt uploaded by jbigelow (license 5091) ........
	  Merged revisions 400508,400842-400843,400848 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /: Fixed merge tracking for r400360, which was somehow lost

2013-10-11 16:28 +0000 [r400850]  Richard Mudgett <rmudgett@digium.com>

	* bridges/bridge_softmix.c, /: Softmix: Fix crash when switching
	  from softmix to another bridge technology. The crash is caused by
	  a race condition when switching between native RTP and softmix
	  bridging technologies. In this situation, the bridging technology
	  is switched from native RTP to softmix, and then back to native
	  RTP fast enough that the softmix private data gets destroyed
	  before the softmix mixing thread gets started. Thanks to Kinsey
	  Moore for the crash analysis. * Fix race condition when starting
	  the softmix mixing thread and switching to another bridge
	  technology. (closes issue ASTERISK-22678) Reported by: John
	  Bigelow Patches: jira_asterisk_22678_v12.patch (license #5621)
	  patch uploaded by rmudgett Tested by: John Bigelow ........
	  Merged revisions 400849 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-10 18:21 +0000 [r400825-400834]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip/location.c: Perform validation of permanent
	  contacts on AORs in res_pjsip. ........ Merged revisions 400833
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c: Fix an
	  assertion in res_pjsip when specifying an invalid outbound proxy.
	  This change fixes two issues when setting an outbound proxy: 1.
	  The outbound proxy URI was not parsed and validated during
	  configuration. 2. If an outgoing dialog was created and the
	  outbound proxy could not be set an assertion would occur because
	  the usage count on the dialog was not decremented. The
	  documentation has also been updated to specify that a full URI
	  must be specified for the outbound proxy. (closes issue
	  ASTERISK-22672) Reported by: Antti Yrjola ........ Merged
	  revisions 400824 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-09 11:02 +0000 [r400772-400813]  Matthew Jordan <mjordan@digium.com>

	* res/res_pjsip_header_funcs.c, /: Use 'z' as the format specifier
	  for size_t Using 'lu' will produce a compiler warning for some
	  versions of gcc and on some architectures. 'z' should be portable
	  as a format specifier for size_t. ........ Merged revisions
	  400812 from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_header_funcs.c (added), /: Add PJSIP_HEADER
	  function for manipulation of SIP headers in the PJSIP stack This
	  patch adds support to the PJSIP stack in Asterisk for SIP header
	  manipulation. Note that this is analagous to
	  SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming
	  supplemental session callback is registered that takes the
	  pjsip_hdrs from the incoming session and stores them in a linked
	  list in the session datastore. Calls to PJSIP_HEADER traverse
	  over the list and return the nth matching header where 'n' is the
	  'number' argument to the function. When adding a header, the
	  first call creates a datastore and linked list and adds the
	  datastore to the session. The header is then created as a
	  pjsip_hdr and added to the list. An outgoing supplemental session
	  callback then traverses the list and adds the headers to the
	  outgoing pjsip_msg. When removing a header, the list created with
	  PJSIP_HEADER(add,...) is traversed and all matching entries are
	  removed. (closes issue ASTERISK-22498) Reported by: George Joseph
	  patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph
	  (License 6322) ........ Merged revisions 400771 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-08 22:33 +0000 [r400770]  Kinsey Moore <kmoore@digium.com>

	* /, configure, configure.ac: Add warning when compiling with iODBC
	  support When running configure, libiodbc2 development headers
	  will fulfill the requirement for ODBC development headers, but
	  will not function properly. This adds a warning when libiodbc2
	  development headers are detected instead of unixodbc development
	  headers. (closes issue ASTERISK-22459) Reported by: Patrick
	  Maille Tested by: Walter Doekes Patches:
	  issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes
	  (License 5674) ........ Merged revisions 400767 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400768 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400769 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-08 21:20 +0000 [r400759]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_agent_pool.c, /: app_agent_pool: Fix AMI/CLI AgentLogoff
	  soft preventing agents from logging back in. * Clear the
	  deferred_logoff flag when an agent logs in. (closes issue
	  ASTERISK-22669) Reported by: John Bigelow ........ Merged
	  revisions 400754 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-08 20:52 +0000 [r400750]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip.c, res/res_pjsip/config_transport.c: Switch from
	  using pjsip_strerror to pj_strerror. pjsip_strerror is only aware
	  of PJSIP-specific error codes. pj_strerror() is aware of all
	  PJProject error codes and OS-specific error codes. This
	  specifically fixes an oft-seen error in transport configuration
	  code where EADDRINUSE would result in "Unknown PJSIP error
	  120098" instead of a useful message. ........ Merged revisions
	  400749 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-08 20:18 +0000 [r400728-400744]  Richard Mudgett <rmudgett@digium.com>

	* configs/confbridge.conf.sample, /,
	  apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
	  CHANGES, apps/confbridge/conf_config_parser.c: app_confbridge:
	  Can now set the language used for announcements to the
	  conference. ConfBridge now has the ability to set the language of
	  announcements to the conference. The language can be set on a
	  bridge profile in confbridge.conf or by the dialplan function
	  CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983)
	  Reported by: Jonathan White Patches: M19983_rev2.diff (license
	  #5138) patch uploaded by junky (modified) Tested by: rmudgett
	  ........ Merged revisions 400741 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400742 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix
	  duplicate default_user profile. * Fixed looking in the wrong
	  profiles container to see if the default_user profile is already
	  created in verify_default_profiles(). The bridge profile
	  container is never going to hold user profiles. :) ........
	  Merged revisions 400723 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400724 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-08 18:19 +0000 [r400684-400704]  Kinsey Moore <kmoore@digium.com>

	* funcs/func_config.c, /: Fix func_config list entry allocation The
	  AST_CONFIG dialplan function defined in func_config.c allocates
	  its config file list entries using ast_malloc. List entry
	  allocations destined for use with Asterisk's linked list API must
	  be ast_calloc()d or otherwise initialized so that list pointers
	  are set to NULL. These uses of ast_malloc have been replaced by
	  ast_calloc to prevent dereferencing of uninitialized pointer
	  values when traversing the list. (closes issue ASTERISK-22483)
	  Reported by: Brian Scott ........ Merged revisions 400694 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400697 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400701 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_rtp_asterisk.c, /: Fix STUN crash when using IPv6 any
	  address Ensure that when chan_sip binds to the IPv6 any address
	  ([::]), IPv4 candidates are also added. (closes issue
	  ASTERISK-21917) Reported by: Torrey Searle Patches:
	  0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License
	  5334) ........ Merged revisions 400681 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400682 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-08 15:44 +0000 [r400683]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip/pjsip_options.c, /: Push CLI qualify into the
	  threadpool. If you run Asterisk in the background and then
	  connect to it through a separate console, the thread that runs
	  CLI commands is not registered with PJLIB. Thus PJLIB does not
	  like it when you attempt to send OPTIONS requests from that
	  thread. So now we push the task into the threadpool, which we
	  know to be registered with PJLIB. Thanks to Antti Yrjola for
	  reporting this. ........ Merged revisions 400680 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-08 15:12 +0000 [r400662-400672]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_agi.c, apps/app_queue.c: Make app_queue and res_agi
	  independent of AMI being enabled. The
	  https://reviewboard.asterisk.org/r/2888/ review changes manager
	  to not subscribe to stasis when it is disabled for performance
	  reasons. When manager is disabled app_queue and res_agi decline
	  to load and fail to clean up what they have already allocated. *
	  Made app_queue and res_agi clean up allocated resources when they
	  decline to load. * Made app_queue and res_agi use their own
	  subscriptions to the stasis topics instead of borrowing manager's
	  message router structure inappropriately. (closes issue
	  ASTERISK-22604) Reported by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/2902/ ........ Merged
	  revisions 400671 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, include/asterisk/stasis.h, apps/app_queue.c,
	  include/asterisk/manager.h: Miscellaneous stand alone comment
	  cleanups. ........ Merged revisions 400661 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-06 17:13 +0000 [r400625]  Michael L. Young <elgueromexicano@gmail.com>

	* /, apps/app_queue.c: app_queue: Fix Queuelog EXITWITHKEY only
	  logging two of four fields Commit r62462 added two extra fields
	  for logging "the original position the caller entered the queue
	  at, and the amount of time the caller was waiting in the queue."
	  But when r75969 was merged from 1.4 into trunk (r75977), these
	  two fields disappeared. Those two extra fields were not logged in
	  1.4 and when the patch was merged, those fields went away.
	  Therefore, this is a regression and was caught by the reporter
	  because he was reading the awesome "Asterisk: The Definitive
	  Guide" book. (closes issue ASTERISK-22197) Reported by: Dalius M.
	  Tested by: Dalius M. Patches:
	  asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2901/ ........ Merged
	  revisions 400622 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400623 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400624 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-05 00:59 +0000 [r400593]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/iax2/include/parser.h: chan_iax2: Fix compile error.
	  ........ Merged revisions 400588 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-04 21:41 +0000 [r400568]  Michael L. Young <elgueromexicano@gmail.com>

	* main/acl.c, include/asterisk/netsock2.h, CHANGES,
	  channels/chan_iax2.c, channels/iax2/parser.c, main/netsock.c,
	  main/netsock2.c, /, channels/iax2/include/parser.h: Add IPv6
	  Support To chan_iax2 This patch adds IPv6 support to chan_iax2.
	  Yay! (closes issue ASTERISK-22025) Patches:
	  iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026)
	  Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged
	  revisions 400567 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-04 19:32 +0000 [r400553]  David M. Lee <dlee@digium.com>

	* rest-api/api-docs/applications.json (added), /: Added missing
	  file from r400522 ........ Merged revisions 400552 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-04 19:11 +0000 [r400533-400543]  Jonathan Rose <jrose@digium.com>

	* res/res_pjsip_logger.c, /: chan_pjsip: Make logger togglable
	  without loading/unloading This patch makes the res_pjsip_logger
	  do a few things... First, it will be built and installed by
	  default now, so end users won't need to enable it in menuselect.
	  Second, while it is loaded, it no longer will immediately issue
	  log messages. Upon loading, it is in the disabled state and must
	  be turned on with the new CLI command. The CLI command 'pjsip set
	  logger <on/off/host> has been added and can be used to do the
	  following: pjsip set logger on: Enables logger for all PJSIP
	  traffic pjsip set logger off: Disables logger for all PJSIP
	  traffic pjsip set logger host <host>: Enables logger for the
	  specific host Review: https://reviewboard.asterisk.org/r/2900/
	  ........ Merged revisions 400542 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /,
	  contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py
	  (added), configs/extconfig.conf.sample,
	  configs/sorcery.conf.sample,
	  contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py:
	  chan_pjsip: Add alembic scripts for generating db tables for
	  PJSIP Also updates sample configurations for sorcery and
	  extconfig to demonstrate how to use databases created by that
	  alembic script. (closes issue ASTERISK-22133) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/2892/ ........
	  Merged revisions 400532 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-04 16:01 +0000 [r400523]  Matthew Jordan <mjordan@digium.com>

	* res/res_stasis.c, main/asterisk.c,
	  rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
	  res/stasis/app.c, /,
	  rest-api-templates/ari_model_validators.h.mustache,
	  include/asterisk/endpoints.h, res/res_ari_applications.c (added),
	  res/ari/resource_endpoints.h, include/asterisk/stasis_app.h,
	  res/stasis/app.h, rest-api/resources.json,
	  include/asterisk/_private.h, res/ari/ari_model_validators.c,
	  main/endpoints.c, res/ari/ari_model_validators.h, main/json.c,
	  res/res_ari_model.c, res/ari.make,
	  res/ari/resource_applications.c (added),
	  res/ari/resource_applications.h (added): ARI: Add subscription
	  support This patch adds an /applications API to ARI, allowing
	  explicit management of Stasis applications. * GET /applications -
	  list current applications * GET /applications/{applicationName} -
	  get details of a specific application * POST
	  /applications/{applicationName}/subscription - explicitly
	  subscribe to a channel, bridge or endpoint * DELETE
	  /applications/{applicationName}/subscription - explicitly
	  unsubscribe from a channel, bridge or endpoint Subscriptions work
	  by a reference counting mechanism: if you subscript to an event
	  source X number of times, you must unsubscribe X number of times
	  to stop receiveing events for that event source. Review:
	  https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451)
	  Reported by: Matt Jordan ........ Merged revisions 400522 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-04 15:49 +0000 [r400511-400521]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip.c: Enclose the To URI and update its user
	  portion if a request user has been specified. ........ Merged
	  revisions 400520 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_session.c, /: Replace the connection address at the
	  SDP level if altering the SDP with the external media address.
	  ........ Merged revisions 400510 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 23:20 +0000 [r400482]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: chan_sip: Don't ignore expires value in
	  contact header if it lacks semicolon (closes issue
	  ASTERISK-22574) Reported by: Filip Jenicek Patches:
	  chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
	  ........ Merged revisions 400469 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400470 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400471 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 21:46 +0000 [r400461]  Matthew Jordan <mjordan@digium.com>

	* /, main/channel_internal_api.c: Remove publication of a channel
	  snapshot when the technology is set This patch removes said
	  publication for a few reasons: (1) It is unnecessary. Association
	  of the channel technology with a specific channel is an
	  implementation detail that should be assumed to "just happen",
	  and consumers of Stasis don't need to be informed about it. (2)
	  Publication of said message can now cause crashes, as the actual
	  creation of a channel in normal locations now stages its
	  messages. As a result, things that create dummy channels (such as
	  the SIP RTP QOS unit test) and associate them with a channel
	  technology were now crashing, as the channel itself was not known
	  by Stasis. ........ Merged revisions 400460 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 20:22 +0000 [r400452]  Mark Michelson <mmichelson@digium.com>

	* bridges/bridge_native_rtp.c, /,
	  include/asterisk/bridge_technology.h: Fix assumption in
	  bridge_native_rtp.c regarding number of participants in a bridge.
	  When a party leaves a bridge, there may be more participants in
	  the bridge than expected. As such, it is important not to make
	  assumptions regarding the list of channels in a bridge. This
	  change makes it so that when a party leaves a native RTP bridge,
	  we unbridge it and the party it was bridged with. Previously, the
	  first and last channels in the list were unbridged since it was
	  assumed that these were the two channels that had been bridged.
	  As previously stated, a new party had been inserted into the
	  bridge, so this logic did not work properly. (closes issue
	  ASTERISK-22615) reported by Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2899 ........ Merged revisions
	  400403 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 19:32 +0000 [r400443]  Joshua Colp <jcolp@digium.com>

	* /, main/cdr.c: When serializing CDR variables (like for "core
	  show channels") don't output an error if CDRs aren't enabled.
	  ........ Merged revisions 400442 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 19:30 +0000 [r400441]  Kinsey Moore <kmoore@digium.com>

	* /, main/security_events.c: Fix security events for AMI invalid
	  password In r337595, additional security events were added for
	  chan_sip authentication failures. The new IEs added to the
	  existing invalid password event were defined as required IEs, but
	  existing users of the event did not set the new IEs and could not
	  since they didn't apply to existing uses. They are now marked as
	  optional IEs. (closes issue ASTERISK-22578) Reported by: Matt
	  Jordan ........ Merged revisions 400421 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400440 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 19:06 +0000 [r400402]  Joshua Colp <jcolp@digium.com>

