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asterisk/ChangeLog
Asterisk Autobuilder 0aaa1abcc0 Update ChangeLog, .version, remove old summaries 
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/12.5.0@421398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:01:15 +00:00

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2014-08-19 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.5.0 Released.
2014-08-11 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.5.0-rc1 Released.
2014-08-11 18:48 +0000 [r420805] Matthew Jordan <mjordan@digium.com>
* rest-api/api-docs/playbacks.json, UPGRADE.txt,
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.
2014-08-11 18:45 +0000 [r420795-420802] 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
* res/stasis/stasis_bridge.c, include/asterisk/stasis_app.h,
res/res_stasis.c, res/ari/resource_bridges.c: 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/
2014-08-11 10:37 +0000 [r420656-420716] 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
* 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
2014-08-08 12:31 +0000 [r420533] Matthew Jordan <mjordan@digium.com>
* main/message.c: main/message: remove debug message
2014-08-08 02:51 +0000 [r420513] 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.
2014-08-07 21:48 +0000 [r420436] 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
2014-08-07 21:16 +0000 [r420408-420414] 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.
* 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/
2014-08-07 19:43 +0000 [r420385-420387] Mark Michelson <mmichelson@digium.com>
* main/bridge.c: Ensure bridges exist when trying to determine
bridged parties when publishing transfer information.
* res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
res/res_pjsip_pidf_body_generator.c, main/bridge.c,
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:
Revert previous patch since it had some unreviewed content in it.
* 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, main/bridge.c,
res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c,
res/res_pjsip_xpidf_body_generator.c: Ensure bridges actually
exist when trying to determine the bridged peer.
2014-08-07 15:19 +0000 [r420325] Kinsey Moore <kmoore@digium.com>
* res/ari/ari_model_validators.c, main/cel.c, apps/app_queue.c,
main/stasis_bridges.c, main/channel.c,
res/ari/ari_model_validators.h, include/asterisk/datastore.h,
tests/test_cel.c, 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, res/stasis/stasis_bridge.c,
main/bridge_basic.c, res/stasis/command.h,
include/asterisk/stasis_bridges.h, include/asterisk/stasis_app.h,
res/stasis/app.h, res/stasis/control.h: 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/
2014-08-06 21:47 +0000 [r420211-420262] Richard Mudgett <rmudgett@digium.com>
* main/format.c: Change comment.
* 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,
contrib/ast-db-manage/voicemail/versions/39428242f7f5_increase_recording_column_size.py
(added): 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/
2014-08-06 16:10 +0000 [r420148] George Joseph <george.joseph@fairview5.com>
* main/pbx.c, /, pbx/pbx_lua.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
2014-08-05 21:47 +0000 [r420089-420099] Matthew Jordan <mjordan@digium.com>
* res/stasis/messaging.c: stasis: Fix compilation issue with ao2
tagged objects
* tests/test_message.c: test_message: Fix strict-aliasing
compilation issue
* /: Remove automerge properties :-(
* res/ari/resource_channels.c, res/res_stasis.c, main/message.c,
res/stasis/messaging.c (added), rest-api/api-docs/endpoints.json,
res/ari/resource_endpoints.c, rest-api/api-docs/events.json, /,
include/asterisk/vector.h, 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,
res/ari/ari_model_validators.c, include/asterisk/manager.h,
CHANGES, res/ari/ari_model_validators.h, main/json.c,
res/res_ari_endpoints.c, include/asterisk/message.h: 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
2014-08-05 19:12 +0000 [r420060] Richard Mudgett <rmudgett@digium.com>
* main/format.c, /: format.c: Add reason comments for the
format_list ordering. ........ Merged revisions 420054 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-08-04 19:45 +0000 [r419944] 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
2014-07-31 11:57 +0000 [r419823-419824] Matthew Jordan <mjordan@digium.com>
* /: Get rid of automerge properties
* res/res_rtp_asterisk.c, main/rtp_engine.c, /, res/res_hep_rtcp.c
(added), CHANGES, channels/chan_pjsip.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
2014-07-29 10:52 +0000 [r419750-419764] 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/
* res/res_pjsip_pidf_body_generator.c,
res/res_pjsip_xpidf_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/
2014-07-28 18:50 +0000 [r419686] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /, funcs/func_frame_trace.c, main/abstract_jb.c,
apps/app_speech_utils.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
2014-07-25 14:46 +0000 [r419565-419566] Matthew Jordan <mjordan@digium.com>
* CHANGES: Update CHANGES for r419565
* res/res_stasis_recording.c, res/ari/ari_model_validators.c,
rest-api/api-docs/recordings.json,
res/ari/ari_model_validators.h: 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
2014-07-25 10:53 +0000 [r419536-419538] 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/
* 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/
2014-07-24 17:57 +0000 [r419442] Corey Farrell <git@cfware.com>
* /, channels/chan_sip.c: chan_sip: sip_subscribe_mwi_destroy
should not call sip_destroy sip_subscribe_mwi_destroy calls
sip_destroy on the reference counted mwi->call. This results in
the fields of mwi->call being freed, but mwi->call itself it
leaked. If other code is still using mwi->call it can cause
problems. This change uses dialog_unref instead, to balance the
ref provided by sip_alloc(). ASTERISK-24087 #close Reported by:
Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/
........ Merged revisions 419440 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 419441 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-07-24 16:50 +0000 [r419376] 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
2014-07-23 16:45 +0000 [r419316-419318] 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.
* res/res_pjsip_refer.c: res_pjsip_refer: remove stray debugging
line How'd those @ symbols get in there...
2014-07-23 13:58 +0000 [r419285] 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
2014-07-22 16:12 +0000 [r419196] Matthew Jordan <mjordan@digium.com>
* 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,
include/asterisk/endpoints.h,
rest-api/api-docs/applications.json, include/asterisk/xmpp.h,
main/channel_internal_api.c, channels/chan_motif.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
2014-07-22 14:13 +0000 [r419163] Kinsey Moore <kmoore@digium.com>
* addons/ooh323c/src/printHandler.c, tests/test_sorcery_realtime.c,
addons/ooh323c/src/ooq931.c, tests/test_json.c,
tests/test_astobj2_thrash.c, addons/chan_ooh323.c, /,
tests/test_abstract_jb.c, apps/app_meetme.c,
tests/test_optional_api.c, tests/test_logger.c,
tests/test_event.c, tests/test_format_api.c,
tests/test_hashtab_thrash.c, channels/chan_gtalk.c,
res/res_mwi_external_ami.c, res/res_jabber.c,
tests/test_sorcery.c, channels/chan_jingle.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
2014-07-18 21:25 +0000 [r419021] Matthew Jordan <mjordan@digium.com>
* CHANGES, 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: 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
2014-07-18 21:24 +0000 [r418996-419019] 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().
* 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/
2014-07-18 16:46 +0000 [r418937] Richard Mudgett <rmudgett@digium.com>
* funcs/func_audiohookinherit.c: func_audiohookinherit.c: Fixup
some XML documentation wording.
2014-07-18 16:01 +0000 [r418914] Jonathan Rose <jrose@digium.com>
* 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,
main/channel.c, funcs/func_audiohookinherit.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/
2014-07-17 22:17 +0000 [r418886] Scott Griepentrog <sgriepentrog@digium.com>
* 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/
2014-07-17 14:27 +0000 [r418810] 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.
2014-07-16 23:06 +0000 [r418787] 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.
2014-07-16 13:58 +0000 [r418756] Matthew Jordan <mjordan@digium.com>
* 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/res_pjsip/pjsip_configuration.c, CHANGES, res/res_pjsip.c:
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
2014-07-15 23:03 +0000 [r418715] 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.
2014-07-15 22:20 +0000 [r418714] Matthew Jordan <mjordan@digium.com>
* 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. Review:
https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close
........ Merged revisions 418713 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-07-15 17:45 +0000 [r418650] 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
2014-07-15 17:14 +0000 [r418636] Scott Griepentrog <sgriepentrog@digium.com>
* 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/
2014-07-14 14:46 +0000 [r418586] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/logger.h: logger.h: Extract DEBUG_ATLEAST() to
complement VERBOSITY_ATLEAST().
2014-07-13 21:55 +0000 [r418466-418506] 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
* 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
2014-07-11 21:09 +0000 [r418396] 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/
2014-07-11 16:40 +0000 [r418369] 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
2014-07-10 15:35 +0000 [r418324] 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
2014-07-10 01:52 +0000 [r418225-418263] 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
* res/stasis/stasis_bridge.c (added),
include/asterisk/bridge_channel.h, main/bridge_basic.c,
res/stasis/stasis_bridge.h (added), main/bridge_channel.c,
res/res_stasis.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/
2014-07-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.4.0 Released.
2014-07-08 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.4.0-rc1 Released.
2014-07-08 14:47 +0000 [r418172-418182] Matthew Jordan <mjordan@digium.com>
* 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,
rest-api/api-docs/playbacks.json, UPGRADE.txt,
rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
rest-api/resources.json: manager/ARI: Update version to
2.4.0/1.4.0; Update UPGRADE.txt
* 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
2014-07-07 16:05 +0000 [r418116] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c,
include/asterisk/res_pjsip_presence_xml.h,
include/asterisk/res_pjsip_body_generator_types.h,
res/res_pjsip_dialog_info_body_generator.c (added):
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/
2014-07-07 02:13 +0000 [r418089] Matthew Jordan <mjordan@digium.com>
* res/ari/resource_channels.c, res/res_stasis.c, res/stasis/app.c,
include/asterisk/stasis_app.h: 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
2014-07-07 01:18 +0000 [r418066-418071] 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/
* main/http.c: HTTP: Fix build for gcc 4.10
2014-07-03 22:06 +0000 [r417880-417958] Richard Mudgett <rmudgett@digium.com>
* UPGRADE.txt, channels/sig_pri.c, channels/sig_pri.h,
channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /:
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
* res/res_ari.c, main/manager.c, res/ari/resource_channels.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
* configs/http.conf.sample, include/asterisk/http.h, main/tcptls.c,
res/res_ari.c, main/manager.c, res/res_phoneprov.c, main/http.c,
UPGRADE.txt, include/asterisk/tcptls.h, res/res_http_post.c,
res/res_http_websocket.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/
2014-07-03 16:07 +0000 [r417878] sgalarneau <sgalarneau@localhost>:
* res/ari/resource_channels.h, rest-api/api-docs/events.json,
res/ari/resource_events.h, rest-api/api-docs/channels.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/
2014-07-03 11:26 +0000 [r417799] Matthew Jordan <mjordan@digium.com>
* /, 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
2014-07-01 14:40 +0000 [r417678-417705] 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.
* main/sdp_srtp.c, res/res_pjsip_sdp_rtp.c,
res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
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,
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: 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/
2014-06-30 03:25 +0000 [r417589] Matthew Jordan <mjordan@digium.com>
* /, 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
2014-06-27 23:11 +0000 [r417565] 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.
2014-06-27 19:27 +0000 [r417483-417509] 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
* 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
2014-06-27 13:48 +0000 [r417311-417460] Matthew Jordan <mjordan@digium.com>
* res/res_pjsip_pubsub.c, res/res_pjsip_registrar.c,
include/asterisk/res_pjsip.h,
res/res_pjsip_outbound_registration.c,
res/res_pjsip/pjsip_configuration.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/
* 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
* 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
* 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
* 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/ari/internal.h, configs/ari.conf.sample,
res/res_pjsip.c, res/res_http_websocket.c: 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
2014-06-26 10:05 +0000 [r417212-417250] 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
* 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/
2014-06-23 18:49 +0000 [r417142] 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
2014-06-23 15:53 +0000 [r417119] 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().
2014-06-22 18:44 +0000 [r416995] 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
2014-06-20 23:16 +0000 [r416871-416931] 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
* 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
2014-06-20 20:46 +0000 [r416849] 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/
2014-06-20 15:22 +0000 [r416737-416806] 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/
* 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/
2014-06-19 19:35 +0000 [r416734] Kinsey Moore <kmoore@digium.com>
* channels/sip/reqresp_parser.c, main/logger.c, main/test.c, /,
main/bridge.c, res/parking/parking_tests.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
2014-06-19 16:03 +0000 [r416582-416669] 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
* /, 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
2014-06-18 04:16 +0000 [r416557] 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)
2014-06-17 18:43 +0000 [r416442-416502] 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
* res/res_pjsip_xpidf_body_generator.c,
res/res_pjsip_pidf_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.
2014-06-17 16:26 +0000 [r416441] 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
2014-06-16 09:02 +0000 [r416338] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* main/db.c, res/res_config_sqlite3.c, cdr/cdr_sqlite3_custom.c, /,
cel/cel_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
2014-06-16 02:39 +0000 [r416255-416318] 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.
* 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.
* 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.
* 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
2014-06-13 18:16 +0000 [r416211] Matthew Jordan <mjordan@digium.com>
* main/channel.c, main/dial.c, main/manager.c,
include/asterisk/stasis_channels.h, 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: 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/
2014-06-13 13:10 +0000 [r416148-416152] 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
* 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/
2014-06-13 05:13 +0000 [r416070] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h,
main/tcptls.c, main/manager.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
2014-06-12 21:17 +0000 [r416001] 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. ASTERISK-23764 Review:
https://reviewboard.asterisk.org/r/3610/ ........ Merged
revisions 415998 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 415999 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-06-12 20:13 +0000 [r415980] Kinsey Moore <kmoore@digium.com>
* res/res_pjsip_pubsub.c: Fix build in devmode for GCC 4.10
2014-06-12 16:41 +0000 [r415896] 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
2014-06-12 15:50 +0000 [r415838] Jonathan Rose <jrose@digium.com>
* /, UPGRADE.txt: Correct UPGRADE.txt notes in r415825 The change
was marked against the wrong version of Asterisk. My apologies.
........ Merged revisions 415837 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-06-12 15:41 +0000 [r415836] 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
2014-06-12 15:26 +0000 [r415832] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt, apps/app_mixmonitor.c, /: 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
2014-06-12 14:38 +0000 [r415812] 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
2014-06-12 14:03 +0000 [r415794] 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
2014-06-12 11:33 +0000 [r415766] Joshua Colp <jcolp@digium.com>
* 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/res_pjsip.c,
res/res_pjsip_pubsub.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/
2014-06-12 07:47 +0000 [r415748] Walter Doekes <walter+asterisk@wjd.nu>
* contrib/scripts/safe_asterisk, Makefile, UPGRADE.txt:
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)
2014-06-11 22:54 +0000 [r415729] 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
2014-06-10 18:31 +0000 [r415678] Kinsey Moore <kmoore@digium.com>
* main/channel.c: Fix build in dev mode due to signed/unsigned
mismatch
2014-06-10 15:46 +0000 [r415658] 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/
2014-06-10 09:18 +0000 [r415602] 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
2014-06-09 20:20 +0000 [r415579] 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.
2014-06-09 12:08 +0000 [r415523] 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
2014-06-09 03:48 +0000 [r415465] 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
2014-06-08 18:11 +0000 [r415443] Matthew Jordan <mjordan@digium.com>
* main/channel.c, main/pbx.c, main/framehook.c,
main/bridge_after.c, include/asterisk/channel.h,
bridges/bridge_native_rtp.c, main/bridge_channel.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/
2014-06-07 00:41 +0000 [r415427] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridge.h: bridge.h: Remove redundant struct
ast_bridge_channel forward declaration.
2014-06-06 21:35 +0000 [r415410] Jonathan Rose <jrose@digium.com>
* main/manager.c, /, channels/chan_sip.c,
include/asterisk/config.h, include/asterisk/manager.h,
main/config.c: 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
2014-06-06 19:08 +0000 [r415342] 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
2014-06-06 15:19 +0000 [r415336] Richard Mudgett <rmudgett@digium.com>
* utils: utils: Update svn:ignore for the new astobj2 split files.
v12 only
2014-06-06 14:08 +0000 [r415317-415318] George Joseph <george.joseph@fairview5.com>
* utils/Makefile: Update utils/Makefile so refcounter compiles post
astobj2 split. utils/refcounter was removed from trunk so this is
a 12-only patch to keep refcounter from failing to build.
https://reviewboard.asterisk.org/r/3576/
* 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,
tests/test_astobj2.c: 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/
2014-06-06 12:48 +0000 [r415301] 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/
2014-06-05 17:51 +0000 [r415230] Richard Mudgett <rmudgett@digium.com>
* 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
2014-06-05 14:39 +0000 [r415207] 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
2014-06-05 11:55 +0000 [r415191] 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/
2014-06-04 20:14 +0000 [r415172] 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
2014-06-04 14:12 +0000 [r415115-415117] 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.
* 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.
2014-06-04 07:23 +0000 [r415078] Corey Farrell <git@cfware.com>
* apps/app_confbridge.c, /, apps/confbridge/include/confbridge.h:
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
2014-06-03 07:34 +0000 [r414999] 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 ~ASTERISK-23582 #comment test
-ASTERISK-23582 #comment test2 X-ASTERISK-23582 #comment test3
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
2014-06-01 15:31 +0000 [r414975] 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/
2014-05-30 14:46 +0000 [r414948] 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
2014-05-30 12:39 +0000 [r414882-414934] Matthew Jordan <mjordan@digium.com>
* main/stasis_channels.c, 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: 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
* /, 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
2014-05-29 18:44 +0000 [r414860] 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
2014-05-28 20:52 +0000 [r414780] 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.
2014-05-28 17:45 +0000 [r414763-414765] 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, UPGRADE.txt,
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.
* 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.
2014-05-28 16:54 +0000 [r414747-414749] Richard Mudgett <rmudgett@digium.com>
* res/res_pjsip_t38.c, res/res_pjsip_session.c,
include/asterisk/res_pjsip_session.h: 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/
* apps/app_agent_pool.c, CHANGES: 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/
2014-05-28 11:37 +0000 [r414695] 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
2014-05-28 09:43 +0000 [r414566-414678] 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
* /, 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
* 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
2014-05-25 02:30 +0000 [r414542] 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.
2014-05-23 17:35 +0000 [r414528] 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.
2014-05-23 14:35 +0000 [r414474] Kinsey Moore <kmoore@digium.com>
* main/event.c, res/res_pjsip/config_transport.c,
channels/chan_pjsip.c, res/parking/parking_bridge_features.c,
res/parking/parking_manager.c, res/res_pjsip_refer.c,
res/parking/parking_bridge.c, main/bridge.c,
res/res_pjsip_sdp_rtp.c: Fix signed/unsigned build warnings
2014-05-29 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.3.0 Released.