	* res/ari/resource_channels.c, /: Fix a crash caused by muting and
	  unmuting a channel in ARI without specifying a direction. (closes
	  issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by
	  Matt Jordan, whose office I have taken over in the name of
	  Canada. ........ Merged revisions 400401 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 18:51 +0000 [r400399]  Richard Mudgett <rmudgett@digium.com>

	* /, main/cel.c: cel: Some whitespace cleanups ........ Merged
	  revisions 400398 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 18:32 +0000 [r400385-400397]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_multicast.c, /: res_rtp_multicast: Ensure SSRC is set
	  properly This fixes a bug where the SSRC field on multicast RTP
	  can be stuck at 0 which can cause problems for endpoints trying
	  to make sense of incoming streams. (closes issue ASTERISK-22567)
	  Reported by: Simone Camporeale Patches:
	  22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale
	  (License 6536) ........ Merged revisions 400393 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400394 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400395 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/xml.c: Detect and use xsltCleanupGlobals when available This
	  introduces usage of an additional libxslt cleanup function,
	  xsltCleanupGlobals, when the configure script detects that it is
	  available. Early versions of the library did not include this
	  function. (closes issue ASTERISK-22570) Reported by: Corey
	  Farrell Patches: xsltCleanupGlobals.patch uploaded by Corey
	  Farrell (License 5909) ........ Merged revisions 400384 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 16:28 +0000 [r400374]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_vpb.cc, /: chan_vpb: Make compile again. ........
	  Merged revisions 400373 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 14:59 +0000 [r400363-400364]  Mark Michelson <mmichelson@digium.com>

	* tests/test_cel.c, /: Get rid of uses of stasis_topic_wait()
	  ........ Merged revisions 400362 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* pbx/pbx_spool.c, main/manager.c, main/format_cap.c,
	  channels/chan_skinny.c, res/res_agi.c, channels/chan_motif.c,
	  channels/chan_alsa.c, apps/app_confbridge.c,
	  addons/chan_mobile.c, channels/chan_mgcp.c,
	  res/res_clioriginate.c, channels/chan_bridge_media.c,
	  channels/chan_sip.c, tests/test_format_api.c,
	  res/res_pjsip_sdp_rtp.c, bridges/bridge_simple.c,
	  apps/app_originate.c, res/parking/parking_applications.c,
	  main/core_local.c, channels/chan_console.c, channels/chan_oss.c,
	  include/asterisk/format_cap.h, res/res_pjsip_session.c,
	  res/ari/resource_bridges.c, channels/chan_jingle.c,
	  channels/chan_misdn.c, channels/dahdi/bridge_native_dahdi.c,
	  res/res_pjsip/pjsip_configuration.c, main/file.c,
	  channels/chan_h323.c, channels/chan_nbs.c,
	  bridges/bridge_native_rtp.c, tests/test_config.c,
	  res/res_stasis.c, channels/chan_pjsip.c, channels/chan_unistim.c,
	  channels/chan_multicast_rtp.c, addons/chan_ooh323.c,
	  main/rtp_engine.c, /, main/ccss.c, apps/app_meetme.c,
	  bridges/bridge_holding.c, main/bridge_basic.c,
	  bridges/bridge_softmix.c, channels/chan_gtalk.c,
	  channels/chan_iax2.c, main/media_index.c, main/channel.c,
	  channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c: Cache
	  string values of formats on ast_format_cap() to save processing.
	  Channel snapshots have string representations of the channel's
	  native formats. Prior to this change, the format strings were
	  re-created on ever channel snapshot creation. Since channel
	  native formats rarely change, this was very wasteful. Now, string
	  representations of formats may optionally be stored on the
	  ast_format_cap for cases where string representations may be
	  requested frequently. When formats are altered, the string cache
	  is marked as invalid. When strings are requested, the cache
	  validity is checked. If the cache is valid, then the cached
	  strings are copied. If the cache is invalid, then the string
	  cache is rebuilt and copied, and the cache is marked as being
	  valid again. Review: https://reviewboard.asterisk.org/r/2879
	  ........ Merged revisions 400356 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-03 14:52 +0000 [r400361]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c, /: Fix crashes in
	  res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and
	  external_media_address is set. The callback function for changing
	  the media address in streams wrongly assumes that a connection
	  line will always be present. This is false as no line is present
	  if a stream has been rejected. (closes issue ASTERISK-22645)
	  Reported by: Rusty Newton ........ Merged revisions 400360 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 22:22 +0000 [r400335]  Mark Michelson <mmichelson@digium.com>

	* main/stasis_wait.c (removed), res/ari/resource_endpoints.c, /,
	  include/asterisk/stasis.h, tests/test_cel.c,
	  include/asterisk/stasis_endpoints.h, channels/chan_pjsip.c,
	  main/stasis.c, main/stasis_endpoints.c: Multiple revisions
	  400318-400319 ........ r400318 | mmichelson | 2013-10-02 17:08:49
	  -0500 (Wed, 02 Oct 2013) | 12 lines Remove unnecessary waits from
	  stasis. Since caches are updated on publisher threads, there is
	  no need to wait for the cache updates to occur after a stasis
	  message is published. In the case of chan_pjsip device state
	  changes, this set of changes caused an improvement to
	  performance. Review: https://reviewboard.asterisk.org/r/2890
	  ........ r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed,
	  02 Oct 2013) | 3 lines Remove svn:mergeinfo property. ........
	  Merged revisions 400318-400319 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 21:33 +0000 [r400317]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_iax2.c, /: Cast Integer Argument To Unsigned Char
	  The member reg in the peercnt structure is an unsigned char and
	  peercnt_modify() is expecting an unsigned char argument which
	  gets assigned to peercnt->reg. This patch fixes that by casting
	  the integer argument being passed to peercnt_modify to unsigned
	  char. ........ Merged revisions 400314 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400315 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400316 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 21:26 +0000 [r400313]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, main/manager.c, /, main/cel.c: Only create Stasis
	  subscriptions when enabled Subscribing to Stasis isn't free. As
	  such, this patch makes AMI, CDR, and CEL - the "big 3" - only
	  subscribe when enabled. Toggling their availability via a .conf
	  file will unsubscribe/subscribe as appropriate. Review:
	  https://reviewboard.asterisk.org/r/2888/ ........ Merged
	  revisions 400312 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 20:31 +0000 [r400304]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /: Originate: Make setting caller id on outgoing call
	  use either name or number. Previous code was requiring both name
	  and number to be available. Also restored a comment block on why
	  caller id is also set on an outgoing call leg in addition to
	  connected line from earlier versions of Asterisk. ........ Merged
	  revisions 400303 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 19:20 +0000 [r400295]  Kinsey Moore <kmoore@digium.com>

	* /, rest-api/api-docs/asterisk.json: Correct allowable values for
	  ARI general information filter ........ Merged revisions 400291
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 19:17 +0000 [r400287]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /: Fix the CDR CLI command 'cdr show active
	  {channel}' When the switch from channel names to channel unique
	  IDs happened, the poor CLI command got left in the dust. This
	  fixes the command so that users can once again see how Asterisk
	  is messing up your billing information. ........ Merged revisions
	  400286 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 18:44 +0000 [r400285]  Joshua Colp <jcolp@digium.com>

	* /, res/res_pjsip_t38.c: Fix a crash in res_pjsip_t38 caused by
	  the wrong assumption that a session will always have a channel.
	  When starting up or shutting down this assumption is false.
	  ........ Merged revisions 400284 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 18:28 +0000 [r400282]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* Makefile, doc/astdb2sqlite3.8 (added), /, doc/astdb2bdb.8
	  (added): man pages for astdb2bdb and astdb2sqlite3 Review:
	  https://reviewboard.asterisk.org/r/2898/ ........ Merged
	  revisions 400279 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400281 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 17:12 +0000 [r400269-400271]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_stack.c, res/stasis_recording/stored.c, main/json.c,
	  main/stasis_cache.c, res/res_ari.c, /, main/utils.c:
	  MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is
	  enabled. * There were several places in ARI where an external
	  library was mallocing memory that must always be released with
	  free(). When MALLOC_DEBUG is enabled, free() is redirected to the
	  MALLOC_DEBUG version. Since the external library call still uses
	  the normal malloc(), MALLOC_DEBUG complains that the freed memory
	  block is not registered and will not free it. These cases must
	  use ast_std_free(). * Changed calls to asprintf() and vasprintf()
	  to the equivalent ast_asprintf() and ast_vasprintf() versions
	  respectively. ........ Merged revisions 400270 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/sig_ss7.c, /: sig_ss7: Fix compiler warnings. ........
	  Merged revisions 400268 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-02 16:23 +0000 [r400246-400266]  Joshua Colp <jcolp@digium.com>

	* channels/chan_alsa.c, main/stasis_channels.c, channels/sig_ss7.c,
	  channels/chan_pjsip.c, channels/chan_mgcp.c,
	  channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, /,
	  channels/chan_sip.c, main/bridge.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, channels/chan_console.c,
	  channels/sig_pri.c, channels/chan_iax2.c, channels/chan_jingle.c,
	  main/channel.c, channels/chan_dahdi.c, main/dial.c,
	  include/asterisk/stasis_channels.h, channels/chan_skinny.c,
	  channels/chan_motif.c: Reduce channel snapshot creation and
	  publishing by up to 50%. This change introduces the ability to
	  stage channel snapshot creation and publishing by suppressing the
	  implicit creation and publishing that some functions have. Once
	  all operations are executed the staging is marked as done and a
	  single snapshot is created and published. Review:
	  https://reviewboard.asterisk.org/r/2889/ ........ Merged
	  revisions 400265 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_session.c, /: Fix a random one way audio issue in
	  PJSIP. Due to the asynchronous design of the PJMEDIA SDP
	  negotiator it was possible for the SDP to be negotiated *after* a
	  channel was created and after it was being wait on by an
	  application. It is only after negotiation occurs that the file
	  descriptors for RTP are placed on the channel. Since the channel
	  was already being waited on these file descriptors were not
	  monitored, causing incoming media to never be read. This change
	  wakes up any application waiting on the channel so that added
	  file descriptors end up being monitored. (closes issue AST-1227)
	  Reported by: John Bigelow ........ Merged revisions 400256 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/stasis/control.c, include/asterisk/stasis_app.h,
	  res/ari/resource_channels.c: Allow specifying a channel to dial
	  an extension and context in an ARI dial operation. (issue
	  ASTERISK-22625) Reported by: Scott Griepentrog ........ Merged
	  revisions 400254 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip_session.c: Retrieve and store the hostname only
	  once so multiple threads do not potentially initialize it at the
	  same time. ........ Merged revisions 400245 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-01 21:19 +0000 [r400228-400237]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: chan_dahdi: Fix
	  analog parking using flash-hook. Transferring an analog call
	  using a flash-hook to parking would fail to park the call and
	  result in an invalid ao2 object unref. * Park the correct bridged
	  channel. ........ Merged revisions 400236 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/features_config.c, /: Features: Rearm the parking config
	  options have moved warning for each reload. ........ Merged
	  revisions 400227 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-10-01 15:54 +0000 [r400218]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /: Filter out internal channels for bridge leave
	  messages and parked call messages Granted, if you manage to park
	  a Conference announcer channel, something has gone horrifically
	  wrong. ........ Merged revisions 400217 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-30 21:40 +0000 [r400206]  Jonathan Rose <jrose@digium.com>

	* configs/features.conf.sample, /, configs/res_parking.conf.sample:
	  configuration samples: Pull all parking related stuff out of
	  features.conf This patch also adds documentation for parking from
	  features.conf to res_parking.conf ........ Merged revisions
	  400205 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-30 19:58 +0000 [r400195-400197]  Matthew Jordan <mjordan@digium.com>

	* /, funcs/func_cdr.c: Parse arguments passed to the CDR_PROP
	  function correctly I can only blame this on a bad merge, because
	  this in no way worked properly the way it was written. Mea culpa.
	  The function should now parse its arguments correctly and
	  function properly. (Note that the API used by the CDR_PROP
	  function has working unit tests... this was merely bad coding of
	  the actual registered function) (closes issue ASTERISK-22613)
	  Reported by: Private Name ........ Merged revisions 400196 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: Remove spurious event raised when CDRs are
	  reloaded The Reload event is now raised by the module loading
	  core. As such, the Reload event in the CDR engine was a duplicate
	  and not needed. ........ Merged revisions 400194 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-30 18:55 +0000 [r400186]  David M. Lee <dlee@digium.com>

	* tests/test_devicestate.c, include/asterisk/sem.h (added),
	  tests/test_taskprocessor.c, res/res_pjsip_mwi.c,
	  res/res_pjsip/include/res_pjsip_private.h, tests/test_stasis.c,
	  res/parking/parking_manager.c, res/res_security_log.c,
	  channels/chan_mgcp.c, main/stasis_cache_pattern.c, main/pbx.c,
	  include/asterisk/vector.h (added), /, main/ccss.c,
	  apps/app_meetme.c, include/asterisk/taskprocessor.h,
	  configs/stasis.conf.sample (removed), configure.ac,
	  res/parking/parking_applications.c, channels/sig_pri.c,
	  apps/app_queue.c, main/cel.c, main/stasis.c,
	  channels/chan_dahdi.c, funcs/func_presencestate.c,
	  main/stasis_message_router.c, configure,
	  apps/confbridge/confbridge_manager.c, res/res_agi.c,
	  main/manager_system.c, res/res_stasis_test.c, main/sem.c (added),
	  main/manager_channels.c, res/res_pjsip_refer.c,
	  main/manager_mwi.c, apps/app_voicemail.c, main/stasis_cache.c,
	  main/stasis_wait.c, main/stasis_config.c (removed),
	  include/asterisk/stasis_internal.h, res/stasis/app.c,
	  channels/chan_sip.c, include/asterisk/autoconfig.h.in,
	  main/manager_endpoints.c, main/channel_internal_api.c,
	  include/asterisk/stasis.h, main/devicestate.c,
	  main/taskprocessor.c, res/res_xmpp.c, main/sounds_index.c,
	  include/asterisk/stasis_message_router.h, channels/chan_iax2.c,
	  res/res_jabber.c, main/endpoints.c, main/astobj2.c,
	  res/res_chan_stats.c, res/parking/parking_bridge_features.c,
	  tests/test_stasis_endpoints.c, main/cdr.c, main/channel.c,
	  main/manager_bridges.c, main/manager.c, channels/chan_skinny.c:
	  Multiple revisions 399887,400138,400178,400180-400181 ........
	  r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1
	  line Minor performance bump by not allocate manager variable
	  struct if we don't need it ........ r400138 | dlee | 2013-09-30
	  10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance
	  improvements This patch addresses several performance problems
	  that were found in the initial performance testing of Asterisk
	  12. The Stasis dispatch object was allocated as an AO2 object,
	  even though it has a very confined lifecycle. This was replaced
	  with a straight ast_malloc(). The Stasis message router was
	  spending an inordinate amount of time searching hash tables. In
	  this case, most of our routers had 6 or fewer routes in them to
	  begin with. This was replaced with an array that's searched
	  linearly for the route. We more heavily rely on AO2 objects in
	  Asterisk 12, and the memset() in ao2_ref() actually became
	  noticeable on the profile. This was #ifdef'ed to only run when
	  AO2_DEBUG was enabled. After being misled by an erroneous comment
	  in taskprocessor.c during profiling, the wrong comment was
	  removed. Review: https://reviewboard.asterisk.org/r/2873/
	  ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep
	  2013) | 24 lines Taskprocessor optimization; switch Stasis to use
	  taskprocessors This patch optimizes taskprocessor to use a
	  semaphore for signaling, which the OS can do a better job at
	  managing contention and waiting that we can with a mutex and
	  condition. The taskprocessor execution was also slightly
	  optimized to reduce the number of locks taken. The only
	  observable difference in the taskprocessor implementation is that
	  when the final reference to the taskprocessor goes away, it will
	  execute all tasks to completion instead of discarding the
	  unexecuted tasks. For systems where unnamed semaphores are not
	  supported, a really simple semaphore implementation is provided.
	  (Which gives identical performance as the original taskprocessor
	  implementation). The way we ended up implementing Stasis caused
	  the threadpool to be a burden instead of a boost to performance.
	  This was switched to just use taskprocessors directly for
	  subscriptions. Review: https://reviewboard.asterisk.org/r/2881/
	  ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep
	  2013) | 28 lines Optimize how Stasis forwards are dispatched This
	  patch optimizes how forwards are dispatched in Stasis.
	  Originally, forwards were dispatched as subscriptions that are
	  invoked on the publishing thread. This did not account for the
	  vast number of forwards we would end up having in the system, and
	  the amount of work it would take to walk though the forward
	  subscriptions. This patch modifies Stasis so that rather than
	  walking the tree of forwards on every dispatch, when forwards and
	  subscriptions are changed, the subscriber list for every topic in
	  the tree is changed. This has a couple of benefits. First, this
	  reduces the workload of dispatching messages. It also reduces
	  contention when dispatching to different topics that happen to
	  forward to the same aggregation topic (as happens with all of the
	  channel, bridge and endpoint topics). Since forwards are no
	  longer subscriptions, the bulk of this patch is simply changing
	  stasis_subscription objects to stasis_forward objects (which,
	  admittedly, I should have done in the first place.) Since this
	  required me to yet again put in a growing array, I finally
	  abstracted that out into a set of ast_vector macros in
	  asterisk/vector.h. Review:
	  https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee
	  | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove
	  dispatch object allocation from Stasis publishing While looking
	  for areas for performance improvement, I realized that an unused
	  feature in Stasis was negatively impacting performance. When a
	  message is sent to a subscriber, a dispatch object is allocated
	  for the dispatch, containing the topic the message was published
	  to, the subscriber the message is being sent to, and the message
	  itself. The topic is actually unused by any subscriber in
	  Asterisk today. And the subscriber is associated with the
	  taskprocessor the message is being dispatched to. First, this
	  patch removes the unused topic parameter from Stasis subscription
	  callbacks. Second, this patch introduces the concept of
	  taskprocessor local data, data that may be set on a taskprocessor
	  and provided along with the data pointer when a task is pushed
	  using the ast_taskprocessor_push_local() call. This allows the
	  task to have both data specific to that taskprocessor, in
	  addition to data specific to that invocation. With those two
	  changes, the dispatch object can be removed completely, and the
	  message is simply refcounted and sent directly to the
	  taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/
	  ........ Merged revisions 399887,400138,400178,400180-400181 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-30 15:57 +0000 [r400142]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c, configs/pjsip.conf.sample,
	  res/res_pjsip_outbound_registration.c, configs/sip.conf.sample,
	  CHANGES: chan_sip: Allow Asterisk to retry after 403 on register
	  This adds a global option in chan_sip to allow it to continue
	  attempting registration if a 403 is received, clearing the cached
	  nonce and treating it as a non-fatal response. Normally, this
	  would cause registration attempts to that endpoint to stop. This
	  also adds a similar per-outbound-registration option to
	  chan_pjsip which allows the retry interval to be altered for 403
	  responses to REGISTER requests. (closes issue ASTERISK-17138)
	  Review: https://reviewboard.asterisk.org/r/2874/ Reported by:
	  Rudi ........ Merged revisions 400137 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400140 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400141 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-28 22:57 +0000 [r400059-400122]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_pjsip_notify.c, configs/pjsip_notify.conf.sample
	  (added): res_pjsip_notify: Add documentation We forgot to add
	  documentation for res_pjsip_notify, which would prevent it from
	  being loaded. Whoops. This patch also updates res_pjsip_notify to
	  use pjsip_notify.conf, which now has its own sample file in the
	  configs directory as well. Review:
	  https://reviewboard.asterisk.org/r/2835/ ........ Merged
	  revisions 400121 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Correct erroneous
	  lost packet information in RTCP reports RTCP's calculation of the
	  number of lost packets in an RTP stream is based on that stream's
	  sequence number count, the number of received packets, and how
	  many packets we expect to receive. When the SSRC for an RTP
	  stream changes, there can - and almost always will be - a large
	  jump in the next packet's timestamp and sequence number. If we
	  don't reset the number of received packets, sequence number
	  count, and other metrics used by RTCP, the next RR/SR report will
	  use the previous SSRC's values to calculate the lost packet count
	  for the new SSRC - resulting in a very large number of lost
	  packets. This patch modifies res_rtp_asterisk such that, if it
	  detects a SSRC change, it will reset the various values used by
	  the RTCP calculations. From the perspective of RTCP, this appears
	  as a new media stream - which is what it is. Review:
	  https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
	  Reported by: Thomas Arimont ........ Merged revisions 400089 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400093 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400108 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, configure, configure.ac: Add check for openSUSE when detecting
	  bfd library In ASTERISK-17842, some additional library checks
	  were added to the configure script so that the bfd library could
	  be found on CentOS and Fedora systems. As it turns out, openSUSE
	  requires an additional library. This patch adds another check to
	  the configure script for openSUSE that will add that library.
	  Review: https://reviewboard.asterisk.org/r/2885/ (closes issue
	  AST-1169) Reported by: Guenther Kelleter ........ Merged
	  revisions 400073 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400075 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400077 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: CDR: Improve handling of parking; resolve
	  assertion when originating into park This patch covers two
	  problems: 1) Currently, when a call is transferred into a parking
	  lot from a bridge (using either the blind transfer or one touch
	  parking mechanisms), the application fails to be set to "Park" in
	  the resulting CDR record for the parked channel. This is due to
	  the ParkedCall message arriving before the BridgeEnter for the
	  channel entering the parking bridge. The ParkedCall message isn't
	  handled as the CDR for the channel has already been finalized
	  (due to the channel having left its two party bridge), and the
	  BridgeEnter - which creates the new CDR - doesn't have the
	  parking information. This patch modifies the behavior so that
	  reception of a ParkedCall message will - if not handled by a CDR
	  chain - cause a new CDR to be created and put into the Parking
	  state. 2) It fixes a FRACK that occurred when a channel is
	  originated into a parking space. The DialedPending state - which
	  occurs for both Dialed and Originated channels - assumed that it
	  couldn't handle the parking transitions due to it having a Party
	  B; however, Originated channels don't have a Party B. As such,
	  the existing CDR needs to transition into the parking state -
	  this patch does that. Review:
	  https://reviewboard.asterisk.org/r/2877/ (closes issue
	  ASTERISK-22482) Reported by: Richard Mudgett ........ Merged
	  revisions 400062 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, apps/app_queue.c: app_queue: Make manager events tolerant of
	  Local channel shenanigans app_queue currently attempts to handle
	  Local channel optimizations in an effort to provide accurate
	  information in Stasis messages (and their corresponding AMI
	  events) as well as the Queue log. Sometimes, however, things
	  don't go as planned. Consider the following scenario: SIP/foo <->
	  L;1 <-> L;2 <-> SIP/agent SIP/agent answers, triggering a Local
	  channel optimization. app_queue will normally do the following: *
	  Listen for the Local optimization events and update our agent
	  accordingly to SIP/agent in the queue log and messages * When we
	  get a hangup, publish the AgentComplete event based on our
	  information (SIP/foo and SIP/agent) However, as with all things
	  that depend on sanity from something as capricious as Local
	  channels, things can go wrong: (1) SIP/agent immediately hangs up
	  upon answering. This triggers a race condition between
	  termination messages coming from SIP/agent and the ongoing Local
	  channel optimization messages. (Note that this can also occur
	  with SIP/foo) (2) In a race condition, Asterisk can (rarely)
	  deliver the hangup messages prior to the Local channel
	  optimization. In that case, the messages *may* arrive to
	  app_queue in the following order: * Hangup SIP/Agent * Hangup
	  SIP/foo * Optimize L;1/L;2 * Hangup L;2 * Hangup L;1 When
	  app_queue receives the hangup of the agent or the caller, it will
	  attempt to publish the AgentComplete event. However, it now has a
	  problem - it thinks its agent is the ;1 side of the Local
	  channel, as it never received the optimization event. At the same
	  time, that channel is already gone. This results in getting NULL
	  from the Stasis cache. What's more, we can't really wait for the
	  optimization message, as we are currently handling the hangup of
	  the channel that the optimization event would tell us to use.
	  This patch modifies the behavior in app_queue such that, since we
	  still have a lot of pertinent queue information (interface, queue
	  name, etc.), we now raise the event with what information we
	  know. The channels involved now may or may not be present. Users
	  will still at least get the "AgentComplete" event, which
	  "completes" the known Agent information. Review:
	  https://reviewboard.asterisk.org/r/2878/ (closes issue
	  ASTERISK-22507) Reported by: Richard Mudgett ........ Merged
	  revisions 400060 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/manager.c, /: manager: Fix crash when appending a manager
	  channel variable In r399887, a minor performance improvement was
	  introduced by not allocating the manager variable struct if it
	  wasn't used. Unfortunately, when directly accessing an
	  ast_channel struct, manager assumed that the struct was always
	  allocated. Since this was no longer the case, things got a bit
	  crashy. This fixes that problem by simply bypassing appending
	  variables if the manager channel variable struct isn't there.
	  ........ Merged revisions 400058 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-27 21:58 +0000 [r400016-400021]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_cdr.c, res/res_parking.c, /: app_cdr and res_parking:
	  Fix some resource leaks. * app_cdr left the ResetCDR application
	  registered. * res_parking leaked a ref to config global. (closes
	  issue ASTERISK-22566) Reported by: Corey Farrell Patches:
	  ASTERISK-22566-r2.patch (license #5909) patch uploaded by Corey
	  Farrell ........ Merged revisions 400020 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/sip/reqresp_parser.c, /, channels/chan_sip.c: chan_sip:
	  Increase some scratch buffer sizes dealing with caller id. *
	  Eliminated an unnecessary initialization in check_user_full().
	  (closes issue ASTERISK-22477) Reported by: Michael Shepelev
	  ........ Merged revisions 400013 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 400014 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 400015 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-27 19:18 +0000 [r400000]  Sean Bright <sean@malleable.com>

	* configs/sip.conf.sample: Remove some trailing whitespace and
	  steal revision 400000.

2013-09-27 18:28 +0000 [r399991]  Kevin Harwell <kharwell@digium.com>

	* /, res/res_pjsip.c, res/res_pjsip_session.c,
	  include/asterisk/res_pjsip.h, res/res_pjsip.exports.in:
	  res_pjsip: crash when using localnet and
	  external_signaling_address options There was a collision of
	  mod_data use on the transaction between using a nat hook and an
	  session response callback. During state change it was assumed
	  what was in the mod_data was nothing or the response callback.
	  However, it was possible for it to also contain a nat hook thus
	  resulting in a bad cast and a crash. Added the ability to store
	  multiple data elements in mod_data via a hash table. In this
	  instance, mod_data now stores a hash table of the two values that
	  can be retrieved using an associated string key. (closes issue
	  ASTERISK-22394) Reported by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/2843/ ........ Merged
	  revisions 399990 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-27 17:46 +0000 [r399978]  Jonathan Rose <jrose@digium.com>

	* channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
	  Reject calls on 200 OKs if no SDP has been received When Asterisk
	  receives a 200 OK in response to an invite, that peer should have
	  sent an SDP at some point by then. If the channel has never
	  received an SDP, media won't have been set and the remote address
	  won't be known. Endpoints in general should not be doing this.
	  This patch makes it so that Asterisk will simply hang up a call
	  if it sends a 200 OK at this point. So far this odd behavior for
	  endpoints has only been observed in tests which involved manually
	  created SIP transactions in SIPp. (closes issue ASTERISK-22424)
	  Reported by: Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/2827/ ........ Merged
	  revisions 399939 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399962 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399976 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-27 17:11 +0000 [r399938]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c,
	  /: astobj2: Remove OBJ_CONTINUE support. OBJ_CONTINUE was a
	  strange feature that came into the world under suspicious
	  circumstances to support an abuse of the ao2_container by
	  chan_iax2. Since chan_iax2 no longer uses OBJ_CONTINUE, it is
	  safe to remove it. The simplified code should help performance
	  slightly and make understanding the code easier. Review:
	  https://reviewboard.asterisk.org/r/2887/ ........ Merged
	  revisions 399937 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-27 14:35 +0000 [r399925]  Mark Michelson <mmichelson@digium.com>

	* /, bridges/bridge_native_rtp.c: Fix refleaks of ast_rtp_instance
	  structures. These refleaks were causing bridged calls not to
	  close their RTP ports. Thus a call would leave open 4 ports (RTP
	  for party A, RTCP for party A, RTP for party B, and RTCP for
	  party B). This led to an eventual depletion of available RTP
	  ports. ........ Merged revisions 399924 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-27 14:08 +0000 [r399913]  Kinsey Moore <kmoore@digium.com>

	* tests/test_cel.c, main/cel.c, /, include/asterisk/cel.h: Restore
	  usefulness of the CEL Peer field This change makes the CEL peer
	  field useful again for BRIDGE_ENTER and BRIDGE_EXIT events and
	  fills the field with a comma-separated list of all channels in
	  the bridge other than the channel that is entering or exiting the
	  bridge. Review: https://reviewboard.asterisk.org/r/2840/ (closes
	  issue ASTERISK-22393) ........ Merged revisions 399912 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-26 18:51 +0000 [r399898]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h,
	  res/res_pjsip.exports.in, /, res/res_pjsip/security_events.c:
	  pjsip: race condition in registrar While handling a registration
	  request a race condition could occur if/when two+ clients
	  registered at the same time. This happened when one request
	  obtained a copy of the current contacts for an AOR and another
	  request did the same before the first request updated. Thus the
	  second would update and overwrite the first (or vice-versa
	  depending on which actually updated first). In the case of it
	  being the same contact two "add" events would be raised. pjsip
	  registration handling is now serialized to alleviate this issue.
	  (closes issue AST-1213) Reported by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/2860/ ........ Merged
	  revisions 399897 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-26 14:13 +0000 [r399875]  Rusty Newton <rnewton@digium.com>