2014-05-28 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.3.0-rc2 Released.
* 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.
* 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
* 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.
* 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.
2014-05-22 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.3.0-rc1 Released.
2014-05-22 16:08 +0000 [r414405] Scott Griepentrog <sgriepentrog@digium.com>
* main/stasis_channels.c, res/res_stasis.c,
main/manager_channels.c, main/stasis_endpoints.c,
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: 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/
2014-05-22 16:00 +0000 [r414404] 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
2014-05-22 15:44 +0000 [r414400] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, main/parking.c, main/bridge.c,
main/bridge_basic.c, res/parking/parking_applications.c,
include/asterisk/parking.h, 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: 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/
2014-05-22 14:01 +0000 [r414330-414347] 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
* main/event.c, main/stasis.c, include/asterisk/devicestate.h,
include/asterisk/event.h, main/stasis_message.c,
include/asterisk/event_defs.h, res/res_corosync.c,
include/asterisk/stasis.h, main/app.c, main/devicestate.c:
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
2014-05-21 22:17 +0000 [r414272] 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
2014-05-21 19:07 +0000 [r414216] 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 ASTEIRSK-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
2014-05-19 19:50 +0000 [r414195] 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/
2014-05-19 13:46 +0000 [r414154] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, /: chan_ooh323: fix h323_log full path name
* fix to use astlogdir option for h323_log file instead of
hardcoded ASTERISK-23754 #close Reported by: Igor Goncharovsky
Patches: ooh323_logger_patch.diff ........ Merged revisions
414152 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 414153 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-05-19 01:09 +0000 [r414122-414137] Matthew Jordan <mjordan@digium.com>
* main/framehook.c, 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:
Undo r414122 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/
2014-05-16 20:05 +0000 [r413993-414069] 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
* /, 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
* /, 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
2014-05-14 15:39 +0000 [r413896] 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
2014-05-13 18:01 +0000 [r413877] 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
2014-05-13 13:52 +0000 [r413789-413792] 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
* 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
2014-05-13 00:25 +0000 [r413766-413771] Richard Mudgett <rmudgett@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
channels/sig_pri.c: 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
* channels/sig_pri.c: Fix compiler warning from GCC 4.10 fixup.
2014-05-12 22:23 +0000 [r413712] 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
2014-05-11 02:05 +0000 [r413650-413681] Joshua Colp <jcolp@digium.com>
* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
include/asterisk/framehook.h, main/channel.c, main/framehook.c,
main/bridge_basic.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/
* main/framehook.c, main/bridge_basic.c,
include/asterisk/channel.h, bridges/bridge_native_rtp.c,
include/asterisk/framehook.h, main/channel.c: Undoing framehook
support. Issues were uncovered by Bamboo.
* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
include/asterisk/framehook.h, main/channel.c, main/framehook.c,
main/bridge_basic.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/
2014-05-09 23:13 +0000 [r413588-413597] 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
* main/slinfactory.c, main/core_unreal.c, main/acl.c,
res/res_pjsip_t38.c, channels/sig_pri.c, channels/chan_jingle.c,
channels/chan_dahdi.c, channels/sig_analog.c,
include/asterisk/astobj.h, res/res_corosync.c,
res/res_stun_monitor.c, apps/app_sms.c, main/audiohook.c,
pbx/pbx_config.c, channels/iax2/firmware.c, apps/app_adsiprog.c,
channels/chan_sip.c, funcs/func_sysinfo.c, main/utils.c,
res/res_format_attr_h263.c, res/res_jabber.c,
res/res_http_websocket.c, res/res_pktccops.c, res/res_monitor.c,
main/file.c, res/res_pjsip/pjsip_configuration.c, main/adsi.c,
channels/sip/include/sip.h, cel/cel_pgsql.c, main/pbx.c,
res/res_calendar_icalendar.c, res/res_crypto.c, main/aoc.c,
channels/chan_gtalk.c, main/netsock.c, res/res_ari_model.c,
res/res_config_odbc.c, res/res_pjsip_outbound_registration.c,
main/event.c, funcs/func_iconv.c, apps/app_stack.c,
res/res_calendar.c, res/res_sorcery_config.c, main/frame.c,
main/parking.c, res/res_format_attr_h264.c, channels/chan_iax2.c,
apps/confbridge/conf_config_parser.c, funcs/func_hangupcause.c,
main/manager.c, formats/format_pcm.c, funcs/func_srv.c,
res/res_format_attr_silk.c, main/asterisk.c, main/xmldoc.c,
res/res_rtp_asterisk.c, main/format.c, main/ccss.c,
res/res_calendar_caldav.c, main/enum.c, main/config.c,
res/res_srtp.c, main/loader.c,
channels/pjsip/dialplan_functions.c, funcs/func_channel.c,
main/bucket.c, main/abstract_jb.c, res/res_stasis_recording.c,
apps/app_verbose.c, main/dsp.c, apps/app_voicemail.c,
main/stun.c, main/security_events.c, apps/app_festival.c,
res/res_timing_dahdi.c, main/devicestate.c, res/res_xmpp.c,
apps/app_getcpeid.c, main/cli.c, res/res_format_attr_celt.c,
main/manager_bridges.c, cel/cel_odbc.c, channels/chan_skinny.c,
funcs/func_frame_trace.c, main/callerid.c, pbx/pbx_dundi.c,
res/res_pjsip_pubsub.c, res/res_fax_spandsp.c,
channels/chan_mgcp.c, res/res_stasis_playback.c, /,
main/translate.c, cdr/cdr_adaptive_odbc.c, res/res_musiconhold.c,
pbx/dundi-parser.c, apps/app_queue.c, res/res_calendar_ews.c,
channels/iax2/parser.c, main/io.c, channels/chan_phone.c,
res/res_agi.c, channels/chan_motif.c, apps/app_minivm.c,
apps/app_dumpchan.c, main/logger.c, apps/app_confbridge.c,
channels/sip/config_parser.c, res/res_odbc.c,
main/manager_channels.c, main/udptl.c, apps/app_dial.c,
res/res_fax.c, funcs/func_env.c, bridges/bridge_softmix.c,
main/taskprocessor.c, res/res_stasis_snoop.c,
res/res_format_attr_opus.c, res/ael/pval.c, main/channel.c,
main/cdr.c, main/data.c, res/res_pjsip/location.c,
main/config_options.c, main/app.c, channels/chan_alsa.c,
main/stdtime/localtime.c, main/bridge_channel.c,
res/res_pjsip_registrar.c, main/sched.c, channels/chan_unistim.c,
main/rtp_engine.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
2014-05-09 16:35 +0000 [r413556] 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
2014-05-08 00:35 +0000 [r413487] 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
2014-05-07 20:58 +0000 [r413452-413454] 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
* /, 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
2014-05-07 17:50 +0000 [r413306-413398] 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
* 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.
* 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)
* 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
2014-05-02 20:35 +0000 [r413226-413282] Mark Michelson <mmichelson@digium.com>
* res/res_config_odbc.c: Correct variable traversal logic in
res_config_odbc's update_odbc function. Closes issue
ASTERISK-23675 Reported by Leando Dardini Patches:
asterisk-23675-odbc-linkedlist-traversal_12.diff uploaded by
Michael L. Young (license #5026)
* 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
* /, 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
2014-05-02 16:33 +0000 [r413210] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c, UPGRADE.txt, res/res_pjsip_refer.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/
2014-05-02 15:58 +0000 [r413196] Jonathan Rose <jrose@digium.com>
* CHANGES, res/parking/parking_bridge_features.c,
res/parking/parking_manager.c, res/parking/res_parking.h:
Parking: Add 'AnnounceChannel' argument to manager action 'Park'
(closes ASTERISK-23397) Reported by: Denis Review:
https://reviewboard.asterisk.org/r/3446/
2014-05-01 15:41 +0000 [r413173] Mark Michelson <mmichelson@digium.com>
* 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
2014-05-01 12:30 +0000 [r413159] 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/
2014-04-30 20:47 +0000 [r413142] 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
2014-04-30 20:38 +0000 [r413140] 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.
2014-04-30 13:06 +0000 [r413124] 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
2014-04-30 12:41 +0000 [r413117-413121] 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/
* 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/
* 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/
2014-04-29 15:09 +0000 [r413102] 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.
2014-04-29 11:19 +0000 [r413088] 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/
2014-04-28 20:01 +0000 [r413073] 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.
2014-04-25 17:48 +0000 [r413009] 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
2014-04-24 14:37 +0000 [r412992] 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/
2014-04-23 20:06 +0000 [r412976] 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/
2014-04-23 18:00 +0000 [r412924] 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
2014-04-22 10:09 +0000 [r412882] 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
2014-04-21 17:54 +0000 [r412823] 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
2014-04-21 16:15 +0000 [r412749] Kinsey Moore <kmoore@digium.com>
* main/manager.c, /, main/http.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
2014-04-21 16:05 +0000 [r412747] Jonathan Rose <jrose@digium.com>
* /, 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
2014-04-21 14:57 +0000 [r412728-412730] Kinsey Moore <kmoore@digium.com>
* 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
* 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
2014-04-21 08:31 +0000 [r412713] 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
2014-04-19 02:13 +0000 [r412657-412698] 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
* /, 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
2014-04-18 20:26 +0000 [r412639-412653] Jonathan Rose <jrose@digium.com>
* 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, 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: ARI: Remove unnecessary \briefs from
automatically generated documentation Review:
https://reviewboard.asterisk.org/r/3440/
* 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,
res/ari/resource_bridges.h, res/stasis/control.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/
2014-04-18 17:16 +0000 [r412587] 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
2014-04-18 16:39 +0000 [r412582] 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
2014-04-18 16:38 +0000 [r412579-412581] Richard Mudgett <rmudgett@digium.com>
* apps/app_originate.c, include/asterisk/pbx.h, main/dial.c,
main/pbx.c: 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/
* 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/
2014-04-18 14:21 +0000 [r412565] 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.
2014-04-17 22:49 +0000 [r412551] Joshua Colp <jcolp@digium.com>
* res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
res/res_pjsip_registrar.c, res/res_pjsip/location.c,
res/res_pjsip/pjsip_configuration.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/
2014-04-17 22:42 +0000 [r412535-412549] Jonathan Rose <jrose@digium.com>
* main/app.c: Fix a silly shadowed variable mistake that was missed
from play tones patch
* 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,
res/ari/resource_bridges.h: 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/
2014-04-17 20:24 +0000 [r412483] 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
2014-04-17 15:16 +0000 [r412453] 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/
2014-04-16 19:13 +0000 [r412439] 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
2014-04-15 18:27 +0000 [r412383-412413] Richard Mudgett <rmudgett@digium.com>
* res/res_parking.c, main/rtp_engine.c, main/stasis_channels.c,
main/features_config.c: Eliminate some more unnecessary
RAII_VAR() uses. RAII_VAR() is not a hammer appropriate to pound
all nails.
* 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, main/pbx.c,
res/res_ari.c, res/res_stasis_playback.c, res/stasis/app.c,
res/res_fax.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.
* 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/
* /, 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
2014-04-15 15:58 +0000 [r412330] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: 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
2014-04-14 15:53 +0000 [r412306] 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
2014-04-11 21:41 +0000 [r412227] 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
2014-04-11 12:35 +0000 [r412193] Kinsey Moore <kmoore@digium.com>
* res/ari/resource_bridges.c, main/bridge.c, main/bridge_basic.c,
include/asterisk/stasis_bridges.h, tests/test_cel.c,
apps/app_confbridge.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
2014-04-11 02:48 +0000 [r412088-412153] Matthew Jordan <mjordan@digium.com>
* build_tools/cflags.xml, /, channels/chan_sip.c,
channels/sip/security_events.c, include/asterisk/astobj2.h,
main/astobj2.c, contrib/scripts/refcounter.py (added),
main/asterisk.c: 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. 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
* 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.
2014-04-10 21:07 +0000 [r412074] 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.
2014-04-09 20:32 +0000 [r412048] 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)
2014-04-09 18:16 +0000 [r412034] Kinsey Moore <kmoore@digium.com>
* res/res_stasis_answer.c: res_stasis_answer: Add missing newlines
2014-04-08 21:23 +0000 [r411945-411985] 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
* 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
* 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
2014-04-08 14:48 +0000 [r411927] 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
2014-04-07 20:39 +0000 [r411883] 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.
2014-04-07 16:02 +0000 [r411868] 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/
2014-04-07 14:55 +0000 [r411809-411811] 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
* configs/res_odbc.conf.sample, /, UPGRADE.txt: configs: Clean up
long line and typo in res_odbc.conf.sample. ........ Merged
revisions 411807 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 411808 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-04-07 14:28 +0000 [r411790-411804] 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.
* 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.
2014-04-04 19:02 +0000 [r411701-411717] 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:
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(). 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
* main/stasis_cache.c, main/utils.c, res/res_musiconhold.c,
main/channel.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/
2014-04-04 15:11 +0000 [r411687] 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.
2014-04-03 11:57 +0000 [r411670] 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/
2014-04-03 11:43 +0000 [r411668] 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.
2014-04-01 22:41 +0000 [r411636-411638] 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.
* main/stasis_channels.c: stasis_channels.c: Eliminate another
overuse of RAII_VAR().
2014-04-01 16:51 +0000 [r411586] 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
2014-04-23 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.2.0 Released.
2014-04-21 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.2.0-rc3 Released.
* 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.
* 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.
2014-04-14 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.2.0-rc2 Released.
* 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
* 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.
* 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.
* 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.
* 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.
* 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.
2014-03-28 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.2.0-rc1 Released.
2014-03-28 18:09 +0000 [r411534] Matthew Jordan <mjordan@digium.com>
* include/asterisk/res_hep.h (added), res/res_hep_pjsip.c (added),
res/res_hep.exports.in (added), CHANGES, configs/hep.conf.sample
(added), 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/
2014-03-28 17:52 +0000 [r411532] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
addons/chan_ooh323.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
2014-03-28 17:35 +0000 [r411529] Matthew Jordan <mjordan@digium.com>
* rest-api/api-docs/applications.json,
rest-api/api-docs/playbacks.json, UPGRADE.txt,
rest-api/api-docs/channels.json, 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/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: 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
2014-03-28 17:08 +0000 [r411514] Mark Michelson <mmichelson@digium.com>
* 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.
2014-03-28 16:48 +0000 [r411512] Matthew Jordan <mjordan@digium.com>
* /, res/res_odbc.exports.in, UPGRADE.txt, res/res_odbc.c,
configs/res_odbc.conf.sample, include/asterisk/res_odbc.h,
res/res_config_odbc.c: res_config_odbc/res_odbc: Fix handling of
non-text columns updates with empty values. 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.
This patch also adds a compatibility setting in res_odbc.conf,
allow_empty_string_in_nontext. It is enabled by default. It
should be disabled for database backends (such as PostgreSQL)
that require NULL instead of an empty string for Integer columns.
Review: https://reviewboard.asterisk.org/r/3375 (issue
ASTERISK-23459) Reported by: zvision patches:
res_config_odbc.diff uploaded by zvision (License 5755) ........
Merged revisions 411399 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 411408 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-03-28 16:17 +0000 [r411465] 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
2014-03-28 16:00 +0000 [r411374-411461] Matthew Jordan <mjordan@digium.com>
* /: Remove block on 411408
* /, 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
* 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.
* 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
2014-03-27 19:15 +0000 [r411311-411315] Corey Farrell <git@cfware.com>
* main/message.c, apps/app_jack.c, funcs/func_dialplan.c,
channels/chan_sip.c, funcs/func_math.c,
funcs/func_jitterbuffer.c, res/res_mutestream.c,
funcs/func_global.c, apps/app_speech_utils.c,
res/res_pjsip_header_funcs.c, funcs/func_callcompletion.c,
funcs/func_blacklist.c, funcs/func_cdr.c, funcs/func_channel.c,
apps/app_stack.c, funcs/func_callerid.c, res/res_calendar.c,
apps/app_voicemail.c, funcs/func_speex.c, /,
funcs/func_strings.c, res/res_xmpp.c, res/res_jabber.c,
main/features_config.c, channels/chan_iax2.c,
apps/confbridge/conf_config_parser.c,
channels/pjsip/dialplan_functions.c, funcs/func_groupcount.c,
funcs/func_pitchshift.c, funcs/func_odbc.c, funcs/func_volume.c,
funcs/func_frame_trace.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
* include/asterisk.h, /, main/format.c: 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
2014-03-27 14:20 +0000 [r411295] 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
2014-03-26 22:44 +0000 [r411245] 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
2014-03-26 16:07 +0000 [r411193] Jonathan Rose <jrose@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: 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
2014-03-26 16:03 +0000 [r411191] 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.
2014-03-25 18:43 +0000 [r411173] 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/
2014-03-25 17:52 +0000 [r411157-411159] Mark Michelson <mmichelson@digium.com>
* tests/test_sorcery.c, tests/test_sorcery_realtime.c,
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:
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
* 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.
2014-03-25 16:55 +0000 [r411141] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/res_pjsip.h, res/res_pjsip/pjsip_options.c,
res/res_pjsip.c: 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/
2014-03-25 16:04 +0000 [r411091] 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
2014-03-25 15:44 +0000 [r411086] 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/
2014-03-24 21:38 +0000 [r411023] 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
2014-03-21 16:01 +0000 [r410995] 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'.
2014-03-20 22:54 +0000 [r410966] 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
2014-03-20 16:27 +0000 [r410949] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, res/ari/resource_channels.c,
res/res_stasis_snoop.c, include/asterisk/rtp_engine.h,
main/dial.c, main/manager.c, main/channel_internal_api.c,
main/core_unreal.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/
2014-03-19 17:26 +0000 [r410933] 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.
2014-03-19 14:24 +0000 [r410904-410918] Joshua Colp <jcolp@digium.com>
* res/res_stasis.c: res_stasis: Fix a bug where the default bridge
type was not set.
* 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/
2014-03-19 02:29 +0000 [r410890] Matthew Jordan <mjordan@digium.com>
* res/res_ari.c: res_ari: Fix documentation schema error
2014-03-18 23:31 +0000 [r410876] 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
2014-03-18 15:28 +0000 [r410861] 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.
2014-03-18 14:51 +0000 [r410858] 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/
2014-03-18 12:45 +0000 [r410844] 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!
2014-03-18 11:51 +0000 [r410830] 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
2014-03-18 02:02 +0000 [r410813] Richard Mudgett <rmudgett@digium.com>
* main/stasis_cache.c: stasis_cache: Use the right variable in the
cache entry ao2 cmp function.
2014-03-17 22:53 +0000 [r410793-410795] Joshua Colp <jcolp@digium.com>
* CHANGES, res/res_pjsip/include/res_pjsip_private.h,
res/res_pjsip.c, main/dns.c, res/res_pjsip/config_system.c,
include/asterisk/dns.h: 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/
* 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/
2014-03-17 21:56 +0000 [r410747-410750] Russ Meyerriecks <rmeyerreicks@digium.com>
* /, main/callerid.c: !fixup: callerid: Logic error in checksum
processing Fixes syntax error in previous commit :-( ........
Merged revisions 410748 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 410749 from
http://svn.asterisk.org/svn/asterisk/branches/11
* 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) ........ Merged revisions 410710
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 410717 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-03-17 18:36 +0000 [r410673-410696] 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.
* 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,
bridges/bridge_softmix.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
2014-03-17 16:42 +0000 [r410671] Richard Mudgett <rmudgett@digium.com>
* apps/confbridge/conf_chan_announce.c: app_confbridge: Add missing
destructor call to announcer channel destructor.
2014-03-16 20:20 +0000 [r410650] 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)
2014-03-14 21:55 +0000 [r410625] 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.
2014-03-14 21:28 +0000 [r410623] 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
2014-03-14 20:53 +0000 [r410590-410607] 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
* 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
* 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
2014-03-14 17:56 +0000 [r410588] 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/
2014-03-14 16:26 +0000 [r410574] Jonathan Rose <jrose@digium.com>
* CHANGES, res/res_pjsip/config_transport.c,
include/asterisk/acl.h, main/acl.c,
res/res_pjsip/pjsip_configuration.c: 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/
2014-03-14 16:11 +0000 [r410559] 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
2014-03-14 16:05 +0000 [r410558] 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/
2014-03-14 15:55 +0000 [r410541-410555] 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/
* 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.
2014-03-13 19:30 +0000 [r410527] Kinsey Moore <kmoore@digium.com>
* res/stasis/control.c, res/stasis/control.h, res/res_stasis.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/
2014-03-13 13:24 +0000 [r410509-410510] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_multihomed.c: res_pjsip_multihomed: Remove change
for testing fix.
* res/res_pjsip_multihomed.c: res_pjsip_multihomed: Fix a bug where
the 200 OK for a REGISTER would contain the wrong contact.
2014-03-12 19:05 +0000 [r410491-410493] 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/
* 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
2014-03-12 12:50 +0000 [r410451-410471] 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.
* 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/
2014-03-10 17:16 +0000 [r410383] 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
2014-03-10 16:32 +0000 [r410368] Scott Griepentrog <sgriepentrog@digium.com>
* main/manager.c, res/ari/resource_channels.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.
2014-03-10 13:25 +0000 [r410329] 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
2014-03-10 12:52 +0000 [r410306] 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
2014-03-08 16:41 +0000 [r410287] George Joseph <george.joseph@fairview5.com>
* res/res_pjsip/config_transport.c, main/sorcery.c,
include/asterisk/res_pjsip.h, 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:
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/
2014-03-08 15:43 +0000 [r410274] 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.
2014-03-07 22:53 +0000 [r410226] 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
2014-03-07 22:07 +0000 [r410211] 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.
2014-03-07 21:53 +0000 [r410194-410209] 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
* 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.
2014-03-07 21:10 +0000 [r410190] 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/
2014-03-07 21:03 +0000 [r410187] Mark Michelson <mmichelson@digium.com>
* tests/test_sorcery_realtime.c, main/sorcery.c,
res/res_sorcery_realtime.c, include/asterisk/sorcery.h: 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
2014-03-07 20:28 +0000 [r410171-410184] 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/
* 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)
2014-03-07 15:46 +0000 [r410157] Scott Griepentrog <sgriepentrog@digium.com>
* addons/chan_mobile.c, main/bridge_channel.c,
channels/chan_pjsip.c, channels/chan_mgcp.c,
channels/chan_unistim.c, res/res_calendar_icalendar.c,
main/pbx.c, channels/chan_bridge_media.c, main/ccss.c,
main/bridge.c, tests/test_stasis_channels.c,
apps/app_originate.c, apps/app_bridgewait.c,
res/parking/parking_applications.c, include/asterisk/channel.h,
res/res_calendar_caldav.c, apps/app_queue.c, apps/app_followme.c,
main/cel.c, res/res_ari_channels.c,
rest-api/api-docs/bridges.json, res/res_calendar_ews.c,
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,
include/asterisk/core_unreal.h, res/res_calendar.c,
addons/chan_ooh323.c, channels/chan_sip.c, res/stasis/control.c,
main/channel_internal_api.c, include/asterisk/stasis_app.h,
channels/chan_console.c, res/res_stasis_snoop.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, tests/test_cdr.c,
res/res_clioriginate.c, res/res_ari_bridges.c,
tests/test_substitution.c, channels/chan_multicast_rtp.c,
res/res_stasis_playback.c, apps/app_meetme.c,
apps/confbridge/conf_chan_record.c, tests/test_app.c,
include/asterisk/channel_internal.h, main/bridge_basic.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,
include/asterisk/pbx.h, res/ari/resource_channels.c,
res/res_stasis.c, include/asterisk/bridge.h,
res/ari/resource_channels.h, apps/app_voicemail.c,
apps/app_dial.c, res/res_calendar_exchange.c,
channels/chan_vpb.cc, apps/app_page.c, apps/app_chanisavail.c,
main/core_local.c, include/asterisk/dial.h,
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:
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/
2014-03-07 04:51 +0000 [r410107] 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
2014-03-06 23:40 +0000 [r410090] 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
2014-03-06 23:35 +0000 [r410089] 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().
2014-03-06 18:50 +0000 [r410028] Jonathan Rose <jrose@digium.com>
* 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,
include/asterisk/acl.h: 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/
2014-03-06 18:18 +0000 [r410025] Joshua Colp <jcolp@digium.com>
* res/res_stasis_recording.c, 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_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/
2014-03-06 15:43 +0000 [r410011] 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.
2014-03-06 15:13 +0000 [r410006] George Joseph <george.joseph@fairview5.com>
* 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, res/res_pjsip/config_auth.c,
funcs/func_sorcery.c (added), res/res_pjsip/location.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 02:05 +0000 [r409991] 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
2014-03-06 00:32 +0000 [r409967-409976] Richard Mudgett <rmudgett@digium.com>
* apps/confbridge/conf_state_inactive.c,
apps/confbridge/conf_state_multi.c: app_confbridge: Remove some
noop code.
* res/res_musiconhold.c: res_musiconhold.c: Remove some unnecessary
RAII_VAR() usage. * Made the moh_register() define use useful
parameter names.
2014-03-05 20:40 +0000 [r409900-409918] 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
* 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
2014-03-05 18:46 +0000 [r409887] 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
2014-03-05 18:40 +0000 [r409885] Richard Mudgett <rmudgett@digium.com>
* contrib/ast-db-manage/cdr/versions (added),
contrib/ast-db-manage/cdr/versions/210693f3123d_create_cdr_table.py
(added),
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 (added),
contrib/ast-db-manage/cdr (added),
contrib/ast-db-manage/cdr/script.py.mako (added): 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/
2014-03-05 16:57 +0000 [r409835] David M. Lee <dlee@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
main/config.c: 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
2014-03-05 12:05 +0000 [r409779] 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
2014-03-05 06:30 +0000 [r409746-409762] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: Correct RTP handling in chan_unistim
and fix transfer process broken in previous fix: - Fixed too
early RTP setup with phone, that cause no ringback tone on caller
side - Handle call transfer cancel only in STATE_CALL case
(related to ASTERISK-23073) (Reported by: Németh Tamás, niurkin
sil) ........ Merged revisions 409761 from
http://svn.asterisk.org/svn/asterisk/branches/11
* 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 04:59 +0000 [r409697-409704] 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
........ Merged revisions 409703 from
http://svn.asterisk.org/svn/asterisk/branches/11
* 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
2014-03-05 00:55 +0000 [r409682] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/stasis_internal.h: stasis: Made
internal_stasis_subscribe() prototype and definition match
exactly.
2014-03-04 19:34 +0000 [r409626] 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
2014-03-04 17:07 +0000 [r409570] 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
2014-03-04 16:53 +0000 [r409568] 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
2014-03-04 14:54 +0000 [r409474] 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
2014-03-03 19:44 +0000 [r409422] Joshua Colp <jcolp@digium.com>
* res/res_stasis_recording.c: res_stasis_recording: Fix memory leak
of the absolute name.
2014-03-03 02:08 +0000 [r409363] 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
2014-03-02 15:14 +0000 [r409346] 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
2014-03-01 20:27 +0000 [r409287] 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.
2014-03-01 00:04 +0000 [r409256-409274] Richard Mudgett <rmudgett@digium.com>
* main/devicestate.c: devicestate.c: Simplified some logic in
_ast_device_state().
* main/stasis_cache.c: stasis_cache.c: Remove some unnecessary
RAII_VAR() usage.
* 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().
* /, 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
2014-02-28 21:24 +0000 [r409234] 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.
2014-02-28 18:02 +0000 [r409158] 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
2014-02-27 19:42 +0000 [r409131] 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
2014-02-27 16:25 +0000 [r409087] 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
2014-02-27 16:05 +0000 [r409054] Corey Farrell <git@cfware.com>
* res/res_fax.c, configs/res_fax.conf.sample, /: res_fax: Warn that
minrate=2400 is not valid for V.27 instead of failing load.
Change minrate from 2400 to 4800 on config reload in response to
changes from ASTERISK-22790 only. Any config with minrate of 2400
that would fail before r405693 will still fail. 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
2014-02-27 12:28 +0000 [r408999] 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/
2014-02-26 23:33 +0000 [r408983] 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().
2014-02-26 19:00 +0000 [r408970] 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/
2014-02-26 17:03 +0000 [r408957] 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/
2014-02-26 13:46 +0000 [r408941-408943] Kinsey Moore <kmoore@digium.com>
* include/asterisk/res_pjsip_session.h: PJSIP: Fix some bad spacing
* 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
2014-02-25 17:50 +0000 [r408880-408882] 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/
* 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/
2014-02-25 17:43 +0000 [r408878] 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
2014-02-22 19:56 +0000 [r408855] 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
2014-02-22 17:57 +0000 [r408839] 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
2014-02-22 02:29 +0000 [r408787] 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
2014-02-21 18:34 +0000 [r408730] 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
2014-02-21 18:17 +0000 [r408711-408715] 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.
* res/res_stasis_recording.c, main/stasis_channels.c,
res/res_sorcery_astdb.c, include/asterisk/json.h, main/sorcery.c,
res/ari/resource_endpoints.c, apps/app_meetme.c, res/res_fax.c:
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().
* 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.
2014-02-21 16:20 +0000 [r408644-408649] 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
* 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
2014-02-21 10:42 +0000 [r408591] 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
2014-02-21 00:49 +0000 [r408538] 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
2014-02-20 21:09 +0000 [r408518-408522] 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/
* 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/
* 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, include/asterisk/sorcery.h:
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/
2014-02-20 19:02 +0000 [r408502] 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.
2014-02-20 02:43 +0000 [r408449] 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
2014-02-19 19:07 +0000 [r408385-408389] 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
* 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/
2014-02-19 12:00 +0000 [r408314-408331] 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
* 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
2014-02-18 19:18 +0000 [r408297] Richard Mudgett <rmudgett@digium.com>
* 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,
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: alembic: Add svn:ignore
*.pyc to directories and svn:executable to *.py files.
2014-02-17 15:21 +0000 [r408270] 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.
2014-02-16 03:23 +0000 [r408194-408220] 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
* 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
2014-02-14 21:44 +0000 [r408138-408140] 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)
* 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
2014-02-14 13:29 +0000 [r408085] 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
2014-02-14 12:39 +0000 [r408069] 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/
2014-02-13 18:50 +0000 [r407988-408005] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_mwi.c, res/res_pjsip_pubsub.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
* 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.
2014-02-12 08:18 +0000 [r407968] 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/
2014-02-11 03:16 +0000 [r407937] 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/
2014-02-10 16:43 +0000 [r407875] 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
2014-02-10 15:54 +0000 [r407858] Kinsey Moore <kmoore@digium.com>
* 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,
apps/app_confbridge.c: 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
2014-02-07 20:48 +0000 [r407766] 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
2014-02-07 20:09 +0000 [r407747-407750] 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
* 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.
2014-02-07 18:18 +0000 [r407729] Richard Mudgett <rmudgett@digium.com>
* configs/iax.conf.sample, /, channels/chan_iax2.c,
include/asterisk/frame.h: 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
2014-02-07 16:46 +0000 [r407676] 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.
2014-02-07 13:13 +0000 [r407624] 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
2014-03-03 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.1.0 Released.
2014-03-01 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.1.0-rc3 Released.
* 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
2013-02-27 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.1.0-rc2 Released.
* 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/
* res_fax: Warn that minrate=2400 is not valid for V.27 instead of
failing load.
Change minrate from 2400 to 4800 on config reload in response to
changes from ASTERISK-22790 only. Any config with minrate of
2400 that would fail before r405693 will still fail.
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/
* 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
* 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.
* 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
* 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.
* 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.
2013-02-06 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.1.0-rc1 Released.
2014-02-06 20:06 +0000 [r407589] 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.
2014-02-06 19:57 +0000 [r407587] 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/
2014-02-06 17:54 +0000 [r407572] 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.
2014-02-06 17:06 +0000 [r407568] 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/
2014-02-06 16:53 +0000 [r407567] Mark Michelson <mmichelson@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.
2014-02-05 23:03 +0000 [r407513] 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
2014-02-05 20:55 +0000 [r407461] 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.
2014-02-05 20:43 +0000 [r407458] 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
2014-02-05 19:41 +0000 [r407442] 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
2014-02-05 17:27 +0000 [r407423] Kinsey Moore <kmoore@digium.com>
* UPGRADE.txt: UPGRADE: Note change in behavior for device state
subscriptions
2014-02-05 17:12 +0000 [r407419] Jonathan Rose <jrose@digium.com>
* CHANGES: CHANGES: Update changes log to include new bridge fields
added in r404042
2014-02-05 14:22 +0000 [r407389-407402] Matthew Jordan <mjordan@digium.com>
* UPGRADE.txt, rest-api/api-docs/channels.json,
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/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: 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.
* 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.
2014-02-04 20:08 +0000 [r407274-407339] 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
* 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/
* 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
2014-02-04 16:54 +0000 [r407259] 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)
2014-02-04 02:21 +0000 [r407213] 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
2014-02-04 02:04 +0000 [r407197] 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)
2014-02-03 01:14 +0000 [r407166] 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.
2014-02-01 16:23 +0000 [r407153] Joshua Colp <jcolp@digium.com>
* res/ari/ari_model_validators.c, res/res_stasis.c,
main/stasis_bridges.c, res/ari/ari_model_validators.h,
rest-api/api-docs/events.json, res/stasis/app.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/
2014-02-01 00:24 +0000 [r407104] coreyfarrell <coreyfarrell@localhost>:
* /, 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
2014-01-31 23:40 +0000 [r407082-407084] Matthew Jordan <mjordan@digium.com>
* main/manager_channels.c, apps/app_dial.c, main/cdr.c, main/pbx.c,
main/bridge_after.c, UPGRADE.txt: 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
* 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
2014-01-31 23:14 +0000 [r407034-407036] Kevin Harwell <kharwell@digium.com>
* contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py
(added), configs/pjsip.conf.sample, UPGRADE.txt,
res/res_pjsip_logger.c, CHANGES, res/res_pjsip.c,
include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c:
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/
* 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
2014-01-31 22:38 +0000 [r407031] Mark Michelson <mmichelson@digium.com>
* include/asterisk/res_pjsip_presence_xml.h (added): Add file that
apparently got missed in the merge.
2014-01-31 22:17 +0000 [r407019] Kevin Harwell <kharwell@digium.com>
* UPGRADE.txt,
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:
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/
2014-01-31 22:11 +0000 [r407016] Mark Michelson <mmichelson@digium.com>
* 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,
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: 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
2014-01-31 22:05 +0000 [r407014] Kevin Harwell <kharwell@digium.com>
* 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/
2014-01-31 15:01 +0000 [r407000] 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
2014-01-31 05:28 +0000 [r406987] 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.
2014-01-30 20:34 +0000 [r406935] coreyfarrell <coreyfarrell@localhost>:
* 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
2014-01-30 17:33 +0000 [r406919] 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
2014-01-29 00:42 +0000 [r406862] 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
2014-01-28 23:35 +0000 [r406788-406847] 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
* 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
* 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)
2014-01-28 19:13 +0000 [r406775] 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
2014-01-28 16:41 +0000 [r406723] 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
2014-01-28 00:11 +0000 [r406707] 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.
2014-01-27 20:36 +0000 [r406574-406645] 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
* main/channel.c, /, main/file.c, include/asterisk/channel.h:
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
2014-01-26 23:03 +0000 [r406516] 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
2014-01-26 02:10 +0000 [r406489] 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/
2014-01-24 23:29 +0000 [r406401-406465] 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
* 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
2014-01-24 21:25 +0000 [r406389] 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
2014-01-24 18:04 +0000 [r406342] 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
2014-01-23 23:41 +0000 [r406327] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_pidf.c: Today is not my day for writing code that
compiles.
2014-01-23 22:54 +0000 [r406311] 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)
2014-01-23 21:09 +0000 [r406294-406295] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_pidf.c: 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.
* res/res_pjsip_pidf.c: 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.
2014-01-22 22:23 +0000 [r406264] Scott Griepentrog <sgriepentrog@digium.com>
* utils/extconf.c, main/pbx.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
2014-01-22 19:34 +0000 [r406152-406223] 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
* /, 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
* 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
2014-01-21 21:47 +0000 [r406133] 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.
2014-01-21 21:06 +0000 [r406081] Walter Doekes <walter+asterisk@wjd.nu>
* configs/manager.conf.sample, main/manager.c, /: 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
2014-01-21 20:20 +0000 [r406003-406049] 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
* 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.