	* /, apps/app_dial.c: Adding a few words to the Dial option 'r'
	  help text to clarify its tone argument description ........
	  Merged revisions 399874 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-25 20:38 +0000 [r399844]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c, channels/chan_dahdi.c, /: chan_dahdi: CLI
	  "core stop gracefully" has needless delay for PRI and SS7. The
	  PRI and SS7 link control threads are not stopped correctly when
	  the chan_dahdi.so module is unloaded. The link control threads
	  pri_dchannel() and ss7_linkset() are not awakened from a poll()
	  to cancel the thread. * Added a SIGURG signal after requesting
	  the thread cancel to break the link control thread poll()
	  immediately. For SS7 it was slightly worse, the link poll()
	  timeout would always be whatever was the last libss7 scheduled
	  event time used. If no libss7 scheduled event was pending, the
	  thread could run more often than necessary. * Set nextms to 60
	  seconds for the ss7_linkset() poll() if there is no other libss7
	  scheduled event. ........ Merged revisions 399818 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399834 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399842 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-25 19:43 +0000 [r399799]  Rusty Newton <rnewton@digium.com>

	* /, res/res_pjsip.c: Broke the build - Fixing XML DTD violation
	  added in r399782, missing <para> tags inside a <note> ........
	  Merged revisions 399798 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-25 19:29 +0000 [r399797]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_sip.c: chan_sip: Fix Realtime Peer Update
	  Problem When Un-registering And Expires Header In 200ok 1st Issue
	  When a realtime peer sends an un-REGISTER request, Asterisk
	  un-registers the peer but the database table record still has
	  regseconds and fullcontact for the peer. This results in calls
	  attempting to be routed to the peer which is no longer
	  registered. The expected behavior is to get busy/congested when
	  attempting to call an un-registered peer through the dialplan.
	  What was discovered is that we are clearing out the peer's
	  registration in the database in parse_register_contact() when
	  calling expire_register() but then upon returning from
	  parse_register_contact(), update_peer() is run which stores back
	  in the database table regseconds and fullcontact. 2nd Issue The
	  reporter pointed out that the 200 ok being returned by Asterisk
	  after un-registering a peer contains a Contact header with
	  ;expires= and the Expires header is not set to 0. This is
	  actually a regression. Tests were created for this second issue
	  (ASTERISK-22548). The tests have been reviewed and a Ship It! was
	  received on those tests. This patch does the following: * Do not
	  ignore the Expires header value even when it is set to 0. The
	  patch sets the pvt->expiry earlier on in the function so that it
	  is set properly and used. * If pvt->expiry is 0, do not call
	  update_peer since that means the peer has already been
	  un-registered and there is no need to update the database record
	  again since nothing has changed. (closes issue ASTERISK-22428)
	  Reported by: Ben Smithurst Tested by: Ben Smithurst, Michael L.
	  Young Patches:
	  asterisk-22428-rt-peer-update-and-expires-header.diff by Michael
	  L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2869/ ........ Merged
	  revisions 399794 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399795 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399796 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-25 18:38 +0000 [r399782]  Rusty Newton <rnewton@digium.com>

	* /, res/res_pjsip.c: Fixing documentation for the configOption
	  "external_media_address" of both Endpoints and Transports
	  Re-using some of Mark Michelson's text from an E-mail discussion
	  for: * Modifying synopsis for both options * Adding description
	  to both options * Changing name of "external_media_address" for
	  Endpoint configuration to "media_address" in anticipation of the
	  option name being changed. (As it is not really specific to
	  external destinations) (issue ASTERISK-22405) (closes issue
	  ASTERISK-22405) Reported by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/2850/ ........ Merged
	  revisions 399781 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-24 22:55 +0000 [r399737-399750]  Richard Mudgett <rmudgett@digium.com>

	* /, main/astobj2.c: astobj2: Made use OBJ_SEARCH_xxx identifiers
	  as field enum values internally. * Made ao2_unlink to protect
	  itself from stray OBJ_SEARCH_xxx values passed in. ........
	  Merged revisions 399749 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_iax2.c, /: chan_iax2: Prevent some needless
	  breaking of the native IAX2 bridge. * Clean up some twisted code
	  in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and
	  AST_CONTROL_SRCCHANGE to a list of frames to prevent the native
	  bridge loop from breaking. * Passing the
	  AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a
	  native IAX2 bridge. (issue ABE-2912) Review:
	  https://reviewboard.asterisk.org/r/2870/ ........ Merged
	  revisions 399697 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399708 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 For v12 and
	  above this is really just documentation until IAX2 native
	  bridging is restored. ........ Merged revisions 399736 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-24 19:22 +0000 [r399667-399696]  Matthew Jordan <mjordan@digium.com>

	* apps/app_queue.c, /: app_queue: Don't be quite so aggressive in
	  initializing the array We only need the first character. ........
	  Merged revisions 399695 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_queue.c, /: app_queue: Initialize array holding
	  MixMonitor exec options If the channel variable MONITOR_EXEC is
	  set, app_queue will pass the specified execution parameters to
	  the MixMonitor application when a queue is recorded. If that
	  channel variable is not set, the buffer that holds the escaped
	  value was not being initialized to NULL, and so would be passed
	  to the MixMonitor application with garbage. Hilarity ensued as
	  app_mixmonitor attempted to execute gobeldy-gook. ........ Merged
	  revisions 399681 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/stasis_bridges.c, tests/test_cdr.c, main/cdr.c, /: Fix a
	  performance problem CDRs There is a large performance price
	  currently in the CDR engine. We currently perform two
	  ao2_callback calls on a container that has an entry for every
	  channel in the system. This is done to create matching pairs
	  between channels in a bridge. As such, the portion of the CDR
	  logic that this patch deals with is how we make pairings when a
	  channel enters a mixing bridge. In general, when a channel enters
	  such a bridge, we need to do two things: (1) Figure out if anyone
	  in the bridge can be this channel's Party B. (2) Make pairings
	  with every other channel in the bridge that is not already our
	  Party B. This is a two step process. In the first step, we look
	  through everyone in the bridge and see if they can be our Party B
	  (single_state_process_bridge_enter). If they can - yay! We mark
	  our CDR as having gotten a Party B. If not, we keep searching. If
	  we don't find one, we wait until someone joins who can be our
	  Party B. Step 2 is where we changed the logic
	  (handle_bridge_pairings and bridge_candidate_process).
	  Previously, we would first find candidates - those channels in
	  the bridge with us - from the active_cdrs_by_channel container.
	  Because a channel could be a candidate if it was Party B to an
	  item in the container, the code implemented multiple
	  ao2_container callbacks to get all the candidates. We also had to
	  store them in another container with some other meta information.
	  This was rather complex and costly, particularly if you have 300
	  Local channels (600 channels!) going at once. Luckily, none of it
	  is needed: when a channel enters a bridge (which is when we're
	  figuring all this stuff out), the bridge snapshot tells us the
	  unique IDs of everyone already in the bridge. All we need to do
	  is: For all channels in the bridge: If the channel is us or our
	  Party B that we got in step 1, skip it Compare us and the
	  candidate to figure out who is Party A (based on some specific
	  rules) If we are Party A: Make a new CDR for us, append it to our
	  chain, and set the candidate as Party B If they are Party A: If
	  they don't have a Party B: Make a new CDR for them, append us to
	  their chain, and us as Party B Otherwise: Copy us over as Party B
	  on their existing CDR. This patch does that. Because we now use
	  channel unique IDs to find the candidates during bridging,
	  active_cdrs_by_channel now looks up things using uniqueid instead
	  of channel name. This makes the more complex code simpler; it
	  does, however, have the drawback that dialplan applications and
	  functions will be slightly slower as they have to iterate through
	  the container looking for the CDR by name. That's a small price
	  to pay however as the bridging code will be called a lot more
	  often. This patch also does two other minor changes: (1) It
	  reduces the container size of the channels in a bridge snapshot
	  to 1. In order to be predictable for multi-party bridges, the
	  order of the channels in the container must be stable; that is,
	  it must always devolve to a linked list. (2) CDRs and the
	  multi-party test was updated to show the relationship between two
	  dialed channels. You still want to know if they talked -
	  previously, dialed channels were always ignored, which is wrong
	  when they have managed to get a Party B. (closes issue
	  ASTERISK-22488) Reported by: Richard Mudgett Review:
	  https://reviewboard.asterisk.org/r/2861/ ........ Merged
	  revisions 399666 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-23 12:03 +0000 [r399625]  Joshua Colp <jcolp@digium.com>

	* res/res_pjsip.c, res/res_pjsip_session.c, /: Fix crash in
	  res_pjsip on load if error occurs, and prevent unloading of
	  res_pjsip and res_pjsip_session. During load time in res_pjsip if
	  an error occurred the operation would attempt to rollback all
	  operations done during load. This is not permitted by PJSIP as it
	  will assert if the operation has not been done. This fix changes
	  the code so it will only rollback what has been initialized
	  already. Further changes also prevent res_pjsip and
	  res_pjsip_session from being unloaded. This is due to limitations
	  within PJSIP itself. The library environment can only be changed
	  to a certain extent and does not provide the ability, currently,
	  to deinitialize certain required functionality. (closes issue
	  ASTERISK-22474) Reported by: Corey Farrell ........ Merged
	  revisions 399624 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-21 04:49 +0000 [r399578-399608]  Richard Mudgett <rmudgett@digium.com>

	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix ref leaks in
	  ast_rtcp_read(). Moved rtcp_report RAII_VAR declaration into the
	  loop so it is unref'ed after every loop. Moved message_blob to
	  loop and switched it to a regular variable. The regular variable
	  was used since message_blob is used in a very contained way.
	  (closes issue ASTERISK-22565) Reported by: Corey Farrell Patches:
	  rtcp_report-leak.patch (license #5909) patch uploaded by Corey
	  Farrell Tested by: Corey Farrell ........ Merged revisions 399607
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/media_index.c: media_index: Fix
	  process_description_file() memory leak of file_id_persist.
	  ........ Merged revisions 399596 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/features_config.c: features_config: Fix config ref leak
	  of parkinglots. This leak happend for just about every channel
	  created. ........ Merged revisions 399585 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, apps/app_queue.c: app_queue: Fix json blob ref leak. The json
	  ref from queue_member_blob_create() was never released. ........
	  Merged revisions 399583 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/json.c, /: json: Make it obvious that ast_json_unref() is
	  NULL safe. It looked like the safety check was done after the
	  NULL pointer was used. ........ Merged revisions 399576 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-20 22:44 +0000 [r399566]  Kinsey Moore <kmoore@digium.com>

	* main/config_options.c, /: Ensure global types in the config
	  framework are initialized If a config object was allocated but
	  one of its global objects was never encountered, then the global
	  object's defaults were never applied. Ensure that global objects
	  are initialized properly upon allocation instead of on
	  configuration. Review: https://reviewboard.asterisk.org/r/2866/
	  ........ Merged revisions 399564 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399565 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-20 22:06 +0000 [r399554]  Jonathan Rose <jrose@digium.com>

	* main/dial.c, /: originate/call forwarding: Fix a crash when
	  forwarding a call from originate (closes issue ASTERISK-22487)
	  Reported by: David M. Lee Review:
	  https://reviewboard.asterisk.org/r/2868/ ........ Merged
	  revisions 399553 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-20 16:18 +0000 [r399533]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_pjsip.c: Add a missing session supplement
	  unregistration in chan_pjsip for ACKs. (closes issue
	  ASTERISK-22453) Reported by: Corey Farrell Patches:
	  chan_pjsip_session_unregister_supplement.patch uploaded by Corey
	  Farrell (license 5909) ........ Merged revisions 399531 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-20 14:26 +0000 [r399515]  Kevin Harwell <kharwell@digium.com>

	* /, main/logger.c: Fix memory leak in logger. Fixed a memory leak
	  discovered in the logger where a temporary string buffer was not
	  being freed. (closes issue ASTERISK-22540) Reported by: John
	  Hardin ........ Merged revisions 399513 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399514 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-19 23:20 +0000 [r399503]  Richard Mudgett <rmudgett@digium.com>

	* /, main/optional_api.c: optional_api: Make always use the
	  standard malloc functions even with MALLOC_DEBUG. ........ Merged
	  revisions 399501 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-19 17:01 +0000 [r399459]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: chan_sip: Make direct media reinvites for
	  T38 put Asterisk in the media path Prior to this patch, Asterisk
	  would incorrectly use the previous endpoint addresses in SDP in
	  spite of providing its own port. T38 is never meant to be done
	  through directmedia and Asterisk should always be in the media
	  path for these streams. (closes issue ASTERISK-17273) Reported
	  by: Kevin Stewart (closes issue ASTERISK-18706) Reported by:
	  Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/
	  ........ Merged revisions 399456 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399457 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399458 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-18 20:04 +0000 [r399405]  Kinsey Moore <kmoore@digium.com>

	* /, main/abstract_jb.c: Fix jitter buffer log file creation This
	  adjusts '/'-to-'#' replacement to replace all instances of '/'
	  instead of just the first to ensure that the jitter buffer log
	  file gets the correct name as per Richard Kenner's suggestion.
	  (closes issue ASTERISK-21036) Reported by: Richard Kenner
	  ........ Merged revisions 399402 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399403 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399404 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-18 17:23 +0000 [r399368-399378]  Matthew Jordan <mjordan@digium.com>

	* /, build_tools/prep_tarball: Update prep_tarball with new
	  documentation files on the Asterisk wiki This will now pull both
	  a command reference for the version being prepared, as well as an
	  Admin Guide that applies to all versions of Asterisk. (issue
	  ASTERISK-22439) Reported by: Olle Johansson ........ Merged
	  revisions 399351 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399373 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399376 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when
	  a timing module isn't loaded If bridge_softmix fails to be
	  created because no timing source is present in Asterisk, this
	  will currently fail gracefully but with (most likely) a generic
	  error message by whatever module tried to create the softmix
	  bridge. This patch adds a more explicit warning so you can
	  actually diagnose and fix the problem. Review:
	  https://reviewboard.asterisk.org/r/2857/ ........ Merged
	  revisions 399353 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399365 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-18 17:15 +0000 [r399352]  Richard Mudgett <rmudgett@digium.com>

	* main/config_options.c: Make config framework able to reload
	  module configs with multiple config files. The config framework
	  is supposed to be able to load configs that come from multiple
	  config files. The principle example is chan_sip's sip.conf and
	  users.conf. Unfortunately, it only does this correctly on initial
	  load. This patch causes the module's config to be reloaded
	  entirely if any of the config files change. (closes issue
	  ASTERISK-22009) Reported by: Richard Mudgett Review:
	  https://reviewboard.asterisk.org/r/2859/