* rest-api-templates/ari_resource.h.mustache,
res/res_ari_device_states.c, res/res_ari_mailboxes.c,
res/res_ari_asterisk.c,
rest-api-templates/res_ari_resource.c.mustache,
res/res_ari_applications.c,
rest-api-templates/body_parsing.mustache (added),
res/res_ari_channels.c, res/ari/resource_playbacks.h,
rest-api-templates/param_parsing.mustache,
res/ari/resource_sounds.h, res/ari/resource_bridges.h,
res/ari/resource_device_states.h, res/ari/resource_mailboxes.h,
rest-api/api-docs/channels.json, res/ari/resource_asterisk.h,
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: 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
2014-01-20 23:18 +0000 [r405982] 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.
2014-01-20 22:15 +0000 [r405928] 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
2014-01-20 21:53 +0000 [r405924] 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.
2014-01-20 18:07 +0000 [r405908] 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/
2014-01-18 23:57 +0000 [r405893] 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.
2014-01-17 22:05 +0000 [r405877] Sean Bright <sean@malleable.com>
* channels/chan_sip.c: Make sure the maxptime attribute is added to
the correct offers.
2014-01-17 21:32 +0000 [r405861-405875] Scott Griepentrog <sgriepentrog@digium.com>
* main/frame.c, include/asterisk/format_pref.h,
res/res_pjsip_sdp_rtp.c, main/format_pref.c, main/sorcery.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/
* 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/
2014-01-17 18:54 +0000 [r405777-405843] Rusty Newton <rnewton@digium.com>
* res/res_pjsip.c: Fixing some XML syntax issues with my previous
commit at r405777 for ASTERISK-23071
* /, 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
* 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
2014-01-16 20:05 +0000 [r405746-405748] 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/
* 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
2014-01-16 19:32 +0000 [r405743] 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
2014-01-16 19:06 +0000 [r405643-405694] Kevin Harwell <kharwell@digium.com>
* /, res/res_fax.c, configs/res_fax.conf.sample, UPGRADE.txt:
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
* 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/
2014-01-15 16:48 +0000 [r405585-405587] Jonathan Rose <jrose@digium.com>
* CHANGES: Remove subversion conflict tag accidentally left in
CHANGES
* CHANGES: Include CHANGES info for r405553
2014-01-15 16:36 +0000 [r405583] 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
2014-01-15 13:14 +0000 [r405565] Kinsey Moore <kmoore@digium.com>
* res/res_pjsip_messaging.c, UPGRADE.txt, res/res_pjsip_t38.c,
res/res_pjsip_caller_id.c, CHANGES,
res/res_pjsip/pjsip_options.c, res/res_pjsip.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, 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:
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/
2014-01-14 23:26 +0000 [r405553] Jonathan Rose <jrose@digium.com>
* rest-api/resources.json, res/ari/ari_model_validators.c,
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): 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/
2014-01-14 21:44 +0000 [r405541] Richard Mudgett <rmudgett@digium.com>
* main/strings.c: string container: Remove unnecessary RAII_VAR
usage and string object lock.
2014-01-14 18:13 +0000 [r405435] 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
2014-01-14 18:03 +0000 [r405432] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/logger.h, main/pbx.c, main/manager.c, /,
funcs/func_timeout.c, apps/app_dumpchan.c, main/logger.c,
UPGRADE.txt, apps/app_verbose.c, main/asterisk.c,
configs/logger.conf.sample, main/cli.c: 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
2014-01-14 03:12 +0000 [r405367] 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.
2014-01-13 17:09 +0000 [r405350] Jonathan Rose <jrose@digium.com>
* res/res_pjsip_session.c: PJSIP: Backport r405270 - Unhold on
reinvite without SDP Adds behavior to unhold on a reinvite
without an SDP section Review:
https://reviewboard.asterisk.org/r/3106/
2014-01-13 13:28 +0000 [r405338] 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
2014-01-12 22:23 +0000 [r405325] 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/
2014-01-12 21:58 +0000 [r405311-405312] Matthew Jordan <mjordan@digium.com>
* funcs/func_cdr.c, include/asterisk/cdr.h, apps/app_cdr.c,
main/cdr.c, apps/app_forkcdr.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/
* 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/
2014-01-10 19:39 +0000 [r405298] 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
2014-01-10 18:00 +0000 [r405282] 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
2014-01-09 23:45 +0000 [r405268] 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.
2014-01-09 23:36 +0000 [r405266] Kevin Harwell <kharwell@digium.com>
* res/res_pjsip_messaging.c, res/res_pjsip.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/
2014-01-09 20:25 +0000 [r405253] 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
2014-01-09 16:51 +0000 [r405235] 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
2014-01-09 15:49 +0000 [r405216] 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/ (closes issue AST-1258)
Reported by: Steve Pitts ........ Merged revisions 405215 from
http://svn.asterisk.org/svn/asterisk/branches/11
2014-01-09 14:14 +0000 [r405162] 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
2014-01-08 16:48 +0000 [r405131] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip/security_events.c: Use proper case for checking if
digest authentication is used.
2014-01-08 16:28 +0000 [r405083-405124] 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
* /, 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
2014-01-07 19:55 +0000 [r405019-405034] 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.
* 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
2014-01-06 21:54 +0000 [r404952-405006] Richard Mudgett <rmudgett@digium.com>
* apps/app_voicemail.c: app_voicemail: Explicitly set
defaultenabled=yes
* 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/
* apps/app_voicemail.c, res/res_mwi_external.c (added),
configs/sorcery.conf.sample, include/asterisk/res_mwi_external.h
(added), res/res_mwi_external.exports.in (added): 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/
2014-01-05 16:00 +0000 [r404923-404935] 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.
* 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
2014-01-04 10:42 +0000 [r404911] 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
2014-01-03 21:59 +0000 [r404859] 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
2014-01-03 21:45 +0000 [r404843-404855] 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.
* 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
2014-01-03 19:00 +0000 [r404781-404786] Kevin Harwell <kharwell@digium.com>
* 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
* /, 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
2014-01-03 18:24 +0000 [r404764] Richard Mudgett <rmudgett@digium.com>
* tests/test_stasis.c: test_stasis.c: Fix ref leak in normal
execution path.
2014-01-03 17:25 +0000 [r404725-404737] 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.
* 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
2014-01-02 19:37 +0000 [r404676] 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
2014-01-02 19:06 +0000 [r404663] 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/
2013-12-31 22:49 +0000 [r404613-404652] 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
* 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
2013-12-31 21:38 +0000 [r404605] 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
2013-12-31 20:26 +0000 [r404592] 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
2013-12-30 23:21 +0000 [r404581] 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
2013-12-24 17:10 +0000 [r404565-404568] David M. Lee <dlee@digium.com>
* UPGRADE.txt: Added note to UPGRADE.txt about the default value of
live_dangerously changing
* 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/
2013-12-24 02:19 +0000 [r404553] 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
2013-12-21 03:34 +0000 [r404531] 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.
2013-12-20 22:02 +0000 [r404509] David M. Lee <dlee@digium.com>
* res/ari/resource_channels.h, rest-api/api-docs/applications.json,
rest-api/api-docs/channels.json, res/ari/resource_channels.c,
res/res_ari_channels.c: 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
2013-12-20 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 12.0.0 Released.
2013-12-20 22:02 +0000 [r404509] 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
2013-12-20 21:25 +0000 [r404480-404488] Matthew Jordan <mjordan@digium.com>
* /: Remove automerge properties
* 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,
include/asterisk/config.h, main/config.c, main/channel.c,
res/res_pjsip/location.c, include/asterisk/res_pjsip_cli.h
(added): 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)
2013-12-20 21:16 +0000 [r404458] 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
2013-12-20 20:11 +0000 [r404439] 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.
2013-12-20 20:04 +0000 [r404437] 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".
2013-12-20 19:52 +0000 [r404434] Richard Mudgett <rmudgett@digium.com>
* res/res_pjsip/location.c, tests/test_cel.c,
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:
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/
2013-12-20 19:17 +0000 [r404421] 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.
2013-12-20 19:06 +0000 [r404419] Richard Mudgett <rmudgett@digium.com>
* main/sounds_index.c: Whitespace fixes.
2013-12-20 17:21 +0000 [r404405] 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/
2013-12-19 18:15 +0000 [r404375] Richard Mudgett <rmudgett@digium.com>
* CHANGES: Put notice in CHANGES as well as UPGRADE.txt.
2013-12-19 17:58 +0000 [r404369-404371] 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.
* 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.
2013-12-19 17:15 +0000 [r404356] 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
2013-12-19 17:03 +0000 [r404354] 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/
2013-12-19 17:02 +0000 [r404352] 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
2013-12-19 16:37 +0000 [r404348] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.h, channels/chan_h323.c, main/app.c,
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, UPGRADE.txt, res/res_xmpp.c,
configs/skinny.conf.sample, res/res_jabber.c, CHANGES,
channels/chan_iax2.c, channels/h323/chan_h323.h,
channels/sig_pri.c, configs/iax.conf.sample, channels/sig_pri.h,
include/asterisk/app.h, channels/chan_dahdi.c,
channels/chan_skinny.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/
2013-12-19 16:31 +0000 [r404345] 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
2013-12-19 12:17 +0000 [r404332] Joshua Colp <jcolp@digium.com>
* main/channel.c: channel: Add a missing ast_channel_unlock when
allocating a Surrogate channel.
2013-12-19 08:19 +0000 [r404320] 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
2013-12-19 02:53 +0000 [r404306] 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.
2013-12-19 00:47 +0000 [r404294] Matthew Jordan <mjordan@digium.com>
* apps/app_cdr.c, main/cdr.c, apps/app_forkcdr.c, main/pbx.c,
funcs/func_cdr.c, apps/app_disa.c, UPGRADE.txt,
include/asterisk/cdr.h, CHANGES: 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/
2013-12-19 00:29 +0000 [r404292] 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.
2013-12-18 22:50 +0000 [r404279] 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
2013-12-18 20:57 +0000 [r404263] 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
2013-12-18 20:46 +0000 [r404237-404261] 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
* main/pickup.c, include/asterisk/aoc.h,
include/asterisk/stasis_bridges.h, apps/app_disa.c,
apps/app_userevent.c, include/asterisk/channelstate.h,
channels/chan_console.c, 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,
pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c,
main/bridge_channel.c, addons/chan_mobile.c,
res/parking/parking_manager.c, channels/chan_pjsip.c,
tests/test_cdr.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, main/cel.c,
apps/app_queue.c, channels/sig_pri.c, main/stasis_bridges.c,
channels/chan_jingle.c, main/dial.c, channels/chan_dahdi.c,
channels/chan_phone.c, include/asterisk/stasis_channels.h,
channels/sig_analog.c, 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: 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
2013-12-18 19:20 +0000 [r404204] Joshua Colp <jcolp@digium.com>
* main/channel.c, channels/chan_dahdi.c, channels/chan_phone.c,
channels/chan_skinny.c, res/parking/parking_tests.c,
tests/test_voicemail_api.c, channels/chan_motif.c,
channels/chan_alsa.c, main/message.c, addons/chan_mobile.c,
tests/test_cdr.c, channels/chan_mgcp.c, main/pbx.c,
channels/chan_sip.c, tests/test_app.c,
apps/confbridge/conf_chan_record.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, apps/app_voicemail.c, res/res_calendar.c,
channels/chan_unistim.c, tests/test_substitution.c,
addons/chan_ooh323.c, channels/chan_vpb.cc,
channels/chan_multicast_rtp.c, apps/app_meetme.c,
res/res_stasis_snoop.c, channels/chan_gtalk.c,
channels/chan_iax2.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/
2013-12-18 12:36 +0000 [r404184] Matthew Jordan <mjordan@digium.com>
* 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,
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: ari: Bump the version of ARI to 1.0.0
(closes issue ASTERISK-23007)
2013-12-18 12:00 +0000 [r404137] 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
2013-12-18 00:35 +0000 [r404099] 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
2013-12-18 00:16 +0000 [r404050] Matthew Jordan <mjordan@digium.com>
* LICENSE: LICENSE: Update language to include ARI
2013-12-17 23:50 +0000 [r404048] 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.
2013-12-17 23:36 +0000 [r404046] 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)
2013-12-17 23:17 +0000 [r404042] Jonathan Rose <jrose@digium.com>
* 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,
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: 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/
2013-12-17 18:34 +0000 [r404027-404029] 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.
* 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/
2013-12-17 14:33 +0000 [r404006] David M. Lee <dlee@digium.com>
* configs/asterisk.conf.sample, main/asterisk.c: Changed the
default for live_dangerously to no
2013-12-17 12:51 +0000 [r403993] 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
2013-12-16 18:31 +0000 [r403959] David M. Lee <dlee@digium.com>
* UPGRADE.txt, 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: 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
2013-12-16 18:22 +0000 [r403957] 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)
2013-12-16 16:11 +0000 [r403856-403864] 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
* 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
2013-12-15 01:38 +0000 [r403823] 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.
2013-12-14 17:25 +0000 [r403808-403811] 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/
* 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/
2013-12-13 21:24 +0000 [r403796] Jonathan Rose <jrose@digium.com>
* res/res_pjsip_messaging.c, main/message.c: documentation: Add
PJSIP technology to messaging documentation
2013-12-13 20:06 +0000 [r403782] 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.
2013-12-13 20:04 +0000 [r403781] Jonathan Rose <jrose@digium.com>
* include/asterisk/bridge.h, res/parking/parking_bridge_features.c,
res/parking/parking_manager.c, main/bridge.c,
main/bridge_basic.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/
2013-12-13 19:55 +0000 [r403779-403780] 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. ........ Merged revisions
403726 from http://svn.asterisk.org/svn/asterisk/trunk
* main/app.c, apps/app_voicemail.c, include/asterisk/app.h,
include/asterisk/doxyref.h: 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. ........ Merged revisions 403643 from
http://svn.asterisk.org/svn/asterisk/trunk
2013-12-13 18:24 +0000 [r403749-403767] Kevin Harwell <kharwell@digium.com>
* channels/chan_sip.c, include/asterisk/channel.h,
bridges/bridge_native_rtp.c, channels/chan_pjsip.c,
main/channel.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)
* main/http.c, rest-api/api-docs/channels.json,
res/ari/resource_channels.c, res/res_ari_channels.c,
res/ari/resource_channels.h: 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/
* 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,
res/res_stasis_answer.c, rest-api/api-docs/bridges.json,
res/ari/resource_bridges.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/
2013-12-13 16:27 +0000 [r403748] David M. Lee <dlee@digium.com>
* channels/pjsip: Setting svn:ignore
2013-12-13 05:00 +0000 [r403736] Matthew Jordan <mjordan@digium.com>
* channels/Makefile: channels/Makefile: clean pjsip directory
2013-12-12 19:44 +0000 [r403713] 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/
2013-12-12 19:12 +0000 [r403705] 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
2013-12-11 20:11 +0000 [r403680] Kevin Harwell <kharwell@digium.com>
* res/res_pjsip/pjsip_configuration.c, res/res_pjsip_messaging.c,
res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c,
include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
configs/pjsip.conf.sample: 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/
2013-12-11 19:18 +0000 [r403639] 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
2013-12-11 13:05 +0000 [r403616-403618] Matthew Jordan <mjordan@digium.com>
* funcs/func_channel.c, channels/pjsip/include (added),
channels/pjsip/include/dialplan_functions.h (added),
res/res_pjsip_t38.c, channels/pjsip/include/chan_pjsip.h (added),
channels/Makefile, channels/chan_pjsip.c, main/xmldoc.c,
channels/pjsip/dialplan_functions.c (added),
include/asterisk/res_pjsip_session.h, channels/pjsip (added):
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/
* 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
2013-12-09 22:47 +0000 [r403587] 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/
2013-12-09 19:23 +0000 [r403545-403559] 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 '^'.
* 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().
2013-12-09 18:31 +0000 [r403542] 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
2013-12-09 18:31 +0000 [r403527-403541] Richard Mudgett <rmudgett@digium.com>
* 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.
* main/sorcery.c: sorcery: Whitespace You would think that a new
file would start off without any whitespace oddities.
2013-12-09 16:40 +0000 [r403510] 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
2013-12-09 03:19 +0000 [r403435-403458] 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
* 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.
2013-12-05 20:49 +0000 [r403398] David M. Lee <dlee@digium.com>
* main/core_unreal.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,
include/asterisk/stasis_channels.h, channels/sig_analog.c,
channels/chan_motif.c, res/res_agi.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_disa.c,
apps/app_userevent.c, main/core_local.c, channels/chan_console.c,
include/asterisk/channelstate.h, 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,
channels/chan_alsa.c, pbx/pbx_realtime.c, main/stasis_channels.c,
channels/chan_nbs.c, main/bridge_channel.c, addons/chan_mobile.c,
channels/chan_pjsip.c, tests/test_cdr.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: Reverting r403311. It's causing
ARI tests to hang.
2013-12-04 21:41 +0000 [r403377] 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
2013-12-04 20:53 +0000 [r403364] 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".
2013-12-04 18:40 +0000 [r403349] 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.
2013-12-03 20:17 +0000 [r403342] David M. Lee <dlee@digium.com>
* 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.
2013-12-03 17:59 +0000 [r403329] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_session.c, configure,
include/asterisk/autoconfig.h.in, configure.ac:
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.
2013-12-03 17:23 +0000 [r403324] Richard Mudgett <rmudgett@digium.com>
* main/bucket.c, 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:
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.
2013-12-03 16:37 +0000 [r403312] 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.
2013-12-03 16:33 +0000 [r403311] Mark Michelson <mmichelson@digium.com>
* include/asterisk/stasis_bridges.h, apps/app_disa.c,
apps/app_userevent.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,
channels/chan_alsa.c, pbx/pbx_realtime.c, main/stasis_channels.c,
channels/chan_nbs.c, main/bridge_channel.c, addons/chan_mobile.c,
channels/chan_pjsip.c, tests/test_cdr.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,
main/core_unreal.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,
include/asterisk/stasis_channels.h, channels/sig_analog.c,
channels/chan_motif.c, res/res_agi.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, main/pickup.c,
channels/chan_sip.c, include/asterisk/aoc.h: 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.
2013-12-03 16:32 +0000 [r403310] Joshua Colp <jcolp@digium.com>
* res/res_ari.c: 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.
2013-12-02 18:34 +0000 [r403304] David M. Lee <dlee@digium.com>
* res/res_ari.c: Fixed the filename for the ari.conf docs
2013-12-02 18:03 +0000 [r403290-403291] Alexandr Anikin <may@telecom-service.ru>
* /: remove unwanted property svn:mergeinfo
* /, 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
2013-12-01 21:12 +0000 [r403256-403271] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_session.c: res_pjsip_session: Apply fromuser and
fromdomain to all requests as documented.
* 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.
* res/res_pjsip_transport_websocket.c,
include/asterisk/res_pjsip.h, res/res_pjsip/location.c,
res/res_pjsip/security_events.c, res/res_pjsip/pjsip_options.c,
res/res_pjsip.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/
2013-11-30 14:11 +0000 [r403240] Joshua Colp <jcolp@digium.com>
* res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
rest-api/api-docs/events.json: res_ari: Add Recording events to
the validator.
2013-11-28 02:12 +0000 [r403179-403223] 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
* 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
* 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/
* 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/
* 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
2013-11-27 15:36 +0000 [r403175] 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 [r403160-403173] 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
* 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.
2013-11-23 17:38 +0000 [r403131-403134] Kevin Harwell <kharwell@digium.com>
* 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, res/ari.make,
rest-api/api-docs/applications.json,
include/asterisk/stasis_app_device_state.h (added),
res/ari/resource_device_states.h (added),
res/ari/resource_applications.h, res/res_stasis.c,
include/asterisk/devicestate.h,
res/res_stasis_device_state.exports.in (added),
rest-api/api-docs/events.json, res/res_ari_device_states.c
(added), res/stasis/app.c: 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/
* 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, include/asterisk/sorcery.h,
res/res_pjsip/pjsip_configuration.c, include/asterisk/strings.h,
res/res_pjsip_pubsub.c,
res/res_pjsip/include/res_pjsip_private.h,
res/res_pjsip/config_transport.c, 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_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/
2013-11-23 12:51 +0000 [r403117-403119] Joshua Colp <jcolp@digium.com>
* res/ari/ari_model_validators.h, 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: 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/
* main/audiohook.c, res/ari/resource_channels.c,
res/res_stasis_snoop.c (added), res/res_ari_channels.c,
res/ari/resource_channels.h, res/res_stasis_snoop.exports.in
(added), include/asterisk/stasis_app_snoop.h (added),
rest-api/api-docs/channels.json: 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/
2013-11-22 23:44 +0000 [r403094] 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.
2013-11-22 22:24 +0000 [r403082] 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.
2013-11-22 20:01 +0000 [r403069] Kinsey Moore <kmoore@digium.com>
* 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_app.h,
include/asterisk/stasis.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,
res/res_stasis.c: 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
2013-11-22 17:19 +0000 [r403022] 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/
2013-11-22 17:11 +0000 [r403016] 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
2013-11-21 22:35 +0000 [r402981-402993] 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
* 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.
2013-11-21 19:21 +0000 [r402968] 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!)
2013-11-21 19:08 +0000 [r402944-402956] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/app.h: voicemail: Fixup some doxygen comments.
* main/bucket.c: bucket: Fix scheme ref leak in
__ast_bucket_scheme_register().
2013-11-21 17:52 +0000 [r402940-402941] 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.
* 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).
2013-11-21 15:55 +0000 [r402926] David M. Lee <dlee@digium.com>
* res/stasis/control.c, 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: 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/
2013-11-19 23:17 +0000 [r402891] 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
2013-11-16 13:44 +0000 [r402864] 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
2013-11-15 14:35 +0000 [r402838] 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
2013-11-14 15:01 +0000 [r402817] David M. Lee <dlee@digium.com>
* res/res_stasis.c: stasis: Fixed scoping problem with bridge
tracking.
2013-11-13 23:09 +0000 [r402804] Joshua Colp <jcolp@digium.com>
* res/stasis/control.c, 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_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.
2013-11-12 23:16 +0000 [r402787-402793] 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
* 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
2013-11-12 19:11 +0000 [r402767-402769] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Switch to a scoped lock to avoid missing unlocks
in failure returns.
* 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.
2013-11-12 16:45 +0000 [r402757] 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
2013-11-12 16:33 +0000 [r402755] 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.
2013-11-12 15:26 +0000 [r402738] David M. Lee <dlee@digium.com>
* res/ari/ari_model_validators.h, rest-api/api-docs/events.json:
Fixed a typ.
2013-11-12 15:02 +0000 [r402710] 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
2013-11-11 19:26 +0000 [r402687] 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
2013-11-11 15:36 +0000 [r402647] 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
2013-11-08 23:04 +0000 [r402606] 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
2013-11-08 20:20 +0000 [r402593] 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/
2013-11-08 19:28 +0000 [r402584] Jonathan Rose <jrose@digium.com>
* configs/manager.conf.sample, CHANGES, include/asterisk/manager.h,
main/manager.c, main/security_events.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/
2013-11-08 19:22 +0000 [r402582] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip.c: Clarify an ambiguous error message.
2013-11-08 18:48 +0000 [r402561-402570] David M. Lee <dlee@digium.com>
* res/res_pjsip/config_system.c: res_pjsip: Print a helpful error
message if sorcery registration fails
* res/ari/resource_playbacks.h: Changes from make ari-stubs after
r402560
2013-11-08 17:39 +0000 [r402560] Kevin Harwell <kharwell@digium.com>
* 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), rest-api/resources.json,
res/ari/resource_playback.h (removed), res/res_ari_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
2013-11-08 17:28 +0000 [r402555] 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/
2013-11-07 23:16 +0000 [r402537] 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.
2013-11-07 21:09 +0000 [r402528] David M. Lee <dlee@digium.com>
* rest-api-templates/swagger_model.py, res/ari/resource_asterisk.h,
rest-api-templates/ari_resource.c.mustache,
rest-api-templates/asterisk_processor.py, res/res_ari_bridges.c,
rest-api/api-docs/endpoints.json, res/ari/resource_endpoints.c,
res/ari/resource_endpoints.h, res/res_ari_applications.c,
res/res_ari_playback.c, res/res_ari_channels.c,
rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
res/res_ari_recordings.c, res/ari/resource_bridges.h,
res/res_ari_events.c, 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, res/res_ari_sounds.c,
rest-api/api-docs/recordings.json, res/ari/resource_recordings.c,
res/ari/resource_channels.h,
rest-api-templates/ari_resource.h.mustache,
res/ari/resource_events.c, res/ari/resource_recordings.h,
rest-api-templates/rest_handler.mustache, res/res_ari_asterisk.c,
rest-api-templates/res_ari_resource.c.mustache,
res/ari/resource_events.h, rest-api/api-docs/sounds.json,
res/ari/resource_sounds.c, res/ari/resource_sounds.h,
rest-api/api-docs/asterisk.json,
rest-api/api-docs/applications.json, res/res_ari_endpoints.c,
res/ari/resource_asterisk.c, rest-api/api-docs/playback.json:
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/
2013-11-06 21:57 +0000 [r402517] 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/
2013-11-05 21:16 +0000 [r402501-402507] 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.
* 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
* include/asterisk/netsock2.h, include/asterisk/acl.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.
* 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/
2013-11-04 20:56 +0000 [r402452] 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
2013-11-02 04:30 +0000 [r402398-402438] 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++.
* main/stasis.c, main/stasis_message_router.c,
include/asterisk/vector.h, include/asterisk/lock.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/
* 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
* /, 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
* main/app.c, include/asterisk/app.h: 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.
2013-11-01 21:49 +0000 [r402387] Scott Griepentrog <sgriepentrog@digium.com>
* main/bridge.c, include/asterisk/bridge.h, main/manager_bridges.c:
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/
2013-11-01 16:31 +0000 [r402367] 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)
2013-11-01 14:37 +0000 [r402358] 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/
2013-11-01 12:38 +0000 [r402348] Kinsey Moore <kmoore@digium.com>
* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c, /,
channels/chan_sip.c: 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
2013-11-01 12:31 +0000 [r402336-402346] 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/
* 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/
2013-10-31 22:08 +0000 [r402327] Mark Michelson <mmichelson@digium.com>
* 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,
contrib/scripts/sip_to_pjsip/sip_to_pjsip.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
2013-10-31 16:04 +0000 [r402285-402289] 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
* 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.
2013-10-31 14:43 +0000 [r402276] 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.
2013-10-30 17:52 +0000 [r402265] Kevin Harwell <kharwell@digium.com>
* res/res_pjsip_messaging.c, channels/chan_pjsip.c:
pjsip_messaging: Added debug for in dialog messaging (issue
ASTERISK-22777) Reported by: Matt Jordan
2013-10-29 23:43 +0000 [r402226] 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
2013-10-29 12:53 +0000 [r402154] Matthew Jordan <mjordan@digium.com>
* /, main/translate.c, main/xmldoc.c, main/channel.c, main/pbx.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
2013-10-29 12:51 +0000 [r402148-402152] Kinsey Moore <kmoore@digium.com>
* res/ari/resource_channels.h, rest-api/api-docs/channels.json,
res/ari/resource_channels.c, res/res_ari_channels.c: 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/
* 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)
2013-10-29 11:15 +0000 [r402139] Joshua Colp <jcolp@digium.com>
* res/res_ari_playback.c, rest-api/api-docs/playback.json:
res_ari_playback: Add missing 404 error response for GET and
DELETE. (closes issue ASTERISK-22722) Reported by: Richard
Mudgett
2013-10-28 21:30 +0000 [r402127] David M. Lee <dlee@digium.com>
* doc: Ignore full docs
2013-10-28 15:05 +0000 [r402112-402115] Michael L. Young <elgueromexicano@gmail.com>
* UPGRADE-11.txt, UPGRADE.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.
* UPGRADE.txt, /, channels/chan_sip.c: 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
2013-10-27 23:22 +0000 [r402081-402090] 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.
* main/cdr.c, cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c,
cdr/cdr_sqlite.c, UPGRADE.txt, 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: 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/
2013-10-26 12:55 +0000 [r402064] Joshua Colp <jcolp@digium.com>
* include/asterisk/res_pjsip_session.h, channels/chan_pjsip.c:
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
2013-10-26 00:34 +0000 [r402044-402055] 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.
* 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.
2013-10-25 23:48 +0000 [r402043] Scott Griepentrog <sgriepentrog@digium.com>
* include/asterisk/rtp_engine.h, main/rtp_engine.c, /: 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
2013-10-25 22:02 +0000 [r402003] Richard Mudgett <rmudgett@digium.com>
* res/stasis/app.c: You'd think that new files would be free of
whitespace issues. But you would be wrong.
2013-10-25 21:53 +0000 [r401973-402001] Jonathan Rose <jrose@digium.com>
* rest-api/api-docs/channels.json, res/ari/resource_channels.c,
res/res_ari_channels.c, rest-api/api-docs/bridges.json,
res/ari/resource_bridges.c, res/res_ari_bridges.c: 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/
* 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, res/res_stasis_recording.c,
rest-api/api-docs/channels.json, res/ari/resource_channels.c,
res/ari/ari_model_validators.c: 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/
2013-10-25 20:47 +0000 [r401961] 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
2013-10-25 17:34 +0000 [r401897-401938] 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/
* 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
* /, 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
2013-10-25 16:07 +0000 [r401885] 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
2013-10-25 13:48 +0000 [r401872] 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/
2013-10-24 20:56 +0000 [r401835] 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
2013-10-24 20:34 +0000 [r401706-401831] 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
* 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
* /, 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
* /, 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
* 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
* 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
2013-10-24 03:12 +0000 [r401701] David M. Lee <dlee@digium.com>
* rest-api-templates/ari_model_validators.c.mustache,
rest-api-templates/models.wiki.mustache,
rest-api/api-docs/events.json,
rest-api-templates/swagger_model.py: 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/
2013-10-23 20:02 +0000 [r401621-401662] Jonathan Rose <jrose@digium.com>
* /, 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
* res/res_rtp_asterisk.c, /, main/translate.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
2013-10-23 16:49 +0000 [r401581] 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
2013-10-23 16:37 +0000 [r401578] 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/
2013-10-23 15:23 +0000 [r401539] 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
2013-10-23 11:14 +0000 [r401499] 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
2013-10-23 02:31 +0000 [r401488] 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/
2013-10-22 22:50 +0000 [r401447] 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
2013-10-22 19:03 +0000 [r401420-401434] 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
* 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/
* 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/
* 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/
2013-10-21 21:05 +0000 [r401364] 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)
2013-10-21 19:48 +0000 [r401327] 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
2013-10-21 18:58 +0000 [r401315] David M. Lee <dlee@digium.com>
* rest-api/api-docs/channels.json: Fixing r401281; the model name
is Channel, with a capital C
2013-10-19 21:53 +0000 [r401291] 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/
2013-10-19 14:43 +0000 [r401281] Joshua Colp <jcolp@digium.com>
* res/res_ari_channels.c, res/ari/resource_channels.h, main/pbx.c,
rest-api/api-docs/channels.json, res/ari/resource_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/
2013-10-18 22:51 +0000 [r401271] Richard Mudgett <rmudgett@digium.com>
* res/parking/parking_controller.c: res_parking: Remove setting
useless flag.
2013-10-18 21:49 +0000 [r401261] 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/
2013-10-18 18:33 +0000 [r401248] Mark Michelson <mmichelson@digium.com>
* main/manager.c, main/bridge.c, main/bucket.c, main/sorcery.c,
main/cli.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.
2013-10-18 16:52 +0000 [r401232-401239] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, main/channel.c, res/res_fax.c,
include/asterisk/channel.h: 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/
* include/asterisk/bridge.h: Tweak ast_bridge_depart() doxygen.
2013-10-18 16:05 +0000 [r401212-401223] Mark Michelson <mmichelson@digium.com>
* include/asterisk/bridge.h: Remove the bit about requiring
ast_bridge_depart() to be called before ast_bridge_destroy().
* include/asterisk/bridge.h: Clarify in ast_bridge_destroy() about
how departable channels must be handled.
2013-10-18 15:13 +0000 [r401183] 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
2013-10-18 14:47 +0000 [r401180] 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
2013-10-17 20:37 +0000 [r401168] 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
2013-10-17 18:16 +0000 [r401158] 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
2013-10-17 15:40 +0000 [r401121] 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
2013-10-16 21:20 +0000 [r401107] 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
2013-10-16 21:17 +0000 [r401096-401106] David M. Lee <dlee@digium.com>
* res/res_ari.c: Fixed malformed Access-Control-Allow-Methods
header. Was causing Safari to barf on POST and DELETE.
* rest-api/resources.json: Oops. Leftover /stasis reference
2013-10-16 14:01 +0000 [r401087] Kinsey Moore <kmoore@digium.com>
* res/ari/resource_bridges.h, rest-api/api-docs/channels.json,
rest-api/api-docs/bridges.json, res/ari/resource_channels.h:
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
2013-10-16 12:12 +0000 [r401077] 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
2013-10-16 00:02 +0000 [r401040] Paul Belanger <paul.belanger@polybeacon.com>
* res/res_ari_bridges.c, rest-api/api-docs/bridges.json: Use POST /
DELETE to toggle ARI bridge moh Review:
https://reviewboard.asterisk.org/r/2911/
2013-10-15 20:25 +0000 [r401030] Richard Mudgett <rmudgett@digium.com>
* channels/dahdi/bridge_native_dahdi.c: bridge_native_dahdi: Return
channel join failure if could not make the channels compatible.
2013-10-15 20:02 +0000 [r401018] 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
2013-10-15 20:01 +0000 [r401017] Richard Mudgett <rmudgett@digium.com>
* /, 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
2013-10-15 15:26 +0000 [r400999] 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/
2013-10-15 15:21 +0000 [r400984] 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
2013-10-15 13:43 +0000 [r400958] 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)
2013-10-14 21:55 +0000 [r400911] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_dahdi.h, channels/chan_dahdi.c: 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
2013-10-14 21:52 +0000 [r400910] 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
2013-10-14 15:52 +0000 [r400890] 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/
2013-10-14 14:57 +0000 [r400881] 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.
2013-10-13 15:41 +0000 [r400872] 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
2013-10-12 16:49 +0000 [r400863] 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.
2013-10-11 16:53 +0000 [r400849-400854] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h: channel.h: whitespace changes.
* 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
2013-10-11 16:18 +0000 [r400842-400848] David M. Lee <dlee@digium.com>
* res/ari/resource_playback.h, rest-api/api-docs/playback.json: Fix
a stupid copy/paste error in ARI docs. Patches: ari-doc-patch.txt
uploaded by jbigelow (license 5091)
* res/ari/resource_bridges.h, rest-api/api-docs/channels.json,
rest-api/api-docs/bridges.json, res/ari/resource_channels.h:
Updated /play resource docs. The playback of http: resources
isn't implemented... yet
* rest-api-templates/models.wiki.mustache,
rest-api-templates/api.wiki.mustache,
rest-api-templates/asterisk_processor.py: Correct some ARI wiki
rendering errors
2013-10-10 18:21 +0000 [r400824-400833] Joshua Colp <jcolp@digium.com>
* res/res_pjsip/location.c: Perform validation of permanent
contacts on AORs in res_pjsip.
* 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
2013-10-09 11:00 +0000 [r400771-400812] 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.
* 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)
2013-10-08 22:30 +0000 [r400769] 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
2013-10-08 21:19 +0000 [r400754] 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
2013-10-08 20:51 +0000 [r400749] 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.
2013-10-08 20:16 +0000 [r400724-400742] Richard Mudgett <rmudgett@digium.com>
* apps/app_confbridge.c, CHANGES,
apps/confbridge/conf_config_parser.c,
configs/confbridge.conf.sample, /,
apps/confbridge/include/confbridge.h: 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
* 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
2013-10-08 18:19 +0000 [r400682-400701] 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
* 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
2013-10-08 15:36 +0000 [r400680] 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.
2013-10-08 15:11 +0000 [r400661-400671] 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/
* include/asterisk/stasis.h, apps/app_queue.c,
include/asterisk/manager.h: Miscellaneous stand alone comment
cleanups.
2013-10-06 17:11 +0000 [r400624] 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
2013-10-05 00:41 +0000 [r400588] Richard Mudgett <rmudgett@digium.com>
* channels/iax2/include/parser.h: chan_iax2: Fix compile error.