2013-09-18 14:56 +0000 [r399340]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_messaging.c, /: res_pjsip_messaging: Register
	  message technology as pjsip pjsip's message technology was being
	  registered as 'sip', which was causing it to not load due it
	  conflicting with chan_sip's registered 'sip' technology for
	  messaging. It now registers as 'pjsip'. However, due to this
	  change the "to" field for outgoing pjsip messages need to be
	  prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to
	  res_pjsip_messaging will automatically have their "to" fields
	  altered in order to accommodate the change. Outgoing messages
	  also handle changing it back to 'sip' before being sent so the
	  pjsip library will properly handle it. (closes issue
	  ASTERISK-22445) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2833/ ........ Merged
	  revisions 399339 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-18 00:13 +0000 [r399295]  Michael L. Young <elgueromexicano@gmail.com>

	* /, main/features_config.c: Fix Segfault In features-config.c When
	  Application Has No Arguments Some applications do not require
	  arguments. Therefore, when parsing application maps in
	  features.conf, it is possible that app_data will be set to NULL.
	  * This patch sets app_data to "" if it is NULL. Review:
	  https://reviewboard.asterisk.org/r/2804 ........ Merged revisions
	  399294 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-17 23:10 +0000 [r399284]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_sdp_rtp.c, res/res_pjsip/pjsip_configuration.c,
	  res/res_pjsip_t38.c, include/asterisk/res_pjsip.h, /: Change the
	  "external_media_address" PJSIP endpoint option to
	  "media_address". The endpoint option does not apply to
	  communication with external entities. Rather, the option is
	  applied to all communications with the endpoint. The
	  external_media_address transport configuration option may
	  override the endpoint option if it turns out that we are going to
	  be communicating with an external entity. Two things of note: 1)
	  I have not updated the XML documentation. This is being taken
	  care of by Rusty as part of his work on issue ASTERISK-22405 2)
	  This commit is likely to cause testsuite failures since there are
	  tests that use the external_media_address endpoint option, and
	  they will need to be changed over. Well, I'm planning to get that
	  updated ASAP after this commit. (closes issue ASTERISK-22528)
	  reported by Rusty Newton ........ Merged revisions 399283 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-17 18:44 +0000 [r399269]  Kevin Harwell <kharwell@digium.com>

	* main/logger.c, main/asterisk.c, /: Remote console: more output
	  discrepancies The remote console continued to have issues with
	  its output. In this case CLI command output would either not show
	  up (if verbose level = 0) or would contain verbose prefixes (if
	  verbose level > 0) once log messages were sent to the remote
	  console. The fix now now adds verbose prefix data to all new
	  lines contained in a verbose log string. (closes issue
	  ASTERISK-22450) Reported by: David Brillert (closes issue
	  AST-1193) Reported by: Guenther Kelleter Review:
	  https://reviewboard.asterisk.org/r/2825/ ........ Merged
	  revisions 399267 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399268 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-17 17:55 +0000 [r399258]  Richard Mudgett <rmudgett@digium.com>

	* /, include/asterisk/features_config.h: Fix doxygen to use correct
	  units of features.conf options. ........ Merged revisions 399257
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-17 17:10 +0000 [r399238-399248]  Mark Michelson <mmichelson@digium.com>

	* main/bridge_basic.c, main/features_config.c, /: Fix other
	  timeouts (atxferloopdelay and atxfernoanswertimeout) to use
	  seconds instead of milliseconds. Thanks to Richard Mudgett for
	  pointing this out. ........ Merged revisions 399247 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/features_config.c, /, include/asterisk/features_config.h,
	  main/bridge_basic.c: Switch transferdigittimeout to be configured
	  as seconds instead of milliseconds. This was an unintentional
	  consequence of the update of features.conf to use the config
	  framework in Asterisk 12. Thanks to Marco Signorini on the
	  Asterisk developers list for pointing out the problem. ........
	  Merged revisions 399237 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-17 14:58 +0000 [r399226]  Kevin Harwell <kharwell@digium.com>

	* apps/confbridge/conf_state_multi_marked.c, /: Confbridge: empty
	  conference not being torn down Confbridge would not properly tear
	  down an empty conference bridge when all users were kicked via
	  end_marked=yes and at least one user was also set to wait_marked.
	  This occurred because while end_marked users were being kicked
	  and at least one was also set to wait_marked then the leave
	  wait_marked handler would be called on that user, but there would
	  be no waiting user (still considered active). The waiting users
	  would decrement and now be negative. The conference would remain,
	  but be put into an inactive state. The solution was to move from
	  the active list to the wait list, those users with wait_marked
	  set right before kicking. This allows both the active and wait
	  users to decrement correctly and the confbridge to tear down
	  properly. A crashed also occurred when trying to list the
	  specific conference from the CLI. This happened because the
	  conference specified was invalid. Since the conference properly
	  tears down now there is no way to reference it thus alleviating
	  the crash as well. (closes issue ASTERISK-21859) Reported by:
	  Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/
	  ........ Merged revisions 399222 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399225 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-16 18:36 +0000 [r399161-399208]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_ari_model.c, /: Fix module load errors for
	  test_ari_model.so. You cannot use a function pointer variable
	  with an external function from another dynamically loaded module
	  because data variables are always resolved even with RTLD_LAZY. *
	  Added wrapper functions for ast_ari_validate_int() and
	  ast_ari_validate_string() to use instead for the function pointer
	  variable. (closes issue ASTERISK-22457) Reported by: David M. Lee
	  ........ Merged revisions 399207 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_speech_utils.c, /, res/res_speech.exports.in:
	  app_speech_utils: Fix unresolved symbol ast_speech_get_setting().
	  Fixes regression introduced by -r374096. * Made
	  res_speech.export.in export ast_* symbols instead of specific
	  functions. * Made app_speech_utils.c declare that it is dependent
	  upon res_speech. (issue ASTERISK-17136) Reported by: Richard
	  Kenner ........ Merged revisions 399197 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_iax2.c, /: chan_iax2: Fix saving the wrong expiry
	  time in astdb. When a new IAX2 client registers, the astdb
	  database is updated with the value of minregexpire defined in
	  iax.conf instead of using the expiry time that is provided by the
	  client. The provided expiry time of the client is updated after
	  inserting the astdb entry. As a consequence, restarting or
	  reloading asterisk creates clients whose registration may expire
	  before they reregister. The clients are therefore unavailable
	  after minregexpire seconds until they reregister. * Move updating
	  of the expiry time to before inserting into the astdb. (closes
	  issue ASTERISK-22504) Reported by: Stefan Wachtler Patches:
	  chan_iax2.c.patch (license #6533) patch uploaded by Stefan
	  Wachtler ........ Merged revisions 399158 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399159 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399160 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-16 02:37 +0000 [r399147]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /: Filter internal channels out of bridge enter/leave
	  message handling Some channels exist merely as an implementation
	  detail in Asterisk, such as ConfBridge's announcer/recorder
	  channels. These channels should never be exposed to the outside
	  world, or to interfaces that report on Asterisk. We already
	  filter out such channels in snapshot processing; however, we
	  failed to filter out bridge related messages that involved these
	  channels. This patch filters out bridge related messages that are
	  for such channels. This prevents a spurious WARNING message from
	  being displayed when those channels move in and out of bridges.
	  ........ Merged revisions 399146 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-13 22:19 +0000 [r399138]  Richard Mudgett <rmudgett@digium.com>

	* res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
	  include/asterisk/features.h, main/channel.c,
	  res/parking/parking_tests.c, include/asterisk/bridge_channel.h,
	  main/features.c, tests/test_cel.c, main/bridge_channel.c,
	  tests/test_cdr.c, apps/confbridge/conf_chan_announce.c,
	  include/asterisk/bridge.h, res/res_pjsip_refer.c, /,
	  channels/chan_sip.c, res/stasis/control.c, main/bridge.c,
	  main/bridge_basic.c, main/core_unreal.c,
	  res/parking/parking_applications.c, main/core_local.c: Restore
	  Dial, Queue, and FollowMe 'I' option support. The Dial, Queue,
	  and FollowMe applications need to inhibit the bridging initial
	  connected line exchange in order to support the 'I' option. *
	  Replaced the pass_reference flag on ast_bridge_join() with a
	  flags parameter to pass other flags defined by enum
	  ast_bridge_join_flags. * Replaced the independent flag on
	  ast_bridge_impart() with a flags parameter to pass other flags
	  defined by enum ast_bridge_impart_flags. * Since the Dial, Queue,
	  and FollowMe applications are now the only callers of
	  ast_bridge_call() and ast_bridge_call_with_flags(), changed the
	  calling contract to require the initial COLP exchange to already
	  have been done by the caller. * Made all callers of
	  ast_bridge_impart() check the return value. It is important. As a
	  precaution, I also made the compiler complain now if it is not
	  checked. * Did some cleanup in parking_tests.c as a result of
	  checking the ast_bridge_impart() return value. An independent,
	  but associated change is: * Reduce stack usage in
	  ast_indicate_data() and add a dropping redundant connected line
	  verbose message. (closes issue ASTERISK-22072) Reported by:
	  Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/
	  ........ Merged revisions 399136 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-13 20:55 +0000 [r399101]  David M. Lee <dlee@digium.com>

	* /, main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not
	  defined. If MALLOC_DEBUG is enabled, then the debug destructor
	  for the container is used, which would erroneously write to
	  /tmp/refs. This patch only uses the debug destructor if ref_debug
	  is used. (closes issue ASTERISK-22536) ........ Merged revisions
	  399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 399099 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399100 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-13 14:50 +0000 [r399082-399084]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c,
	  include/asterisk/res_pjsip.h, res/res_pjsip.exports.in, /: Create
	  more accurate Contact headers for dialogs when we are the UAS.
	  (closes issue AST-1207) reported by John Bigelow Review:
	  https://reviewboard.asterisk.org/r/2842 ........ Merged revisions
	  399083 from http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip/config_auth.c, /,
	  res/res_pjsip_outbound_authenticator_digest.c,
	  res/res_pjsip_authenticator_digest.c: Change how realms are
	  handled for outbound authentication. With this change, if no
	  realm is specified in an outbound auth section, then we will
	  simply match the realm that was present in the 401/407 challenge.
	  (closes issue ASTERISK-22471) Reported by George Joseph (closes
	  issue ASTERISK-22386) Reported by Rusty Newton Patches:
	  outbound_auth_realm_v4.patch uploaded by George Joseph (License
	  #6322) ........ Merged revisions 399059 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-13 14:43 +0000 [r399080-399081]  David M. Lee <dlee@digium.com>

	* /: Recorded merge of revisions 399035,399049 from
	  http://svn.asterisk.org/svn/asterisk/branches/12 These were lost
	  in r399071

	* /: Put merge tracking for r399039 back.

2013-09-13 14:27 +0000 [r399071]  Rusty Newton <rnewton@digium.com>

	* /, res/res_pjsip_endpoint_identifier_ip.c: Broke the build!
	  Forgot para tags within my description.
	  https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304
	  ........ Merged revisions 399064 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-13 14:22 +0000 [r399042-399051]  David M. Lee <dlee@digium.com>

	* res/res_pjsip_log_forwarder.c (added), res/res_pjsip_logger.c,
	  res/res_rtp_asterisk.c, /: res_pjsip: Forward PJSIP logging to
	  Asterisk logging This patch uses PJSIP's pj_log_set_log_func() to
	  forward PJSIP's log messages to Asterisk's logger. This is done
	  in a new module: res_pjsip_log_forwarder.so. This patch sets
	  defaultenabled on the existing res_pjsip_logger.so to no, since
	  logging every SIP packet seems a bit odd to do by default, and is
	  (hopefully) less necessary with regular PJSIP logging. It also
	  removes res_rtp_asterisk's disabling of PJSIP logging. (closes
	  issue ASTERISK-22360) Reported by: Joshua Colp Review:
	  https://reviewboard.asterisk.org/r/2830/ ........ Merged
	  revisions 399049 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_http_websocket.c: ARI: Fix WebSocket response when
	  subprotocol isn't specified When I moved the ARI WebSocket from
	  /ws to /ari/events, I added code to allow a WebSocket to connect
	  without specifying the subprotocol if there's only one
	  subprotocol handler registered for the WebSocket. Naively, I
	  coded it to always respond with the subprotocol in use.
	  Unfortunately, according to RFC 6455, if the server's response
	  includes a subprotocol header field that "indicates the use of a
	  subprotocol that was not present in the client's handshake [...],
	  the client MUST _Fail the WebSocket Connection_.", emphasis
	  theirs. This patch correctly omits the Sec-WebSocket-Protocol if
	  one is not specified by the client. (closes issue ASTERISK-22441)
	  Review: https://reviewboard.asterisk.org/r/2828/ ........ Merged
	  revisions 399039 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-13 14:17 +0000 [r399036]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_meetme.c: Fix several crashes in MeetMeAdmin This
	  change ensures that MeetMeAdmin commands requiring a user
	  actually get a user and fixes another issue where an extra
	  dereference could occur for a last-entered user being ejected if
	  a user identifier was also provided. (closes issue
	  ASTERISK-21907) Reported by: Alex Epshteyn Review:
	  https://reviewboard.asterisk.org/r/2844/ ........ Merged
	  revisions 399033 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 399034 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 399035 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-13 13:28 +0000 [r399032]  Rusty Newton <rnewton@digium.com>

	* /, res/res_pjsip_endpoint_identifier_ip.c: 'identify'
	  configObject doesn't have a synopsis Add a straightforward
	  synopsis and description to the identify config object in XML
	  documentation. (issue ASTERISK-22311) (closes issue
	  ASTERISK-22311) Reported By: Rusty Newton ........ Merged
	  revisions 399031 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-12 23:42 +0000 [r399020-399022]  Richard Mudgett <rmudgett@digium.com>

	* /, main/bridge.c: CLI bridge: Fix "bridge destroy <id>" and
	  "bridge kick <id> <chan>" tab completion. These two commands must
	  deal with the live bridges container for tab completion and not
	  the stasis cache. ........ Merged revisions 399021 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/bridge.c, /: astobj2: Register the bridges container for
	  debug inspection. ........ Merged revisions 399019 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-12 23:23 +0000 [r399018]  Rusty Newton <rnewton@digium.com>

	* /, res/res_pjsip_acl.c: Documentation fix and improvements to XML
	  configuration help res_pjsip_acl * One bug fix. Made the synopsis
	  for "type" to accurate. * changing the usage of "IP-domains" to
	  "IP addresses" * clarifying the usage for the options, by adding
	  a relevant description for each * modified other areas of the XML
	  help for clarity, such as the module description and a few
	  synopsis changes here and there. See the patch. (issue
	  ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty
	  Newton Review: https://reviewboard.asterisk.org/r/2823/ ........
	  Merged revisions 399017 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-12 20:27 +0000 [r399006]  Jonathan Rose <jrose@digium.com>

	* channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
	  Revert r398835 due to failing tests involving originate (issue
	  ASTERISK-22424) Reported by: Jonathan Rose ........ Merged
	  revisions 398977 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398986 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398991 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-12 16:44 +0000 [r398939]  Richard Mudgett <rmudgett@digium.com>