2013-10-04 21:40 +0000 [r400567] Michael L. Young <elgueromexicano@gmail.com>
* channels/iax2/include/parser.h, main/acl.c,
include/asterisk/netsock2.h, CHANGES, channels/chan_iax2.c,
channels/iax2/parser.c, main/netsock.c, main/netsock2.c: 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/
2013-10-04 19:31 +0000 [r400552] David M. Lee <dlee@digium.com>
* rest-api/api-docs/applications.json (added): Added missing file
from r400522
2013-10-04 18:42 +0000 [r400532-400542] 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/
* contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py,
configs/extconfig.conf.sample, configs/sorcery.conf.sample,
contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py
(added): 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/
2013-10-04 15:54 +0000 [r400522] Matthew Jordan <mjordan@digium.com>
* res/stasis/app.h, rest-api/resources.json,
include/asterisk/_private.h, main/endpoints.c,
res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
res/res_ari_model.c, main/json.c, res/ari.make,
res/ari/resource_applications.c (added),
res/ari/resource_applications.h (added), res/res_stasis.c,
main/asterisk.c, rest-api/api-docs/endpoints.json,
rest-api/api-docs/events.json, res/stasis/app.c,
include/asterisk/endpoints.h,
rest-api-templates/ari_model_validators.h.mustache,
res/res_ari_applications.c (added), res/ari/resource_endpoints.h,
include/asterisk/stasis_app.h: 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
2013-10-04 15:48 +0000 [r400510-400520] 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.
* res/res_pjsip_session.c: Replace the connection address at the
SDP level if altering the SDP with the external media address.
2013-10-04 04:54 +0000 [r400508] David M. Lee <dlee@digium.com>
* rest-api/api-docs/playback.json, res/res_ari_playback.c:
Corrected response class for stopPlayback
2013-10-03 23:11 +0000 [r400471] 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
2013-10-03 21:40 +0000 [r400460] 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.
2013-10-03 19:31 +0000 [r400442] 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.
2013-10-03 19:29 +0000 [r400440] 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
2013-10-03 19:11 +0000 [r400403] Mark Michelson <mmichelson@digium.com>
* include/asterisk/bridge_technology.h,
bridges/bridge_native_rtp.c: 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 (closes issue
ASTERISK-22532) reported by Matt Jordan Review:
https://reviewboard.asterisk.org/r/2899
2013-10-03 19:05 +0000 [r400401] 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.
2013-10-03 18:44 +0000 [r400398] Richard Mudgett <rmudgett@digium.com>
* main/cel.c: cel: Some whitespace cleanups
2013-10-03 18:28 +0000 [r400384-400395] 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
* 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)
2013-10-03 17:55 +0000 [r400383] Matthew Jordan <mjordan@digium.com>
* 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.
2013-10-03 16:22 +0000 [r400373] Richard Mudgett <rmudgett@digium.com>
* channels/chan_vpb.cc: chan_vpb: Make compile again.
2013-10-03 14:56 +0000 [r400362] Mark Michelson <mmichelson@digium.com>
* tests/test_cel.c: Get rid of uses of stasis_topic_wait()
2013-10-03 14:51 +0000 [r400360] 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
2013-10-02 22:34 +0000 [r400318-400356] Mark Michelson <mmichelson@digium.com>
* main/rtp_engine.c, addons/chan_ooh323.c,
channels/chan_multicast_rtp.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,
main/manager.c, pbx/pbx_spool.c, channels/chan_skinny.c,
main/format_cap.c, channels/chan_motif.c, res/res_agi.c,
channels/chan_alsa.c, apps/app_confbridge.c,
addons/chan_mobile.c, channels/chan_mgcp.c,
res/res_clioriginate.c, channels/chan_sip.c,
channels/chan_bridge_media.c, res/res_pjsip_sdp_rtp.c,
tests/test_format_api.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,
channels/chan_h323.c, main/file.c,
res/res_pjsip/pjsip_configuration.c, tests/test_config.c,
channels/chan_nbs.c, bridges/bridge_native_rtp.c,
res/res_stasis.c, channels/chan_pjsip.c, channels/chan_unistim.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
* /: Remove svn:mergeinfo property.
* main/stasis_endpoints.c, 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: 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
2013-10-02 21:32 +0000 [r400316] 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
2013-10-02 21:25 +0000 [r400312] Matthew Jordan <mjordan@digium.com>
* main/cel.c, main/cdr.c, main/manager.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/
2013-10-02 20:30 +0000 [r400303] 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.
2013-10-02 19:19 +0000 [r400291] Kinsey Moore <kmoore@digium.com>
* rest-api/api-docs/asterisk.json: Correct allowable values for ARI
general information filter
2013-10-02 18:57 +0000 [r400286] 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.
2013-10-02 18:42 +0000 [r400284] 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.
2013-10-02 18:25 +0000 [r400281] 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
2013-10-02 17:11 +0000 [r400268-400270] Richard Mudgett <rmudgett@digium.com>
* main/utils.c, apps/app_stack.c, res/stasis_recording/stored.c,
main/json.c, main/stasis_cache.c, res/res_ari.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.
* channels/sig_ss7.c: sig_ss7: Fix compiler warnings.
2013-10-02 16:20 +0000 [r400245-400265] Joshua Colp <jcolp@digium.com>
* channels/chan_jingle.c, main/channel.c, main/dial.c,
channels/chan_dahdi.c, include/asterisk/stasis_channels.h,
channels/chan_skinny.c, channels/chan_motif.c,
channels/chan_alsa.c, main/stasis_channels.c,
channels/chan_pjsip.c, channels/sig_ss7.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: 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/
* 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
* 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
* res/res_pjsip_session.c: Retrieve and store the hostname only
once so multiple threads do not potentially initialize it at the
same time.
2013-10-01 21:17 +0000 [r400227-400236] 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.
* main/features_config.c: Features: Rearm the parking config
options have moved warning for each reload.
2013-10-01 15:48 +0000 [r400217] 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.
2013-09-30 21:31 +0000 [r400205] Jonathan Rose <jrose@digium.com>
* configs/res_parking.conf.sample, configs/features.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
2013-09-30 19:57 +0000 [r400194-400196] 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
* 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.
2013-09-30 18:48 +0000 [r400178-400181] David M. Lee <dlee@digium.com>
* include/asterisk/stasis.h, main/devicestate.c, res/res_xmpp.c,
main/taskprocessor.c, main/sounds_index.c, main/endpoints.c,
channels/chan_iax2.c, res/res_jabber.c,
res/parking/parking_bridge_features.c, res/res_chan_stats.c,
main/cdr.c, main/manager_bridges.c, main/manager.c,
channels/chan_skinny.c, tests/test_devicestate.c,
res/res_pjsip_mwi.c, tests/test_taskprocessor.c,
tests/test_stasis.c, res/parking/parking_manager.c,
channels/chan_mgcp.c, res/res_security_log.c, main/pbx.c,
main/ccss.c, apps/app_meetme.c, include/asterisk/taskprocessor.h,
res/parking/parking_applications.c, channels/sig_pri.c,
apps/app_queue.c, main/cel.c, main/stasis.c,
channels/chan_dahdi.c, main/stasis_message_router.c,
funcs/func_presencestate.c, apps/confbridge/confbridge_manager.c,
res/res_agi.c, res/res_stasis_test.c, main/manager_channels.c,
main/manager_mwi.c, res/res_pjsip_refer.c, apps/app_voicemail.c,
main/stasis_cache.c, main/stasis_wait.c, res/stasis/app.c,
include/asterisk/stasis_internal.h, channels/chan_sip.c,
main/manager_endpoints.c: 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/
* main/manager_system.c, tests/test_stasis.c,
main/manager_channels.c, main/manager_mwi.c,
main/stasis_cache_pattern.c, include/asterisk/vector.h (added),
res/stasis/app.c, main/channel_internal_api.c,
include/asterisk/stasis.h, apps/app_queue.c, main/cel.c,
main/stasis.c, tests/test_stasis_endpoints.c, main/cdr.c,
main/manager_bridges.c, main/manager.c: 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/
* configure, include/asterisk/autoconfig.h.in,
configs/stasis.conf.sample (removed), include/asterisk/sem.h
(added), configure.ac, include/asterisk/stasis.h,
main/taskprocessor.c, main/sem.c (added), main/stasis.c,
main/stasis_config.c (removed), include/asterisk/taskprocessor.h:
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/
2013-09-30 15:55 +0000 [r400141] Kinsey Moore <kmoore@digium.com>
* configs/pjsip.conf.sample, res/res_pjsip_outbound_registration.c,
configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c:
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
2013-09-30 15:24 +0000 [r400138] David M. Lee <dlee@digium.com>
* main/astobj2.c, main/stasis.c, main/stasis_message_router.c,
main/taskprocessor.c, include/asterisk/stasis_message_router.h,
res/res_pjsip/include/res_pjsip_private.h, tests/test_stasis.c:
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/
2013-09-28 22:56 +0000 [r400058-400121] 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/
* 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
* configure.ac, /, configure: 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
* 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
* 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
* 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.
2013-09-27 21:56 +0000 [r400015-400020] 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
* /, channels/chan_sip.c, channels/sip/reqresp_parser.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
2013-09-27 18:26 +0000 [r399990] Kevin Harwell <kharwell@digium.com>
* include/asterisk/res_pjsip.h, res/res_pjsip.exports.in,
res/res_pjsip.c, res/res_pjsip_session.c: 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/
2013-09-27 17:34 +0000 [r399976] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c, channels/sip/include/sip.h: 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
2013-09-27 17:03 +0000 [r399937] 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/
2013-09-27 14:29 +0000 [r399924] 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.
2013-09-27 14:01 +0000 [r399912] Kinsey Moore <kmoore@digium.com>
* include/asterisk/cel.h, tests/test_cel.c, main/cel.c: 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)
2013-09-26 18:48 +0000 [r399897] Kevin Harwell <kharwell@digium.com>
* res/res_pjsip/security_events.c, res/res_pjsip_registrar.c,
include/asterisk/res_pjsip.h, res/res_pjsip.exports.in: 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/
2013-09-26 15:41 +0000 [r399887] David M. Lee <dlee@digium.com>
* main/channel.c: Minor performance bump by not allocate manager
variable struct if we don't need it
2013-09-26 14:12 +0000 [r399874] 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
2013-09-25 20:36 +0000 [r399842] 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
2013-09-25 19:40 +0000 [r399798] 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>
2013-09-25 19:28 +0000 [r399796] 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
2013-09-25 18:36 +0000 [r399781] 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/
2013-09-24 22:50 +0000 [r399736-399749] 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.
* 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.
2013-09-24 19:22 +0000 [r399666-399695] 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.
* 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.
* main/cdr.c, main/stasis_bridges.c, tests/test_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/
2013-09-23 12:02 +0000 [r399624] 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
2013-09-21 04:48 +0000 [r399576-399607] 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
* main/media_index.c: media_index: Fix process_description_file()
memory leak of file_id_persist.
* main/features_config.c: features_config: Fix config ref leak of
parkinglots. This leak happend for just about every channel
created.
* apps/app_queue.c: app_queue: Fix json blob ref leak. The json ref
from queue_member_blob_create() was never released.
* 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.
2013-09-20 22:41 +0000 [r399565] 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
2013-09-20 22:04 +0000 [r399553] 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/
2013-09-20 16:17 +0000 [r399531] 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)
2013-09-20 14:25 +0000 [r399514] 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
2013-09-19 23:16 +0000 [r399501] Richard Mudgett <rmudgett@digium.com>
* main/optional_api.c: optional_api: Make always use the standard
malloc functions even with MALLOC_DEBUG.
2013-09-19 16:53 +0000 [r399458] 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
2013-09-18 19:59 +0000 [r399404] 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
2013-09-18 17:23 +0000 [r399365-399376] 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
* 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
2013-09-18 14:34 +0000 [r399339] 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/
2013-09-18 00:12 +0000 [r399294] 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
2013-09-17 23:08 +0000 [r399283] Mark Michelson <mmichelson@digium.com>
* include/asterisk/res_pjsip.h, res/res_pjsip_sdp_rtp.c,
res/res_pjsip/pjsip_configuration.c, res/res_pjsip_t38.c: 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
2013-09-17 18:37 +0000 [r399268] Kevin Harwell <kharwell@digium.com>
* main/asterisk.c, /, main/logger.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
2013-09-17 17:54 +0000 [r399257] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/features_config.h: Fix doxygen to use correct
units of features.conf options.
2013-09-17 17:09 +0000 [r399237-399247] Mark Michelson <mmichelson@digium.com>
* main/features_config.c, main/bridge_basic.c: Fix other timeouts
(atxferloopdelay and atxfernoanswertimeout) to use seconds
instead of milliseconds. Thanks to Richard Mudgett for pointing
this out.
* include/asterisk/features_config.h, main/bridge_basic.c,
main/features_config.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.
2013-09-17 14:48 +0000 [r399225] 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
2013-09-16 18:34 +0000 [r399160-399207] 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
* res/res_speech.exports.in, apps/app_speech_utils.c:
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
* /, 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
2013-09-16 02:33 +0000 [r399146] 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.
2013-09-13 22:05 +0000 [r399136] Richard Mudgett <rmudgett@digium.com>
* 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,
res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
include/asterisk/features.h, main/channel.c,
include/asterisk/bridge_channel.h, res/parking/parking_tests.c,
main/features.c, tests/test_cel.c, main/bridge_channel.c,
include/asterisk/bridge.h, apps/confbridge/conf_chan_announce.c,
tests/test_cdr.c, res/res_pjsip_refer.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/
2013-09-13 20:54 +0000 [r399100] 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
2013-09-13 14:49 +0000 [r399083] 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
2013-09-13 14:25 +0000 [r399064] 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
2013-09-13 14:24 +0000 [r399059] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_outbound_authenticator_digest.c,
res/res_pjsip_authenticator_digest.c,
res/res_pjsip/config_auth.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)
2013-09-13 14:21 +0000 [r399039-399049] David M. Lee <dlee@digium.com>
* res/res_pjsip_logger.c, res/res_rtp_asterisk.c,
res/res_pjsip_log_forwarder.c (added): 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/
* 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/
2013-09-13 13:54 +0000 [r399035] 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
2013-09-13 13:27 +0000 [r399031] 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
2013-09-12 23:41 +0000 [r399019-399021] 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.
* main/bridge.c: astobj2: Register the bridges container for debug
inspection.
2013-09-12 23:21 +0000 [r399017] 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/
2013-09-12 20:20 +0000 [r398991] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c, channels/sip/include/sip.h: 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
2013-09-12 16:38 +0000 [r398938] 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/
2013-09-12 15:23 +0000 [r398927] 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_.
2013-09-12 00:04 +0000 [r398882-398886] 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
* 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
2013-09-11 19:56 +0000 [r398837] Jonathan Rose <jrose@digium.com>
* channels/sip/include/sip.h, /, channels/chan_sip.c: 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
2013-09-11 18:02 +0000 [r398821] 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
2013-09-11 14:14 +0000 [r398806] 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/
2013-09-10 18:03 +0000 [r398759] Richard Mudgett <rmudgett@digium.com>
* 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, main/event.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
2013-09-10 17:49 +0000 [r398750-398754] 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
* utils/ael_main.c, utils/conf2ael.c, utils/check_expr.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
2013-09-09 23:23 +0000 [r398726] 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
2013-09-09 21:59 +0000 [r398694] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_endpoint_identifier_ip.c: Add extra debugging to
res_pjsip_endpoint_identifier_ip
2013-09-09 20:12 +0000 [r398638-398651] David M. Lee <dlee@digium.com>
* main/lock.c, /, main/utils.c, include/asterisk/lock.h: 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
* res/ari/resource_channels.h: Added note about expected behavior
of originate (the rest of the commit)
* rest-api/api-docs/channels.json: Added note about expected
behavior of originate
2013-09-08 23:25 +0000 [r398628] 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.
2013-09-07 01:02 +0000 [r398580-398619] 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
* /, 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
2013-09-06 21:16 +0000 [r398579] 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.
2013-09-06 21:03 +0000 [r398578] Kinsey Moore <kmoore@digium.com>
* /: Unblock r398558
2013-09-06 20:20 +0000 [r398533-398572] Richard Mudgett <rmudgett@digium.com>
* 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.
* 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().
* main/parking.c, main/stasis_config.c, include/asterisk/astobj2.h,
main/cel.c, main/features_config.c, apps/app_agent_pool.c,
main/cdr.c, main/udptl.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.
2013-09-06 18:49 +0000 [r398511-398521] Kinsey Moore <kmoore@digium.com>
* res/stasis/app.c, main/http.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)
* 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
2013-09-05 21:46 +0000 [r398381-398498] 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.
* 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
* 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
* 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
2013-09-05 14:09 +0000 [r398368] 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
2013-09-04 23:06 +0000 [r398303] 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
2013-09-04 22:28 +0000 [r398299] 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)
2013-09-04 21:36 +0000 [r398283-398286] 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
* 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
2013-09-04 16:00 +0000 [r398237] 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
2013-09-04 14:29 +0000 [r398226] 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
2013-09-03 19:49 +0000 [r398215] 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
2013-09-03 18:08 +0000 [r398206] 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.
2013-09-03 14:36 +0000 [r398198] David M. Lee <dlee@digium.com>
* Makefile: Fixed 'make clean' for wiki docs
2013-09-03 14:27 +0000 [r398196] 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
2013-08-30 20:58 +0000 [r398149] David M. Lee <dlee@digium.com>
* main/optional_api.c, main/asterisk.c,
include/asterisk/optional_api.h: 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.
2013-08-30 20:34 +0000 [r398147] 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/
2013-08-30 19:51 +0000 [r398116-398139] Kevin Harwell <kharwell@digium.com>
* include/asterisk/sorcery.h, res/res_pjsip.c,
res/res_pjsip/config_transport.c, main/sorcery.c,
res/res_pjsip_outbound_registration.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/
* main/translate.c, main/named_acl.c, main/indications.c,
main/config.c, res/res_security_log.c, /, channels/chan_sip.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
2013-08-30 18:35 +0000 [r398100] Matthew Jordan <mjordan@digium.com>
* 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.
2013-08-30 18:18 +0000 [r398068] Jonathan Rose <jrose@digium.com>
* main/config_options.c, main/features_config.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/
2013-08-30 17:57 +0000 [r398062] 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
2013-08-30 17:10 +0000 [r398023-398025] 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.
* /, 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
2013-08-30 16:57 +0000 [r398020] Jonathan Rose <jrose@digium.com>
* main/udptl.c, main/features_config.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/
2013-08-30 16:26 +0000 [r398002-398016] 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
* main/asterisk.c, /: Check return value on fwrite ........ Merged
revisions 398000 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-30 13:39 +0000 [r397985-397989] David M. Lee <dlee@digium.com>
* build_tools/cflags.xml, configure, res/res_ari_events.c,
include/asterisk/http_websocket.h, main/optional_api.c (added),
rest-api-templates/swagger_model.py, res/ari/ari_websockets.c,
main/asterisk.c, channels/sip/include/sip.h, 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: 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/
* res/stasis_recording (added), res/ari/resource_recordings.c,
res/ari/ari_model_validators.h, res/res_ari_recordings.c,
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: 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/
2013-08-30 01:19 +0000 [r397975-397977] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c: pbx.c: Make pbx_substitute_variables_helper_full()
not mask variables.
* main/pbx.c, tests/test_substitution.c, funcs/func_cdr.c: Revert
last commit.
* funcs/func_cdr.c, main/pbx.c, tests/test_substitution.c: pbx.c:
Make ast_str_substitute_variables_full() not mask variables.
2013-08-30 00:10 +0000 [r397960-397968] 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.
* 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.
2013-08-29 22:45 +0000 [r397958] Kevin Harwell <kharwell@digium.com>
* apps/app_verbose.c, main/asterisk.c, channels/chan_misdn.c, /,
apps/app_dumpchan.c, main/logger.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
2013-08-29 22:24 +0000 [r397955] 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.
2013-08-29 21:34 +0000 [r397946] Kevin Harwell <kharwell@digium.com>
* main/cel.c, main/asterisk.c, main/cdr.c, main/manager.c,
main/stasis_config.c, main/file.c, main/app.c,
main/config_options.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)
2013-08-29 21:33 +0000 [r397945] Mark Michelson <mmichelson@digium.com>
* 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
2013-08-29 20:21 +0000 [r397938] Matthew Jordan <mjordan@digium.com>
* CHANGES, contrib/scripts/safe_asterisk, Makefile,
configs/safe_asterisk.conf.sample (removed): 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.
2013-08-29 16:18 +0000 [r397927] David M. Lee <dlee@digium.com>
* rest-api-templates/asterisk_processor.py,
rest-api-templates/make_ari_stubs.py,
rest-api-templates/api.wiki.mustache: Account for {} in Swagger
notes
2013-08-29 16:04 +0000 [r397924] 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.
2013-08-29 15:42 +0000 [r397921-397922] Mark Michelson <mmichelson@digium.com>
* main/cel.c: Remove extra debug message.
* apps/app_queue.c, main/cel.c, main/stasis_bridges.c: 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.
2013-08-29 12:27 +0000 [r397911] Matthew Jordan <mjordan@digium.com>
* contrib/ast-db-manage/README.md (added),
contrib/ast-db-manage/config/versions (added),
contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py
(added), contrib/ast-db-manage (added),
contrib/ast-db-manage/voicemail/versions (added),
contrib/ast-db-manage/config.ini.sample (added),
contrib/ast-db-manage/config/env.py (added),
contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py
(added), contrib/ast-db-manage/config (added),
contrib/ast-db-manage/config/script.py.mako (added),
contrib/ast-db-manage/voicemail.ini.sample (added),
contrib/ast-db-manage/voicemail/env.py (added),
contrib/ast-db-manage/voicemail (added),
contrib/ast-db-manage/voicemail/script.py.mako (added): Actually
*add* the database schema management utilities In r397874, the
scripts were removed... but not replaced. Thanks to Michael Young
for noticing this!
2013-08-28 23:14 +0000 [r397885-397902] Richard Mudgett <rmudgett@digium.com>
* main/stdtime/localtime.c, main/cdr.c, funcs/func_cdr.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
* 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.
* main/cdr.c: Made the on/off in CLI "cdr set debug [on|off]" case
insensitive.
* main/cdr.c: Make CDR variable name chandling consistently case
insensitive.
* main/cdr.c: Make CDR code deal with channel names case
insensitively.
* funcs/func_cdr.c, main/cdr.c: Some CDR code optimization.
* funcs/func_cdr.c: Whitespace and curly braces.
2013-08-28 21:05 +0000 [r397876] 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.
2013-08-28 20:55 +0000 [r397870-397874] Matthew Jordan <mjordan@digium.com>
* contrib/realtime/mysql/voicemail_messages.sql,
contrib/realtime/postgresql/realtime.sql,
contrib/realtime/mysql/voicemail_data.sql, CHANGES,
contrib/realtime/mysql/musiconhold.sql,
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: 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)
* 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/
2013-08-28 16:12 +0000 [r397856-397859] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c: pbx.c: Make ast_str_substitute_variables_full() not
mask variables.
* include/asterisk/threadstorage.h: Match use of ast_free() with
ast_calloc() and add some curly braces.
2013-08-28 15:40 +0000 [r397854] 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.
2013-08-27 19:15 +0000 [r397816] David M. Lee <dlee@digium.com>
* 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,
rest-api-templates/param_parsing.mustache: 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/
2013-08-27 18:49 +0000 [r397809] 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
2013-08-27 18:05 +0000 [r397759] 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
2013-08-27 16:47 +0000 [r397745] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/chan_sip.c, channels/chan_motif.c, channels/chan_iax2.c,
channels/sig_pri.c, channels/sig_ss7.c: Fix uninitialized value
in struct ast_control_pvt_cause_code usage. ........ Merged
revisions 397744 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-27 16:03 +0000 [r397690-397713] Matthew Jordan <mjordan@digium.com>
* /, 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
* 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.
2013-08-26 21:30 +0000 [r397673] David M. Lee <dlee@digium.com>
* main/bucket.c: Fixed bucket.c for systems where tv_usec is not an
unsigned long.
2013-08-26 16:24 +0000 [r397643-397650] 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.
* channels/Makefile: chan_dahdi: Add some missing build cleanup.
2013-08-25 18:12 +0000 [r397621-397630] 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.
* 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.
* 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
2013-08-23 22:36 +0000 [r397614] Matthew Jordan <mjordan@digium.com>
* / (added): __________ | \ |_______ | | | ______| | / | _ _ _ _ _
| _______| / \ ___| |_ ___ _ __(_)___| | __ / || | / _ \ / __|
__/ _ \ '__| / __| |/ / | || |_______ / ___ \__ \| | __/ | | \__
\ < | || | /_/ \_\___/\__\___|_| |_|___/_|\_\ |_| \__________|
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.
2013-08-23 22:12 +0000 [r397606] Matthew Jordan <mjordan@digium.com>
* main/pbx.c: Fix channel reference leak in Originated channels
When originating channels, ast_pbx_outgoing_* caused the dialed
channel reference to be bumped twice. Ostensibly, this routine is
bumping the channel lifetime such that the channel doesn't get
nuked in between locks/unlocks; however, since the routine should
return the dialed channel with its reference bumped, it only
needs to do this one time.
2013-08-23 21:53 +0000 [r397603] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip.c: Add some clarifying documentation to the
rewrite_contact endpoint option.
2013-08-23 21:51 +0000 [r397602] Richard Mudgett <rmudgett@digium.com>
* main/bridge_channel.c: Blank line tweaks.
2013-08-23 21:49 +0000 [r397599-397600] Joshua Colp <jcolp@digium.com>
* makeopts.in, main/asterisk.c, include/asterisk/bucket.h (added),
main/sorcery.c, include/asterisk/config_options.h,
tests/test_bucket.c (added), build_tools/menuselect-deps.in,
configure, include/asterisk/autoconfig.h.in, main/Makefile,
main/bucket.c (added), configure.ac, main/config_options.c: Add
the bucket API. Bucket is a URI based API for the creation,
retrieval, updating, and deletion of "buckets" and files
contained within them. Review:
https://reviewboard.asterisk.org/r/2715/
* include/asterisk/sorcery.h: Fix a bug where the argc value was
passed as no_doc when registering custom sorcery types. This also
adds a _nodoc equivalent.
2013-08-23 21:02 +0000 [r397593] Mark Michelson <mmichelson@digium.com>
* main/bridge_channel.c: Add test events necessary for bridge tests
to pass in the test suite. (closes issue AST-1200) reported by
John Bigelow Review: https://reviewboard.asterisk.org/r/2790/
2013-08-23 20:14 +0000 [r397585] Matthew Jordan <mjordan@digium.com>
* main/stasis_channels.c: Fix error in using
ast_channel_snapshot_type before initialization Starting Asterisk
would kick back an ERROR message stating that the Stasis message
type ast_channel_snapshot_type was used prior to initialization.
This occurred due to the caching topic being created prior to the
message type that it depended on. This patch re-orders the start
up such that the message type is initialized prior to the caching
topic. It also checks the return value of the initialization of
the agent login/logoff types.
2013-08-23 19:05 +0000 [r397578] Jonathan Rose <jrose@digium.com>
* bridges/bridge_native_rtp.c: bridge_native_rtp: Fix hold chain
bugs caused by native RTP bridge framehook Issuing hold/unhold
would lead to odd behavior. Between two chan_sip devices, a hold
could cause an endless chain of updates while with pjsip a
similar chain would begin but then end somewhat randomly. This
patch fixes that by no longer tweaking the RTP glue on both sides
of the call for every HOLD/UNHOLD/UPDATE_RTP_PEER frame. (issue
ASTERISK-22217) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2794/
2013-08-23 18:33 +0000 [r397577] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridge_channel.h, main/channel_internal_api.c,
bridges/bridge_builtin_interval_features.c,
include/asterisk/channel.h, res/res_musiconhold.c,
main/bridge_channel.c, main/channel.c,
include/asterisk/bridge_channel_internal.h, main/bridge.c: Handle
DTMF and hold wrapup when a channel leaves the bridging system.
DTMF start/end and hold/unhold events have state because a DTMF
begin event and hold event must be ended by something. The
following cases need to be handled when a channel is moved around
in the system. * When a channel leaves a bridge it may owe a DTMF
end event to the bridge. * When a channel leaves a bridge it may
owe an UNHOLD event to the bridge. (This case is explicitly
ignored because things like transfers need explicit control over
this.) * When a channel leaves the bridging system it may need to
simulate a DTMF end event to the channel. * When a channel leaves
the bridging system it may need to simulate an UNHOLD event to
the channel. The patch also fixes the following: * Fixes playing
a file and restarting MOH using the latest MOH class used.
(closes issue ASTERISK-22043) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2791/
2013-08-23 18:10 +0000 [r397571] Matthew Jordan <mjordan@digium.com>
* tests/test_sorcery_realtime.c, tests/test_sorcery_astdb.c,
tests/test_sorcery.c: Fix sorcery unit tests When strict XML
documentation checking was re-enabled, the test objects used in
sorcery would fail to register as the types were not marked
internal and the nodoc option wasn't used for the options. This
fixes that problem, such that, as one would hope, they once again
pass.
2013-08-23 18:07 +0000 [r397570] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/utils.h, include/asterisk/astmm.h, /,
main/backtrace.c, main/logger.c, main/utils.c,
include/asterisk/lock.h, main/astmm.c, channels/sig_pri.c,
main/astobj2.c, include/asterisk/backtrace.h, main/lock.c: Fix
memory corruption when trying to get "core show locks". Review
https://reviewboard.asterisk.org/r/2580/ tried to fix the
mismatch in memory pools but had a math error determining the
buffer size and didn't address other similar memory pool
mismatches. * Effectively reverted the previous patch to go in
the same direction as trunk for the returned memory pool of
ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols()
when BETTER_BACKTRACES is defined. * Fixed some formatting in
ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated
by libpri when MALLOC_DEBUG is enabled. * Fixed
__dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of
compile issues with the utils directory. (closes issue
ASTERISK-22221) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2778/ ........ Merged
revisions 397525 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397528 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-23 18:02 +0000 [r397568] Matthew Jordan <mjordan@digium.com>
* main/config_options.c: Prevent seg fault in off nominal path when
registered option fails to validate If an option is registered to
a type and it is the last known type in the list of registered
types, and the option fails to register, an overrun of the types
array can occur due to the index variable having been already
incremented.
2013-08-23 17:45 +0000 [r397567] Kevin Harwell <kharwell@digium.com>
* contrib/scripts/sip_to_res_sip/sip_to_res_sip.py,
contrib/scripts/sip_to_res_sip/astconfigparser.py,
contrib/scripts/sip_to_res_sip/astdicts.py: PSJIP - sip.conf to
res_sip.conf script Most, if not all, of the backing features of
a conf file should now be implemented (e.g. multi-line comments,
includes, templates, etc...). A few of the options still need to
be mapped. Those are currently listed in the 'sip_to_res_sip.py'
file. Things to do: (1) There is more work to do here, at least
for the sip.conf items that aren't currently parsed. An issue
will be created for that. (2) All of the scripts should probably
be passed through pylint and have as many PEP8 issues fixed as
possible. (3) A public review is probably warranted at that point
of the entire script. Reported by: Matt Jordan
2013-08-23 17:19 +0000 [r397565] David M. Lee <dlee@digium.com>
* rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
res/res_ari_bridges.c, res/stasis/control.c,
include/asterisk/stasis_app.h,
include/asterisk/stasis_app_impl.h: ARI: Correct error codes for
bridge operations This patch adds error checking to ARI bridge
operations, when adding/removing channels to/from bridges. In
general, the error codes fall out as follows: * Bridge not found
- 404 Not Found * Bridge not in Stasis - 409 Conflict * Channel
not found - 400 Bad Request * Channel not in Stasis - 422
Unprocessable Entity * Channel not in this bridge (on remove) -
422 Unprocessable Entity (closes issue ASTERISK-22036) Review:
https://reviewboard.asterisk.org/r/2769/
2013-08-23 15:49 +0000 [r397524-397527] Matthew Jordan <mjordan@digium.com>
* CHANGES: Update CHANGES file to reflect pass through support for
Opus/VP8
* channels/chan_sip.c, res/res_pjsip_sdp_rtp.c,
include/asterisk/opus.h (added), include/asterisk/format.h,
channels/chan_pjsip.c, res/res_format_attr_opus.c (added),
main/channel.c, main/format.c, res/res_rtp_asterisk.c,
main/frame.c, main/rtp_engine.c: Add pass through support for
Opus and VP8; Opus format attribute negotiation This patch adds
pass through support for Opus and VP8. That includes: * Format
attribute negotiation for Opus. Note that unlike some other
codecs, the draft RFC specifies having spaces delimiting the
attributes in addition to ';', so you have "attra=X; attrb=Y".
This broke the attribute parsing in chan_sip, so a small tweak
was also included in this patch for that. * A format attribute
negotiation module for Opus, res_format_attr_opus * Fast picture
update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time. Note that
the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by
Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero
(License 6518)
* main/sorcery.c, include/asterisk/config_options.h,
include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c,
main/config_options.c, main/features_config.c,
res/res_pjsip/pjsip_options.c, res/res_pjsip.c: Update config
framework/sorcery with types/options without documentation There
are times when a configuration option should not have
documentation. 1. Some options are registered with a particular
object merely as a warning to users. These options aren't even
really 'deprecated' - which has its own separate API call - they
are actually provided by a different configuration file. The
options are merely registered so that the user gets a warning
that a different configuration file provides the item. 2. Some
object types - most notably some used by modules that use sorcery
- are completely internal and should never be shown to the user.
3. Sorcery itself has several 'hidden' fields that should never
be shown to a user. This patch updates the configuration
framework and sorcery with additional API calls that allow a
module to register types as internal and options as not requiring
documentation. This bypasses the XML documentation checking. This
patch also re-enables the strict XML documentation checking in
trunk, as well as updates some documentation that was missing.
Review: https://reviewboard.asterisk.org/r/2785/ (closes issue
ASTERISK-22359) Reported by: Matt Jordan (closes issue
ASTERISK-22112) Reported by: Rusty Newton
2013-08-23 13:58 +0000 [r397515] Joshua Colp <jcolp@digium.com>
* channels/chan_pjsip.c: Fix crash when answering after a transport
error occurs. If a response to an initial incoming INVITE results
in a transport error the INVITE transaction is removed from the
INVITE session. Any attempts to answer the INVITE session after
this results in a crash as it requires the INVITE transaction to
exist. This change explicitly locks the dialog and checks to
ensure that the INVITE transaction exists before answering.
(closes issue AST-1203) Reported by: John Bigelow
2013-08-23 13:18 +0000 [r397514] Kinsey Moore <kmoore@digium.com>
* configs/cel.conf.sample: Update CEL sample config
2013-08-23 00:26 +0000 [r397505] Jonathan Rose <jrose@digium.com>
* res/res_stasis.c, rest-api/api-docs/bridges.json,
res/ari/resource_bridges.c, res/res_ari_bridges.c,
res/ari/resource_bridges.h, include/asterisk/stasis_app.h: ARI:
Music on Hold/Background Music for bridges Adds ARI functions to
be able to turn on/off music on hold in a bridge. It actually
functions more as a background music without further actions on
the bridge since if the rest of the channels in the bridge aren't
explicitly muted, they will still be able to communicate. (closes
issue ASTERISK-21974) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2688/
2013-08-22 23:15 +0000 [r397494] Richard Mudgett <rmudgett@digium.com>
* apps/app_followme.c, main/channel.c, bridges/bridge_holding.c:
Minor tweaks with ast_moh_start() callers.
2013-08-22 22:33 +0000 [r397493] Kinsey Moore <kmoore@digium.com>
* include/asterisk/say.h, apps/app_voicemail.c, main/channel.c,
main/pbx.c, main/say.c, res/res_agi.c, CHANGES,
apps/app_directory.c, apps/app_chanspy.c: Add SayAlphaCase and
similar functionality for AGI This adds a new dialplan
application, SayAlphaCase, that performs much the same function
as SayAlpha except that it takes additional options which allow
the user to specify whether the case of each letter should be
announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter. Original Patch by: Kevin Scott Adams Reported
by: Kevin Scott Adams Review:
https://reviewboard.asterisk.org/r/2725/ (closes issue
ASTERISK-20782)
2013-08-22 22:09 +0000 [r397484] Kevin Harwell <kharwell@digium.com>
* res/res_pjsip.c, res/res_pjsip_dtmf_info.c: res_sip_dtmf_info:
Support sending of 'raw' DTMF Added the ability to handle 'raw'
DTMF within the body of an INFO message. Also made it so values
10-16 are mapped to valid DTMF values. (closes issue
ASTERISK-22144) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2776/
2013-08-22 21:39 +0000 [r397483] Kinsey Moore <kmoore@digium.com>
* res/res_pjsip.c: Add missing configOption close tags
2013-08-22 21:29 +0000 [r397482] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/musiconhold.h: Update MOH start/stop routine
doxygen.