	* main/core_unreal.c, /: core_local: Fix memory corruption race
	  condition. The masquerade super test is failing on v12 with high
	  fence violations and crashing. The fence violations are showing
	  that party id allocated memory strings are somehow getting
	  corrupted in the bridge_reconfigured_connected_line_update()
	  function. The invalid string values happen to be the freed memory
	  fill pattern. After much puzzling, I deduced that the
	  bridge_reconfigured_connected_line_update() is copying a string
	  out of the source channel's caller party id struct just as
	  another thread is updating it with a new value. The copying
	  thread is using the old string pointer being freed by the
	  updating thread. A search of the code found the
	  unreal_colp_redirect_indicate() routine updating the caller party
	  id's without holding the channel lock. A latent bug in v1.8 and
	  v11 hatched in v12 because of the bridging and connected line
	  changes. :) (issue ASTERISK-22221) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/2839/ ........ Merged
	  revisions 398938 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-12 15:23 +0000 [r398928]  David M. Lee <dlee@digium.com>

	* res/res_pjsip.c, /: Fix symbol collision with pjsua. We shouldn't
	  be exporting any symbols that start with pjsip_. ........ Merged
	  revisions 398927 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-12 00:04 +0000 [r398883-398887]  Rusty Newton <rnewton@digium.com>

	* /, apps/app_queue.c: 'queue add member' help text correction You
	  are adding dial strings to the queue, not channels. An aribitrary
	  string could be used, but you are typically referencing a
	  channel. Correcting the command help text. (issue ASTERISK-22263)
	  (closes issue ASTERISK-22263) Reported By: Rusty Newton ........
	  Merged revisions 398884 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398885 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398886 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* configs/chan_dahdi.conf.sample, /: Documentation fix -
	  waitfordialtone is not boolean, it's time in milliseconds
	  Changing text in chan_dahdi.conf sample to be accurate. (issue
	  ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
	  Malcolm Davenport ........ Merged revisions 398880 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398881 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398882 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-11 20:03 +0000 [r398838]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip:
	  Reject calls without prior SDP on 200 OK If we receive a 200 OK
	  without SDP, we will now check to see if the remote address has
	  been established for that channel's RTP session and if the to tag
	  for that channel has changed from the most recent to tag in a
	  response less than 200. If either a change has been made since
	  the last to-tag was received or the remote address is unset, then
	  we will drop the call. (closes issue ASTERISK-22424) Reported by:
	  Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/2827/diff/#index_header
	  ........ Merged revisions 398835 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398836 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398837 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-11 18:03 +0000 [r398822]  Russell Bryant <russell@russellbryant.com>

	* configs/confbridge.conf.sample, /: Fix typo in
	  confbridge.conf.sample The denoise filter requires func_speex,
	  not codec_speex. Fix this in the description of the denoise=yes
	  option in confbridge.conf. ........ Merged revisions 398820 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398821 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-11 14:23 +0000 [r398808]  Kevin Harwell <kharwell@digium.com>

	* res/res_pjsip_caller_id.c, channels/chan_pjsip.c, /: pjsip:
	  reinvite for connected line updates occurs when it should not
	  Connected line updates are now only sent out if an actual update
	  needs to occur. This happens under the following conditions: 1.
	  The endpoint we are sending to is trusted. 2. Either a
	  P-Asserted-Identity or Remote Party-ID header needs to be
	  added/sent. 3. The connected id's number and name are valid. Also
	  added an SDP when an update is sent out. (closes issue AST-1212)
	  Reported by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/2831/ ........ Merged
	  revisions 398806 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-10 18:05 +0000 [r398760]  Richard Mudgett <rmudgett@digium.com>

	* main/event.c, res/res_musiconhold.c, main/indications.c,
	  main/asterisk.c, main/xmldoc.c, main/cli.c, /,
	  funcs/func_dialgroup.c, main/heap.c,
	  res/res_pjsip/pjsip_configuration.c: Fix incorrect usages of
	  ast_realloc(). There are several locations in the code base where
	  this is done: buf = ast_realloc(buf, new_size); This is going to
	  leak the original buf contents if the realloc fails. Review:
	  https://reviewboard.asterisk.org/r/2832/ ........ Merged
	  revisions 398757 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398758 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398759 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-10 17:50 +0000 [r398751-398755]  David M. Lee <dlee@digium.com>

	* utils/check_expr.c, /: Fixed utils directory breakage from
	  r398748, this time with extra hate. ........ Merged revisions
	  398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 398753 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398754 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* utils/check_expr.c, /, utils/ael_main.c, utils/conf2ael.c: Fixed
	  utils directory breakage from r398648 ........ Merged revisions
	  398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 398749 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398750 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-09 23:29 +0000 [r398732]  Richard Mudgett <rmudgett@digium.com>

	* main/astmm.c, /: MALLOC_DEBUG: Change fence magic number to be
	  completely different from the freed magic number. Race conditions
	  between freeing a nul terminated string and ast_strdup()'ing it
	  are more likely to be detected if the fence and freed magic
	  numbers are completely different. ........ Merged revisions
	  398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 398721 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398726 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-09 22:00 +0000 [r398695]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_endpoint_identifier_ip.c, /: Add extra debugging to
	  res_pjsip_endpoint_identifier_ip ........ Merged revisions 398694
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-09 20:13 +0000 [r398641-398652]  David M. Lee <dlee@digium.com>

	* /, main/utils.c, include/asterisk/lock.h, main/lock.c: Fix
	  DEBUG_THREADS when lock is acquired in __constructor__ This patch
	  fixes some long-standing bugs in debug threads that were
	  exacerbated with recent Optional API work in Asterisk 12. With
	  debug threads enabled, on some systems, there's a lock ordering
	  problem between our mutex and glibc's mutex protecting its module
	  list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
	  thread, the module list will be locked before acquiring our
	  mutex. In another thread, our mutex will be locked before locking
	  the module list (which happens in the depths of calling
	  backtrace()). This patch fixes this issue by moving backtrace()
	  calls outside of critical sections that have the mutex acquired.
	  The bigger change was to reentrancy tracking for
	  ast_cond_{timed,}wait, which wrongly assumed that waiting on the
	  mutex was equivalent to a single unlock (it actually suspends all
	  recursive locks on the mutex). (closes issue ASTERISK-22455)
	  Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged
	  revisions 398648 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398649 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398651 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/ari/resource_channels.h, /, rest-api/api-docs/channels.json:
	  Multiple revisions 398638-398639 ........ r398638 | dlee |
	  2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line Added note
	  about expected behavior of originate ........ r398639 | dlee |
	  2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line Added note
	  about expected behavior of originate (the rest of the commit)
	  ........ Merged revisions 398638-398639 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-08 23:30 +0000 [r398629]  Matthew Jordan <mjordan@digium.com>

	* tests/test_cdr.c, /: Update CDR Unit tests to reflect container
	  changes in r398579 When a channel joins a multi-party bridge, the
	  ordering of the CDRs that is created is determined by the
	  ordering of the channels who happen to be in that bridge. When
	  r398579 changed the number of buckets in the container to
	  something sensible, it changed the ordering that the CDRs was
	  created in, causing one of the multiparty tests to fail. This
	  fixes the test with the now expected ordering. ........ Merged
	  revisions 398628 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-07 01:03 +0000 [r398603-398620]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_xmpp.c: Prevent XMPP timeout on blank responses
	  Sometimes the Google Voice servers have a bad habit of sending
	  out 1 byte replies to the xmpp resource. When a blank 1 byte
	  reply is received from the socket the buffer attempts to wait
	  (endlessly) for the rest of the reply from google which
	  effectively blocks the socket and google voice calls will no
	  longer come into the server. This patch allows the xmpp module to
	  correctly detect empty packets and send out ping replies to
	  google. It also sets a socket timeout on the default socket which
	  prevents the xmpp socket from closing and preventing future
	  google voice calls from coming into the server. Furthermore
	  instead of sending an empty reply back to google we send a proper
	  xmpp ping reply back. This also adds several more socket
	  messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy
	  Review: https://reviewboard.asterisk.org/r/2771 Patches:
	  xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........
	  Merged revisions 398618 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398619 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_xmpp.c, res/res_jabber.c: Multiple revisions
	  398558,398577 ........ r398558 | kmoore | 2013-09-06 14:28:16
	  -0500 (Fri, 06 Sep 2013) | 17 lines Fix Jabber/XMPP distributed
	  MWI The mailbox and context are swapped on the receiving end for
	  all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and
	  all more recent versions. This swaps those values to be correct
	  when publishing to the internal event system from Jabber/XMPP
	  distributed MWI state. (closes issue ASTERISK-22435) Reported by:
	  abelbeck Tested by: Michael Keuter Patches:
	  asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
	  abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
	  uploaded by abelbeck ........ Merged revisions 398523 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) |
	  10 lines Commit the remainder of r398523 This is a missing part
	  of the commit in revision 398523 that corrects the name of a
	  variable. (issue ASTERISK-22435) ........ Merged revisions 398576
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 398558,398577 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398580 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-06 21:17 +0000 [r398557-398583]  Richard Mudgett <rmudgett@digium.com>

	* main/cdr.c, /: cdr: Change the number of container buckets to be
	  similar to the channels container. * Fix the temporary cdr
	  candidate containers to use a prime number of buckets. ........
	  Merged revisions 398579 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/core_local.c, /: core_local: Fix LocalOptimizationBegin AMI
	  event missing Source channel snapshot. * Fix the
	  LocalOptimizationBegin AMI event by eliminating an artificial
	  buffer size limitation that is too small anyway. ........ Merged
	  revisions 398572 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: cdr: Fix some ref leaks. * Added missing
	  unregister of the cdr container in cdr_engine_shutdown(). * Fixed
	  ref leak in off nominal path of cdr_object_alloc(). * Removed
	  some unnecessary NULL checks in cdr_object_dtor(). ........
	  Merged revisions 398562 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/astobj2.h, main/cel.c, main/features_config.c,
	  apps/app_agent_pool.c, main/cdr.c, main/udptl.c, /,
	  main/parking.c, main/stasis_config.c: astobj2: Add warn unused
	  attribute to some functions. * Fixed resulting warnings with
	  improper use of ao2_global_obj_replace(). * Made a couple uses of
	  ao2_global_obj_replace_unref(x, NULL) into the equivalent and
	  more appropriate ao2_global_obj_release() call. ........ Merged
	  revisions 398533 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-06 18:53 +0000 [r398512-398522]  Kinsey Moore <kmoore@digium.com>

	* main/http.c, /, res/stasis/app.c: Fix build warnings When
	  AST_DEVMODE is not defined, ast_asserts are not compiled into the
	  binary. In some cases, this means variables are not referenced or
	  are set but unused which causes warnings to show up. (closes
	  issue ASTERISK-22446) Reported by: Jason Parker (qwell) ........
	  Merged revisions 398521 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_h323.c: Fix chan_h323 compilation This fixes the
	  things in chan_h323 that were missed or ignored in the great
	  channel opaquification and gets chan_h323 back into a compiling
	  state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov
	  Patches: chan_h323.patch uploaded by Dmitry Melekhov ........
	  Merged revisions 398510 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398511 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-05 21:48 +0000 [r398384-398499]  Richard Mudgett <rmudgett@digium.com>

	* /, main/astobj2.c: astobj2: Only define ao2_bt() once. * Make
	  ao2_bt() not use single char variable names. * Fix ao2_bt()
	  formatting. ........ Merged revisions 398498 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_iax2.c, /: chan_iax2: Reduce indentation in
	  __attempt_transmit(). * Reduce indentation in
	  __attempt_transmit(). * Don't update the static last error time
	  variable every time in __schedule_action() and socket_read().
	  ........ Merged revisions 398456 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398457 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398458 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_iax2.c, /: chan_iax2: Fix stray reference to worker
	  thread idle_list. * Fix stray reference to idle_list in
	  cleanup_thread_list(). This may be the reason for the note in
	  iax2_process_thread() about threads not being removed from the
	  task lists. * Move cleanup_thread_list(&idle_list) to after the
	  other lists are cleaned up. ........ Merged revisions 398416 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398417 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398418 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/chan_iax2.c, /: chan_iax2: Fix bridgecallno deadlock
	  avoidance. * Fix bridgecallno deadlock avoidance. When doing
	  deadlock avoidance, you need to retest the status of values for
	  each loop to see if you still need the lock for bridgecallno. *
	  As a safety check, after acquiring the bridgecallno lock you
	  should check if iaxs[bridgecallno] is NULL just like the current
	  callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
	  to after processing any deferred frames to ensure that the
	  iostate is IDLE when it is placed back into the idle list.
	  defer_full_frame() tries to ensure iax2_process_thread() wakes up
	  to process the frame. ........ Merged revisions 398379 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398380 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398381 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-05 14:10 +0000 [r398369]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip_outbound_registration.c: Clarify server_uri and
	  client_uri registration settings. Used some of Rusty's suggested
	  language plus also included more SIPesque descriptions of where
	  the URIs are actually used in an outgoing REGISTER. (closes issue
	  ASTERISK-22390) reported by Rusty Newton ........ Merged
	  revisions 398368 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-04 23:07 +0000 [r398304]  Richard Mudgett <rmudgett@digium.com>

	* channels/iax2/parser.c, /: chan_iax2: Add missing control frame
	  names to debug frame decode output. ........ Merged revisions
	  398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 398302 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398303 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-04 22:49 +0000 [r398300]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip_outbound_authenticator_digest.c: Give more
	  detail regarding failures to create request with auth
	  credentials. (issue ASTERISK-22386) ........ Merged revisions
	  398299 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-04 21:37 +0000 [r398284-398287]  Jonathan Rose <jrose@digium.com>

	* /, tests/test_voicemail_api.c: unit tests: test_voicemail_api
	  leaks stringfields from snapshots (closes issue ASTERISK-22414)
	  Reported by: Corey Farrell Patches:
	  test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
	  (license 5909) ........ Merged revisions 398285 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398286 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* apps/app_voicemail.c, /: app_voicemail: Fix leaking config
	  objects when msg_id doesn't match (issues ASTERISK-22414)
	  Reported by: Corey Farrell Patch:
	  test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
	  (license 5909) ........ Merged revisions 398281 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398283 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-04 16:03 +0000 [r398238]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: chan_misdn: Fix misdn debug output
	  printed with arbitrary verbose levels. Fix the misdn debug output
	  to remote consoles. chan_misdn uses ast_console_puts() which
	  doesn't know about verbose levels. Better to use ast_verbose()
	  instead. Without this patch the misdn debug messages are appended
	  to the verbose level which ever was set by the message sent to
	  the console before, i.e. any undefined level. (closes issue
	  AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch
	  (license #6372) patch uploaded by Guenther Kelleter ........
	  Merged revisions 398235 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398236 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398237 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-04 14:32 +0000 [r398227]  Kevin Harwell <kharwell@digium.com>

	* /, res/res_pjsip_outbound_registration.c: Debug messages for
	  pjsip outbound registration Added debug messages indicating that
	  an outbound registration attempt was made and it was successful
	  in pjsip. (closes issue ASTERISK-22388) Reported by: Rusty Newton
	  ........ Merged revisions 398226 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-03 20:28 +0000 [r398217]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323c/src/ooh245.c: Fix remote tcs sequence handling
	  on empty tcs received ........ Merged revisions 398214 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398215 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-03 18:09 +0000 [r398207]  Kinsey Moore <kmoore@digium.com>

	* res/res_pjsip_dtmf_info.c, /: Prevent a crash in
	  res_pjsip_dtmf_info.c This change makes sure that a content type
	  header exists before checking the contents of the header against
	  known SIP INFO DTMF content types. ........ Merged revisions
	  398206 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-03 17:19 +0000 [r398205]  David M. Lee <dlee@digium.com>

	* Makefile, /: Fixed 'make clean' for wiki docs ........ Merged
	  revisions 398198 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-09-03 14:29 +0000 [r398197]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, cel/cel_custom.c: Be a little more verbose when loading
	  cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
	  ........ Merged revisions 398167 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398168 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398196 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 20:58 +0000 [r398150]  David M. Lee <dlee@digium.com>

	* main/asterisk.c, include/asterisk/optional_api.h, /,
	  main/optional_api.c: Fix graceful shutdown crash. The cleanup
	  code for optional_api needs to happen after all of the optional
	  API users and providers have unused/unprovided. Unfortunately,
	  regsitering the atexit() handler at the beginning of main() isn't
	  soon enough, since module destructors run after that. ........
	  Merged revisions 398149 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 20:37 +0000 [r398148]  Rusty Newton <rnewton@digium.com>

	* /, configs/pjsip.conf.sample: New pjsip.conf.sample (issue
	  ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........
	  Merged revisions 398147 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 19:55 +0000 [r398124-398140]  Kevin Harwell <kharwell@digium.com>

	* /, res/res_pjsip_outbound_registration.c,
	  include/asterisk/sorcery.h, res/res_pjsip.c,
	  res/res_pjsip/config_transport.c, main/sorcery.c: Add a
	  reloadable option for sorcery type objects Some configuration
	  objects currently won't place nice if reloaded. Specifically, in
	  this case the pjsip transport objects. Now when registering an
	  object in sorcery one may specify that the object is allowed to
	  be reloaded or not. If the object is set to not reload then upon
	  reloading of the configuration the objects of that type will not
	  be reloaded. The initially loaded objects of that type however
	  will remain. While the transport objects will not longer be
	  reloaded it is still possible for a user to configure an endpoint
	  to an invalid transport. A couple of log messages were added to
	  help diagnose this problem if it occurs. (closes issue
	  ASTERISK-22382) Reported by: Rusty Newton (closes issue
	  ASTERISK-22384) Reported by: Rusty Newton Review:
	  https://reviewboard.asterisk.org/r/2807/ ........ Merged
	  revisions 398139 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/config.c, res/res_security_log.c, /, channels/chan_sip.c,
	  main/translate.c, main/named_acl.c, main/indications.c: Fix
	  various memory leaks main/config.c - cleanup cache fie includes
	  res/res_security_log.c - unregister logger level
	  channesl/chan_sip.c - cleanup io context and notify_types
	  main/translator.c - cleanup at shutdown main/named_acl.c -
	  cleanup cli commands main/indications.c -
	  ast_get_indication_tone() unref default_tone_zone if used (closes
	  issues ASTERISK-22378) Reported by: Corey Farrell Patches:
	  config_shutdown.patch uploaded by coreyfarrell (license 5909)
	  res_security_log.patch uploaded by coreyfarrell (license 5909)
	  chan_sip-11.patch uploaded by coreyfarrell (license 5909)
	  indications_refleak.patch uploaded by coreyfarrell (license 5909)
	  named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license
	  5909) translate_shutdown.patch uploaded by coreyfarrell (license
	  5909) ........ Merged revisions 398102 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398103 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398116 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 18:38 +0000 [r398101]  Matthew Jordan <mjordan@digium.com>

	* /, UPGRADE-12.txt (added), UPGRADE.txt: Update UPGRADE.txt file
	  for Asterisk 12 This simply pulls in the changes that were
	  breaking from the CHANGES file and updates a few other areas
	  accordingly. It also removes the 10 => 11 notes, which are
	  traditionally removed from each major version and stored in the
	  appropriate UPGRADE-X.txt file. ........ Merged revisions 398100
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 18:30 +0000 [r398064-398099]  Jonathan Rose <jrose@digium.com>

	* main/features_config.c, /, main/config_options.c:
	  features_config: Ignore parkinglots in features.conf instead of
	  failing to load Parkinglots are defined in res_features.conf now,
	  but this patch fixes features_config so that features don't fail
	  to load when parkinglots are present in features.conf Review:
	  https://reviewboard.asterisk.org/r/2801/ ........ Merged
	  revisions 398068 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/features_config.c, main/udptl.c, /: features_config: Don't
	  require features.conf to be present for Asterisk to load (closes
	  issue ASTERISK-22426) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2806/ ........ Merged
	  revisions 398020 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 17:59 +0000 [r398063]  Kevin Harwell <kharwell@digium.com>

	* main/manager.c, /, res/res_agi.c: Memory leak fix
	  ast_xmldoc_printable returns an allocated block that must be
	  freed by the caller. Fixed manager.c and res_agi.c to stop
	  leaking these results. (closes issue ASTERISK-22395) Reported by:
	  Corey Farrell Patches: manager-leaks-12.patch uploaded by
	  coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
	  by coreyfarrell (license 5909) ........ Merged revisions 398060
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 398061 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398062 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 17:11 +0000 [r398024-398026]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_substitution.c, /: test_substitution: Fix failing
	  test. Revert the -r392190 change. The original test was correct.
	  The CDR code was actually returning an unititialized buffer.
	  ........ Merged revisions 398025 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* tests/test_substitution.c, /: test_substituition: Fix failed test
	  reporting to actually report failure. You cannot put the "Testing
	  <blah> pass/fail" on a single line before actually performing the
	  test. Now any additional failure information is logged before the
	  test pass/fail announcement. * Added an additional CDR(answer,u)
	  test. ........ Merged revisions 398018 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 398019 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398023 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 16:27 +0000 [r398003-398017]  Kevin Harwell <kharwell@digium.com>

	* /, apps/app_mixmonitor.c: Fix memory leaks (closes issue
	  ASTERISK-22368) Reported by: Corey Farrell Patches:
	  issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes
	  (license 5674) ........ Merged revisions 398004 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
	  revisions 398011 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398016 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/asterisk.c, /: Check return value on fwrite ........ Merged
	  revisions 398000 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 398002 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 13:40 +0000 [r397987-397990]  David M. Lee <dlee@digium.com>

	* rest-api-templates/swagger_model.py, res/ari/ari_websockets.c,
	  channels/sip/include/sip.h, main/asterisk.c, res/res_ari.c,
	  tests/test_optional_api.c (added), /, channels/chan_sip.c,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  rest-api-templates/res_ari_resource.c.mustache,
	  res/ari/internal.h, res/res_http_websocket.c, CHANGES,
	  include/asterisk/compiler.h, include/asterisk/ari.h,
	  main/loader.c, include/asterisk/optional_api.h,
	  build_tools/cflags.xml, configure, res/res_ari_events.c,
	  include/asterisk/http_websocket.h, main/optional_api.c (added):
	  optional_api: Fix linking problems between modules that export
	  global symbols With the new work in Asterisk 12, there are some
	  uses of the optional_api that are prone to failure. The details
	  are rather involved, and captured on [the wiki][1]. This patch
	  addresses the issue by removing almost all of the magic from the
	  optional API implementation. Instead of relying on weak symbol
	  resolution, a new optional_api.c module was added to Asterisk
	  core. For modules providing an optional API, the pointer to the
	  implementation function is registered with the core. For modules
	  that use an optional API, a pointer to a stub function, along
	  with a optional_ref function pointer are registered with the
	  core. The optional_ref function pointers is set to the
	  implementation function when it's provided, or the stub function
	  when it's now. Since the implementation no longer relies on
	  magic, it is now supported on all platforms. In the spirit of
	  choice, an OPTIONAL_API flag was added, so we can disable the
	  optional_api if needed (maybe it's buggy on some bizarre platform
	  I haven't tested on) The AST_OPTIONAL_API*() macros themselves
	  remained unchanged, so existing code could remain unchanged. But
	  to help with debugging the optional_api, the patch limits the
	  #include of optional API's to just the modules using the API.
	  This also reduces resource waste maintaining optional_ref
	  pointers that aren't used. Other changes made as a part of this
	  patch: * The stubs for http_websocket that wrap system calls set
	  errno to ENOSYS. * res_http_websocket now properly increments
	  module use count. * In loader.c, the while() wrappers around
	  dlclose() were removed. The while(!dlclose()) is actually an
	  anti-pattern, which can lead to infinite loops if the module
	  you're attempting to unload exports a symbol that was directly
	  linked to. * The special handling of nonoptreq on systems without
	  weak symbol support was removed, since we no longer rely on weak
	  symbols for optional_api. [1]:
	  https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue
	  ASTERISK-22296) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2797/ ........ Merged
	  revisions 397989 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_stasis_playback.c, /,
	  include/asterisk/stasis_app_recording.h,
	  res/ari/resource_recordings.h, res/res_stasis_recording.c,
	  res/Makefile, res/ari/ari_model_validators.c,
	  rest-api/api-docs/recordings.json, res/stasis_recording (added),
	  res/ari/resource_recordings.c, res/ari/ari_model_validators.h,
	  res/res_ari_recordings.c: ARI: Implement /recordings/stored API's
	  his patch implements the ARI API's for stored recordings. While
	  the original task only specified deleting a recording, it was
	  simple enough to implement the GET for all recordings, and for an
	  individual recording. The recording playback operation was
	  modified to use the same code for accessing the recording as the
	  REST API, so that they will behave consistently. There were
	  several problems with the api-docs that were also fixed, bringing
	  the ARI spec in line with the implementation. There were some
	  'wishful thinking' fields on the stored recording model (duration
	  and timestamp) that were removed, because I ended up not
	  implementing a metadata file to go along with the recording to
	  store such information. The GET /recordings/live operation was
	  removed, since it's not really that useful to get a list of all
	  recordings that are currently going on in the system. (At least,
	  if we did that, we'd probably want to also list all of the
	  current playbacks. Which seems weird.) (closes issue
	  ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/
	  ........ Merged revisions 397985 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /: Multiple revisions 397975-397976 ........ r397975 | rmudgett |
	  2013-08-29 20:00:00 -0500 (Thu, 29 Aug 2013) | 1 line pbx.c: Make
	  ast_str_substitute_variables_full() not mask variables. ........
	  r397976 | rmudgett | 2013-08-29 20:00:41 -0500 (Thu, 29 Aug 2013)
	  | 1 line Revert last commit. ........ Merged revisions
	  397975-397976 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 01:20 +0000 [r397978]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /: pbx.c: Make pbx_substitute_variables_helper_full()
	  not mask variables. ........ Merged revisions 397977 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-30 00:11 +0000 [r397962-397969]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip_pidf.c, /: Sanitize XML output for PIDF bodies.
	  PJSIP's PIDF API does not replace angle brackets with their
	  appropriate counterparts for XML. So we have to do it ourself. In
	  this particular case, the problem had to do with attempting to
	  place an unsanitized SIP URI into an XML node. Now we don't get a
	  488 from recipients of our PIDF NOTIFYs. ........ Merged
	  revisions 397968 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_pidf.c, /: Fix method for creating activities
	  string in PIDF bodies. The previous method did not allocate
	  enough space to create the entire string, but adjusted the
	  string's slen value to be larger than the actual allocation. This
	  resulted in garbled text in NOTIFY requests from Asterisk. This
	  method allocates the proper amount of space first and then writes
	  the content into the buffer. ........ Merged revisions 397960
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 22:49 +0000 [r397959]  Kevin Harwell <kharwell@digium.com>

	* apps/app_dumpchan.c, main/logger.c, apps/app_verbose.c,
	  main/asterisk.c, channels/chan_misdn.c, /: Verbose logging
	  discrepancies Refactored cases where a combination of
	  ast_verbose/options_verbose were present. Also in general tried
	  to eliminate, in as many places as possible, where the
	  options_verbose global variable was being used. Refactored the
	  way local and remote consoles handle verbose message logging in
	  an attempt to solve the various discrepancies that sometimes
	  would show between the two. (closes issue AST-1193) Reported by:
	  Guenther Kelleter Review:
	  https://reviewboard.asterisk.org/r/2798/ ........ Merged
	  revisions 397948 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 397958 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 22:26 +0000 [r397956-397957]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip_pubsub.c: Fix when the subscription_terminated
	  callback is called for subscription handlers. The previous
	  placement would result in the resubscribe() callback called
	  instead of the subscription_terminated() callback being called
	  when a subscription was ended via a SUBSCRIBE request. This would
	  result in confusing PJSIP and having it throw an assertion.
	  ........ Merged revisions 397955 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* res/res_pjsip_session.c, /: Fix a race condition where a canceled
	  call was answered. RFC 5407 section 3.1.2 details a scenario
	  where a UAC sends a CANCEL at the same time that a UAS sends a
	  200 OK for the INVITE that the UAC is canceling. When this
	  occurs, it is the role of the UAC to immediately send a BYE to
	  terminate the call. This scenario was reproducible by have a
	  Digium phone with two lines place a call to a second phone that
	  forwarded the call to the second line on the original phone. The
	  Digium phone, upon realizing that it was connecting to itself,
	  would attempt to cancel the call. The timing of this happened to
	  trigger the aforementioned race condition about 80% of the time.
	  Asterisk was not doing its job of sending a BYE when receiving a
	  200 OK on a cancelled INVITE. The result was that the ast_channel
	  structure was destroyed but the underlying SIP session, as well
	  as the PJSIP inv_session and dialog, were still alive. Attempting
	  to perform an action such as a transfer, once in this state,
	  would result in Asterisk crashing. The circumstances are now
	  detected properly and the session is ended as recommended in RFC
	  5407. (closes issue AST-1209) reported by John Bigelow ........
	  Merged revisions 397945 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 21:37 +0000 [r397947]  Kevin Harwell <kharwell@digium.com>

	* main/file.c, main/app.c, main/config_options.c, main/cel.c,
	  main/asterisk.c, main/cdr.c, main/manager.c, /,
	  main/stasis_config.c: Memory leaks fix (closes ASTERISK-22376)
	  Reported by: John Hardin Patches: memleak.patch uploaded by
	  jhardin (license 6512) memleak2.patch uploaded by jhardin
	  (license 6512) ........ Merged revisions 397946 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 20:22 +0000 [r397939]  Matthew Jordan <mjordan@digium.com>

	* configs/safe_asterisk.conf.sample (removed), /, CHANGES,
	  contrib/scripts/safe_asterisk, Makefile: Revert r394939 due to
	  (numerous) objections The patch from ASTERISK-21965 was committed
	  perhaps a bit too hastily. Walter and Tzafrir have pointed out
	  numerous issues with the approach and have propsed an alternative
	  in r/2757. Since it's not a time critical issue and is not worth
	  holding up the release of 12 for it, I've gone ahead and reverted
	  r394939 from 12/trunk and re-opened ASTERISK-21965. ........
	  Merged revisions 397938 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 16:21 +0000 [r397932]  David M. Lee <dlee@digium.com>

	* rest-api-templates/make_ari_stubs.py, /,
	  rest-api-templates/api.wiki.mustache,
	  rest-api-templates/asterisk_processor.py: Account for {} in
	  Swagger notes ........ Merged revisions 397927 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 16:05 +0000 [r397925]  Matthew Jordan <mjordan@digium.com>

	* Makefile, /: Recursively search for '.c' files when making
	  documentation with 'make full' Without this, documentation
	  defined in sub-folders is ignored. Since having properly
	  generated documentation is especially important in Asterisk 12 -
	  not having it can cause a module to not load - 'make full' needs
	  to look in all .c files. ........ Merged revisions 397924 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 15:43 +0000 [r397923]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c, main/cel.c, main/stasis_bridges.c: Multiple
	  revisions 397921-397922 ........ r397921 | mmichelson |
	  2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines Resolve
	  assumptions that bridge snapshots would be non-NULL for transfer
	  stasis events. Attempting to transfer an unbridged call would
	  result in crashes in either CEL code or in the conversion to AMI
	  messages. ........ r397922 | mmichelson | 2013-08-29 10:42:29
	  -0500 (Thu, 29 Aug 2013) | 3 lines Remove extra debug message.
	  ........ Merged revisions 397921-397922 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-29 12:30 +0000 [r397912]  Matthew Jordan <mjordan@digium.com>

	* contrib/ast-db-manage/config,
	  contrib/ast-db-manage/config/script.py.mako,
	  contrib/ast-db-manage/voicemail.ini.sample,
	  contrib/ast-db-manage/voicemail/env.py,
	  contrib/ast-db-manage/voicemail,
	  contrib/ast-db-manage/voicemail/script.py.mako,
	  contrib/ast-db-manage/README.md,
	  contrib/ast-db-manage/config/versions,
	  contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py,
	  contrib/ast-db-manage (added),
	  contrib/ast-db-manage/voicemail/versions, /,
	  contrib/ast-db-manage/config.ini.sample,
	  contrib/ast-db-manage/config/env.py,
	  contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py:
	  Actually *add* the database schema management utilities In
	  r397874, the scripts were removed... but not replaced. Thanks to
	  Michael Young for noticing this! ........ Merged revisions 397911
	  from http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-28 23:15 +0000 [r397886-397903]  Richard Mudgett <rmudgett@digium.com>

	* main/cdr.c, /, funcs/func_cdr.c, main/stdtime/localtime.c: Fix
	  some uninitialized buffers for CDR handling valgrind found. *
	  Made ast_strftime_locale() ensure that the output buffer is
	  initialized. The std library strftime() returns 0 and does not
	  touch the buffer if it has an error. However, the function can
	  also return 0 without an error. (closes issue ASTERISK-22412)
	  Reported by: rmudgett ........ Merged revisions 397902 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: Fixed problems with ast_cdr_serialize_variables().
	  * Fixed return value of ast_cdr_serialize_variables() on error.
	  It needs to return 0 indicating no CDR variables found. * Made
	  ast_cdr_serialize_variables() check the return value of
	  cdr_object_format_property() and assert if nonzero. A member of
	  the cdr_readonly_vars[] was not handled. * Removed unused
	  elements from cdr_readonly_vars[]: total_duration, total_billsec,
	  first_start, and first_answer. ........ Merged revisions 397900
	  from http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: Made the on/off in CLI "cdr set debug [on|off]"
	  case insensitive. ........ Merged revisions 397898 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/cdr.c, /: Make CDR variable name chandling consistently case
	  insensitive. ........ Merged revisions 397896 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, main/cdr.c: Make CDR code deal with channel names case
	  insensitively. ........ Merged revisions 397894 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, funcs/func_cdr.c, main/cdr.c: Some CDR code optimization.
	  ........ Merged revisions 397892 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, funcs/func_cdr.c: Whitespace and curly braces. ........ Merged
	  revisions 397885 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-28 21:09 +0000 [r397877]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_pjsip_refer.c: Improve detection of answer on SIP
	  blind transfer. A problem encountered during testing was that
	  res_pjsip_refer would not ever send a NOTIFY with a 200 OK
	  sipfrag. This is because the framehook that was supposed to send
	  the NOTIFY would never be told that an answer had occurred. This
	  happened for two reasons: 1) The transferee channel on which the
	  framehook was on was already up. 2) Answers are rarely if ever
	  written to channels. Rather, the ast_answer() or ast_raw_answer()
	  function is used to answer channels. Thanks to a suggestion by
	  Matt Jordan, the best way to detect that the call had been
	  answered was to find out when the transferee channel joined a
	  bridge. With stasis this is an easy task. So now, in addition to
	  the framehook logic, there is a stasis subscription used to
	  determine when the transferee has entered a bridge. Once it has
	  entered, an appropriate NOTIFY is sent. ........ Merged revisions
	  397876 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-28 20:55 +0000 [r397872-397875]  Matthew Jordan <mjordan@digium.com>

	* contrib/realtime/mysql/queue_log.sql,
	  contrib/realtime/mysql/voicemail.sql,
	  contrib/realtime/mysql/sippeers.sql, /,
	  contrib/realtime/mysql/iaxfriends.sql,
	  contrib/realtime/mysql/meetme.sql,
	  contrib/realtime/mysql/voicemail_messages.sql,
	  contrib/realtime/postgresql/realtime.sql,
	  contrib/realtime/mysql/voicemail_data.sql, CHANGES,
	  contrib/realtime/mysql/musiconhold.sql: Add database schema
	  management using Alembic This patch replaces contrib/realtime/
	  with a new setup for managing the database schema required for
	  database integration with Asterisk. In addition to initializing a
	  database with the proper schema, alembic can do a database
	  migration to assist with upgrading Asterisk in the future.
	  Hopefully this helps make setting up and operating Asterisk with
	  a database easier. With this the schema only needs to be
	  maintained in one place instead of once per database. The schemas
	  I have added here have a bit of improvement over the examples
	  that were there before (some added consistency and added some
	  missing indexes). Managing the schema in one place here also
	  applies to all databases supported by SQLAlchemy. See
	  contrib/ast-db-manage/README.md for more details. Review:
	  https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant
	  (license 6300) ........ Merged revisions 397874 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* CHANGES, /: Update CHANGES file for Asterisk 12 This updates the
	  Asterisk 12 CHANGES file with the things that were missed during
	  the development cycle. Review:
	  https://reviewboard.asterisk.org/r/2795/ ........ Merged
	  revisions 397870 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-28 16:13 +0000 [r397857-397860]  Richard Mudgett <rmudgett@digium.com>

	* /, main/pbx.c: pbx.c: Make ast_str_substitute_variables_full()
	  not mask variables. ........ Merged revisions 397859 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* main/chanvars.c: ast_free() is null tollerant.

	* include/asterisk/threadstorage.h, /: Match use of ast_free() with
	  ast_calloc() and add some curly braces. ........ Merged revisions
	  397856 from http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-28 15:43 +0000 [r397855]  Mark Michelson <mmichelson@digium.com>

	* res/res_pjsip/pjsip_distributor.c, /: Fix dialog matching in the
	  SIP distributor. Dialog matching is performed in the distributor
	  for the sole purpose of retrieving an associated serializer so
	  the request may be serialized. This patch fixes two problems.
	  First, incoming CANCEL requests that had no to-tag (which really
	  should be *all* CANCEL requests) would not match with a dialog.
	  An earlier bug fix to deal with early CANCEL requests would
	  result in the CANCEL being replied to with a 481. The fix for
	  this is to find the matching INVITE transaction and get the
	  dialog from that transaction. Second, no SIP responses were
	  matching dialogs. This is because we were inverting the tags that
	  we were passing into PJSIP's dialog finding function. This logic
	  has been corrected by setting local and remote tag variables
	  based on whether the incoming message is a request or response.
	  ........ Merged revisions 397854 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-27 19:19 +0000 [r397820]  David M. Lee <dlee@digium.com>

	* rest-api-templates/param_parsing.mustache, res/res_ari_bridges.c,
	  /, res/stasis/app.c, res/res_ari_events.c,
	  res/res_ari_asterisk.c,
	  rest-api-templates/res_ari_resource.c.mustache, res/stasis/app.h,
	  res/res_stasis.c, main/stasis_bridges.c: ARI: WebSocket event
	  cleanup Stasis events (which get distributed over the ARI
	  WebSocket) are created by subscribing to the channel_all_cached
	  and bridge_all_cached topics, filtering out events for
	  channels/bridges currently subscribed to. There are two issues
	  with that. First was a race condition, where messages in-flight
	  to the master subscribe-to-all-things topic would get sent out,
	  even though the events happened before the channel was put into
	  Stasis. Secondly, as the number of channels and bridges grow in
	  the system, the work spent filtering messages becomes excessive.
	  Since r395954, individual channels and bridges have caching
	  topics, and can be subscribed to individually. This patch takes
	  advantage, so that channels and bridges are subscribed to on
	  demand, instead of filtering the global topics. The one case
	  where filtering is still required is handling BridgeMerge
	  messages, which are published directly to the bridge_all topic.
	  Other than the change to how subscriptions work, this patch
	  mostly just moves code around. Most of the work generating JSON
	  objects from messages was moved to .to_json handlers on the
	  message types. The callback functions handling app subscriptions
	  were moved from res_stasis (b/c they were global to the model) to
	  stasis/app.c (b/c they are local to the app now). (closes issue
	  ASTERISK-21969) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2754/ ........ Merged
	  revisions 397816 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-27 18:52 +0000 [r397811]  Richard Mudgett <rmudgett@digium.com>

	* /, main/astmm.c: Made MALLOC_DEBUG less CPU intensive by default.
	  Storing a backtrace for each allocation in anticipation of a
	  memory management problem is very CPU intensive. * Added the CLI
	  "memory backtrace {on|off}" command to request that the backtrace
	  be gathered only on request. The backtrace is off by default.
	  (issue ASTERISK-22221) Reported by: Matt Jordan ........ Merged
	  revisions 397809 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-27 18:10 +0000 [r397753-397760]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
	  SDP If the SIP channel driver processes an invalid SDP that
	  defines media descriptions before connection information, it may
	  attempt to reference the socket address information even though
	  that information has not yet been set. This will cause a crash.
	  This patch adds checks when handling the various media
	  descriptions that ensures the media descriptions are handled only
	  if we have connection information suitable for that media. Thanks
	  to Walter Doekes, OSSO B.V., for reporting, testing, and
	  providing the solution to this problem. (closes issue
	  ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
	  issueA22007_sdp_without_c_death.patch uploaded by wdoekes
	  (License 5674) ........ Merged revisions 397756 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 397757 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
	  revisions 397758 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 397759 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK
	  on dialog that has no channel A remote exploitable crash
	  vulnerability exists in the SIP channel driver if an ACK with SDP
	  is received after the channel has been terminated. The handling
	  code incorrectly assumed that the channel would always be
	  present. This patch adds a check such that the SDP will only be
	  parsed and applied if Asterisk has a channel present that is
	  associated with the dialog. Note that the patch being applied was
	  modified only slightly from the patch provided by Walter Doekes
	  of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
	  Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
	  issueA21064_fix.patch uploaded by wdoekes (License 5674) ........
	  Merged revisions 397710 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 397711 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
	  revisions 397712 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 397713 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-27 16:51 +0000 [r397746]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c,
	  channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/chan_sip.c, channels/chan_motif.c: Fix uninitialized
	  value in struct ast_control_pvt_cause_code usage. ........ Merged
	  revisions 397744 from
	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
	  revisions 397745 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-26 23:48 +0000 [r397691]  Matthew Jordan <mjordan@digium.com>

	* /, main/bridge_channel.c: Better handle clearing the OUTGOING
	  flag when a channel leaves a bridge When a channel with the
	  OUTGOING flag leaves a bridge, and it will survive being pulled
	  from the bridge (either because it will execute dialplan, go into
	  another bridge, or live in a friendly autoloop), we have to clear
	  the OUTGOING flag. This is the signal to the CDR engine that this
	  channel is no longer a second class citizen, i.e., it is not
	  "dialed". The soft hangup flags are only half the picture. If a
	  channel is being moved from one bridge to another, the soft
	  hangup flags aren't set; however, the state of the bridge_channel
	  will not be hung up. Since the channel does not have one of the
	  two hang up states, that implies that the channel is still
	  technically alive. This patch modifies the check so that it
	  checks both the soft hangup flags as well as the bridge_channel
	  state. If either suggests that the channel is going to persist,
	  we clear the OUTGOING flag. ........ Merged revisions 397690 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-26 21:32 +0000 [r397674]  David M. Lee <dlee@digium.com>

	* /, main/bucket.c: Fixed bucket.c for systems where tv_usec is not
	  an unsigned long. ........ Merged revisions 397673 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-26 16:25 +0000 [r397644-397651]  Richard Mudgett <rmudgett@digium.com>

	* /, include/asterisk/bridge_channel.h, main/bridge_channel.c:
	  bridging: Fix a livelock with local channel optimization. Use a
	  better means of waking up the bridge channel thread. ........
	  Merged revisions 397650 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* channels/Makefile, /: chan_dahdi: Add some missing build cleanup.
	  ........ Merged revisions 397643 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-25 18:12 +0000 [r397622-397631]  Matthew Jordan <mjordan@digium.com>

	* tests/test_bucket.c, /: Fix bucket unit tests After the review
	  for buckets was completed (r2715), the handling of names in the
	  bucket core was deferred to the wizards. As such, the bucket unit
	  tests cannot expect that passing a URI with a scheme specified
	  but no actual resource name will automatically fail. The tests
	  have been updated to not make this check. ........ Merged
	  revisions 397630 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* include/asterisk/config_options.h, /, main/config_options.c,
	  tests/test_config.c: Fix the config_options_test The config
	  options test requires the entire configuration item to be
	  transparent from the documentation system. So we let it do that
	  too. As an aside, please do not use this power for evil.
	  Documentation is your friend, and you really should document your
	  configurations. Hiding your module's configuration information
	  from the system attempting to enforce some sanity in the universe
	  is something only a Bond villain would contemplate. ........
	  Merged revisions 397628 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

	* /, res/res_pjsip/pjsip_configuration.c: Add rtpengine
	  configuration parameter The rtpengine configuration parameter was
	  documented in the XML documentation, but it was not actually
	  registered with the sorcery object. This adds the parameter with
	  a default of "asterisk", such that res_rtp_asterisk is chosen as
	  the default RTP implementation. (closes issue ASTERISK-22380)
	  Reported by: Rusty Newton Tested by: Rusty Newton ........ Merged
	  revisions 397621 from
	  http://svn.asterisk.org/svn/asterisk/branches/12

2013-08-23 22:40 +0000 [r397615]  Matthew Jordan <mjordan@digium.com>

	* /: Set new merge properties on 12

2013-08-23 22:20 +0000 [r397613]  Joshua Colp <jcolp@digium.com>

	* main/bucket.c: Fix building of trunk. Note: This is why I commit
	  on the weekend.