2013-08-22 21:21 +0000 [r397481] Rusty Newton <rnewton@digium.com>
* res/res_pjsip.c: Fix missing xml doc configOption 'type' for for
both 'system' and 'global' configObjects (issue ASTERISK-22344)
(closes issue ASTERISK-22344)
2013-08-22 21:09 +0000 [r397472] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridge_channel.h, main/features.c,
bridges/bridge_builtin_interval_features.c,
include/asterisk/bridge_internal.h, apps/app_confbridge.c,
main/bridge_channel.c, res/res_stasis.c,
include/asterisk/bridge.h, apps/app_dial.c, main/bridge.c,
main/bridge_basic.c, apps/app_bridgewait.c,
res/parking/parking_applications.c,
res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
res/res_parking.c, bridges/bridge_builtin_features.c: Bridge API:
Set a cause code on a channel when it is ejected from a bridge.
The cause code needs to be passed from the disconnecting channel
to the bridge peers if the disconnecting channel dissolves the
bridge. * Made the call to an app_agent_pool agent disconnect
with the busy cause code if the agent does not ack the call in
time or hangs up before acking the call. (closes issue
ASTERISK-22042) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2772/
2013-08-22 20:29 +0000 [r397471] Kinsey Moore <kmoore@digium.com>
* main/cel.c: Ensure CEL creates a default config if it isn't
provided with one
2013-08-22 20:18 +0000 [r397466] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Remove set but unused variable 'meid'.
2013-08-22 19:52 +0000 [r397461] Kinsey Moore <kmoore@digium.com>
* main/cel.c: Fix crash when getting CEL config
2013-08-22 18:52 +0000 [r397441-397451] Mark Michelson <mmichelson@digium.com>
* include/asterisk/core_unreal.h, include/asterisk/features.h,
include/asterisk/app.h, main/bridge.c, main/bridge_basic.c,
main/features.c, main/app.c, main/core_local.c, CHANGES,
apps/app_queue.c, include/asterisk/bridge_basic.h: Massively
clean up app_queue. This essentially makes app_queue usable
again. From reviewboard: * Reporting of transfers and call
completion is done by creating stasis subscriptions and listening
for specific events in order to determine when the call is
finished (either via a transfer or hangup). * Dial end messages
have been added where they were previously missing. * Queue stats
are properly being updated again once calls have finished. *
AgentComplete stasis messages and AMI events are now occurring
again. * Mixmonitor starting has been factored into its own
function and uses the Mixmonitor API now instead of using
ast_pbx_run() In addition to the changes in app_queue, there are
several supplementary changes as well: * Queue logging now
differentiates between attended and blind transfers. A note about
this is in the CHANGES file. * Local channel optimization events
now report more information. This includes which of the two local
channels involved is the destination of the optimization, the
channel that is replacing the destination local channel, and an
identifier so that begin and end events can be matched to each
other. The end events are now sent whether the optimization was
successful or not and includes an indicator of whether the
optimization was successful. * Changes were made to features and
bridging_basic so that additional flags may be set on a bridge.
This is necessary because the queue requires that its bridge only
allows move-swap local channel optimizations into the bridge.
(closes issue ASTERISK-21517) Reported by Matt Jordan (closes
issue ASTERISK-21943) Reported by Matt Jordan Review:
https://reviewboard.asterisk.org/r/2694
* res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
res/res_pjsip_mwi.c, res/res_pjsip_pubsub.c: Handle default body
types for SIP event packages in res_pjsip_pubsub Prior to this
change, we would reject SUBSCRIBE requests that had no Accept
headers. Now event package handlers that handle the default type
for the event package indicate that they do so. Therefore, if we
have a handler that can handle the default type, we can allow
SUBSCRIBEs for the handler's event package that have no Accept
headers. (closes issue ASTERISK-22067) reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/2774
2013-08-22 17:34 +0000 [r397440] Richard Mudgett <rmudgett@digium.com>
* main/bridge_channel.c, main/abstract_jb.c: Made the abstract
jitter buffer resync on some more control frames. Resync the
abstract jitter buffer on the following additional control
frames: AST_CONTROL_HOLD AST_CONTROL_UNHOLD
AST_CONTROL_T38_PARAMETERS
2013-08-22 17:13 +0000 [r397431] Kinsey Moore <kmoore@digium.com>
* tests/test_cel.c, main/cel.c, include/asterisk/cel.h: Make CEL
behavior conform to the documentation This modifies the behavior
of the CEL engine to conform to documented behavior for Asterisk
12 as defined on the wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification
The primary changes deal with removal of the peer field from
function calls since it is no longer directly relevant to the
bridging system and removal of the layer of CDR-like business
logic that was providing a partial emulation of Asterisk 11 CEL
functionality. With this change, there is no longer a distinction
between "bridges" and "conferences" and all participation changes
are denoted with bridge enter and bridge exit messages. This
updates the CEL unit tests to handle these changes and simplifies
some of the macros used in the process. This also fixes a
segfault when attempting to ref a configuration that failed to
load. Review: https://reviewboard.asterisk.org/r/2788/ (issue
ASTERISK-21567)
2013-08-22 16:46 +0000 [r397426] Richard Mudgett <rmudgett@digium.com>
* main/bridge.c: Update BUGBUG comment.
2013-08-22 12:28 +0000 [r397379-397415] Walter Doekes <walter+asterisk@wjd.nu>
* main/asterisk.c: Don't store repeated commands in the editline
history buffer. The equivalent of bash HISTCONTROL=ignoredups.
Review: https://reviewboard.asterisk.org/r/2775/
* /, main/asterisk.exports.in, default.exports: Add _IO_stdin_used
in version-script to fix SIGBUSes on Sparc. The
--version-script,asterisk.exports linker flag (and the module
exports) didn't provide _IO_stdin_used in the list of exported
symbols. That causes some kind of libc compatibility mode to kick
in, where stdio file structures (stdout/stderr) land somewhere
else. In the case of the Sparc, they landed on misaligned memory.
This became apparent first after r376428 (Reorder startup
sequence) when a lot of ast_log's were replaced with fprintf's.
Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64
architectures, the Sparc is very picky about memory alignment.)
(issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy
Kister Review: https://reviewboard.asterisk.org/r/2760/ ........
Merged revisions 397377 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397378 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-21 23:09 +0000 [r397366] Jonathan Rose <jrose@digium.com>
* main/udptl.c, /: UDPTL: Fix a regression where UDPTL won't load
default settings If the file udptl.conf is unavailable at
startup, UDPTL will fail to initialize and while it makes some
noise, it isn't immediately obvious why consumers start to fail
when using it. This patch makes UDPTL load as though an empty
config was provided when udptl is unavailable at startup. (closes
issue ASTERISK-22349) Reported by: Jonathan Rose Review:
https://reviewboard.asterisk.org/r/2773/ ........ Merged
revisions 397365 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-21 20:02 +0000 [r397346-397355] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridge_basic.h, main/bridge_basic.c,
main/features.c: * Move ast_bridge_channel_setup_features() into
bridge_basic.c. * Made application map hooks be removed on a
basic bridge personality change.
* main/bridge.c, main/bridge_channel.c: Deferred some more BUGBUG
comments to a JIRA issue or XXX comment.
2013-08-21 17:12 +0000 [r397310] David M. Lee <dlee@digium.com>
* /, main/http.c: Complete http_shutdown. This patch frees up some
resources allocated in http.c. * tcp listeners stopped * tls
settings freed * uri redirects freed * unregister internal http.c
uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell
Patches: http.patch uploaded by Corey Farrell (license 5909)
........ Merged revisions 397308 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397309 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-21 16:31 +0000 [r397307] Matthew Jordan <mjordan@digium.com>
* include/asterisk/frame.h, /: Set 14400 as the default max bit
rate if T38MaxBitRate is not specified If an endpoint fails to
include the T38MaxBitRate attribute during negotiation, Asterisk
will negotiate a bit rate of 2400 instead of the ITU recommended
bit rate of 14400. This patch fixes this by making
AST_T38_RATE_14400 the 'default' value of the enum by assigning
it a value of 0, such that if an endpoint fails to include the
attribute, the default will be 14400. Note that Walter Doekes
included the nice comment in frame.h about why we are
purposefully assigning AST_T38_RATE_14400 a value of 0. (closes
issue ASTERISK-22275) Reported by: Andreas Steinmetz patches:
fax-fix.patch uploaded by anstein (License 6523) ........ Merged
revisions 397256 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397257 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-21 16:23 +0000 [r397295-397306] David M. Lee <dlee@digium.com>
* rest-api/api-docs/asterisk.json, res/ari/resource_asterisk.c,
res/res_ari_asterisk.c, rest-api/api-docs/channels.json,
res/ari/resource_channels.c, res/res_ari_channels.c: ARI: Correct
segfault with /variable calls are missing ?variable parameter.
Both /asterisk/variable and /channel/{channelId}/variable
requires a ?variable parameter to be passed into the query. But
we weren't checking for the parameter being missing, which caused
a segfault. All calls now properly return 400 Bad Request errors
when the parameter is missing. The Swagger api-docs were updated
accordingly. (closes issue ASTERISK-22273)
* main/stasis_endpoints.c: ARI: Remove the 'channel:' scheme from
endpoint's channel list. For times when a reference in ARI might
be ambiguous, the reference is built as an URI (such as
channel:1376341790.3). An endpoint's channel list is not
ambiguous, and in fact the field is named 'channel_ids', but it
had channel URI's instead of channel id's. This patch changes the
list to be the raw id instead of the URI. (closes issue
ASTERISK-22291)
* res/stasis/control.h, res/res_stasis.c: res_stasis: remove call
to missing function control_continue. In the shuffling around of
res_stasis, control_continue was renamed to
stasis_app_control_continue, but the call in res_stasis wasn't
updated. In looking into it, it turns out it wasn't really the
right thing to do in res_stasis anyways. This patch changes the
handling of received a AST_CONTROL_HANGUP frame to be the same as
receiving a NULL frame, and removed the declaration of
control_continue(), since it doesn't exist any more. (closes
issue ASTERISK-22292) Reported by: Denis Smirnov
2013-08-21 15:51 +0000 [r397294] Richard Mudgett <rmudgett@digium.com>
* apps/app_bridgewait.c, include/asterisk/bridge_features.h,
main/bridge_channel.c, res/parking/parking_bridge_features.c,
apps/app_agent_pool.c, bridges/bridge_holding.c, main/bridge.c,
include/asterisk/bridge_channel.h, main/features.c,
bridges/bridge_builtin_interval_features.c: Fix several
interrelated issues dealing with the holding bridge technology. *
Added an option flags parameter to interval hooks. Interval hooks
now can specify if the callback will affect the media path or
not. * Added an option flags parameter to the bridge action
custom callback. The action callback now can specify if the
callback will affect the media path or not. * Made the holding
bridge technology reexamine the participant idle mode option
whenever the entertainment is restarted. * Fixed app_agent_pool
waiting agents needlessly starting and stopping MOH every second
by specifying the heartbeat interval hook as not affecting the
media path. * Fixed app_agent_pool agent alert from restarting
the MOH after the alert beep. The agent entertainment is now
changed from MOH to silence after the alert beep. * Fixed holding
bridge technology to defer starting the entertainment. It was
previously a mixture of immediate and deferred. * Fixed holding
bridge technology to immediately stop the entertainment. It was
previously a mixture of immediate and deferred. If the channel
left the bridging system, any deferred stopping was discarded
before taking effect. * Miscellaneous holding bridge technology
rework coding improvements. Review:
https://reviewboard.asterisk.org/r/2761/
2013-08-21 14:39 +0000 [r397255] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Prevent a crash on outbound SIP MESSAGE
requests. If a From header on an outbound out-of-call SIP MESSAGE
were malformed, the result could crash Asterisk. In addition, if
a From header on an incoming out-of-call SIP MESSAGE request were
malformed, the message was happily accepted rather than being
rejected up front. The incoming message path would not result in
a crash, but the behavior was bad nonetheless. (closes issue
ASTERISK-22185) reported by Zhang Lei ........ Merged revisions
397254 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-21 14:08 +0000 [r397244] Kinsey Moore <kmoore@digium.com>
* res/res_stasis.c: Allow channels in app_stasis to hangup properly
This detects hangups that occur while bridged to allow channels
to exit app_stasis even if the hangup frame was absorbed by the
bridge the channel was in. Reported by: David Lee (closes issue
ASTERISK-22297)
2013-08-21 13:41 +0000 [r397243] Matthew Jordan <mjordan@digium.com>
* CHANGES, channels/chan_sip.c: Allow the SIP_CODEC family of
variables to specify more than one codec The SIP_CODEC family of
variables let you set the preferred codec to be offered on an
outbound INVITE request. However, for video calls, you need to be
able to set both the audio and video codecs to be offered. This
patch lets the SIP_CODEC variables accept a comma delineated list
of codecs. The first codec in the list is set as the preferred
codec; additional codecs are still offered however. This lets a
dialplan writer set both audio and video codecs, e.g.,
Set(SIP_CODEC=ulaw,h264) Note that this feature was written by
both Dennis Guse and Frank Haase Review:
https://reviewboard.asterisk.org/r/2728 (closes issue
ASTERISK-21976) Reported by: Denis Guse Tested by: mjordan,
sysreq patches: patch-channels-chan__sip.c-393919 uploaded by
dennis.guse (license 6513)
2013-08-21 02:15 +0000 [r397206] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix Not Storing Current Incoming Recv
Address In 1.8, r384779 introduced a regression by retrieving an
old dialog and keeping the old recv address since recv was
already set. This has caused a problem when a proxy is involved
since responses to incoming requests from the proxy server, after
an outbound call is established, are never sent to the correct
recv address. In 11, r382322 introduced this regression. The fix
is to revert that change and always store the recv address on
incoming requests. Thank you Walter Doekes for helping to point
out this error and Mark Michelson for your input/review of the
fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin
Tested by: Alex Zarubin, Karsten Wemheuer Patches:
asterisk-22071-store-recvd-address.diff by Michael L. Young
(license 5026) ........ Merged revisions 397204 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397205 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-20 21:01 +0000 [r397111-397193] Mark Michelson <mmichelson@digium.com>
* include/asterisk/res_pjsip.h, res/res_pjsip/config_security.c
(removed), res/res_pjsip/pjsip_configuration.c,
res/res_pjsip_acl.c: Localize and rename ACL configuration. This
is more-or-less a reversion of previous ACL behavior so that it
is more self-contained. ACL sections are now only parsed if
res_pjsip_acl.so is loaded. Moreover, the configuration section
is now "type=acl" instead of "type=security". The original reason
for having ACLs configured in a "type=security" section was to
lump ACLs and other security-related items into the same section.
The problem is that ACLs really should be in their own sections
and there are no other security-related options implemented
anyways.
* /, channels/chan_sip.c: Remove REF_DEBUG definition. ........
Merged revisions 397156 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397157 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c, channels/sip/dialplan_functions.c: Fix
refcounting of sip_pvt in test_sip_rtpqos test and unlink it from
the list of pvts. (closes issue ASTERISK-22248) reported by Corey
Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell
(license #5909) ........ Merged revisions 397112 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397133 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_pjsip.c: Clarify documentation for the "identify_by"
option for SIP endpoints. This also removes documentation for the
options that no longer exist. (closes issue ASTERISK-22306)
reported by Rusty Newton
2013-08-20 15:36 +0000 [r397110] Kinsey Moore <kmoore@digium.com>
* /, main/threadstorage.c, main/astfd.c: Unregister CLI commands on
exit This patch ensures that CLI commands enabled by
DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on
exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell
Tested by: Corey Farrell Patches: debug_cli_unregister.patch
uploaded by Corey Farrell ........ Merged revisions 397106 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 397107 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-20 15:32 +0000 [r397073-397109] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_endpoint_identifier_ip.c: Add debug message to
res_pjsip_endpoint_identifier_ip to indicate when an endpoint is
successfully retrieved. (closes issue ASTERISK-22101) reported by
Rusty Newton
* res/res_pjsip_registrar.c: Add warning messages for registration
failure paths. (closes issue ASTERISK-22089) reported by Rusty
Newton patches: patch1.txt uploaded by John Bigelow (License
#5091)
* res/res_pjsip.c: Add note to transport configuration that a
restart is required to change transports. (closes issue
ASTERISK-22094) reported by Rusty Newton
2013-08-20 14:26 +0000 [r397072] Kinsey Moore <kmoore@digium.com>
* /: Recorded merge of revisions 397067 from
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Fix
xmldoc memory leak This fixes a single-attribute memory leak that
was occurring when the "required" attribute was not true. (closes
issue ASTERISK-22249) Reported by: Corey Farrell Tested by: Corey
Farrell Patches: xmldoc-free_attr_required.patch uploaded by
Corey Farrell ........ Merged revisions 397064 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2013-08-20 11:48 +0000 [r396996] Walter Doekes <walter+asterisk@wjd.nu>
* configs/sip.conf.sample, configs/h323.conf.sample, /: Add
"autoframing" option to sip.conf.sample and h323.conf.sample. The
autoframing option was added to chan_sip.c in r43243 (mogorman,
2006-09-19 01:32:57), but never made its way into the sample
configs. Review: https://reviewboard.asterisk.org/r/2768/
........ Merged revisions 396994 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 396995 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-20 11:33 +0000 [r396993] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_dtmf_info.c: Remove assumption in
res_pjsip_dtmf_info that all INFO messages will contain a body.
(closes issue ASTERISK-22320) Reported by: Matt Jordan
2013-08-20 00:08 +0000 [r396946-396949] Matthew Jordan <mjordan@digium.com>
* /, apps/app_queue.c: Let Queue wrap up time influence member
availability Queue members who happen to be in multiple queues at
the same time may not have any wrap up time. This problem
occurred due to a code change in Asterisk 11.3.0 that unified
device state tracking of Queue members in multiple Queues (which
fixed some other problems, but unfortunately caused this one).
This patch fixes the behavior by having the is_member_available
function check the queue's wrap up time and the time of the
member's last call, such that for a particular queue, the member
won't be considered available if their last call is within the
wrap up time. (closes issue ASTERISK-22189) Reported by: Tony
Lewis Tested by: Tony Lewis ........ Merged revisions 396948 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_meetme.c: Resolve conflicts between
CONFFLAG_DONT_DENOISE and CONFFLAG_INTROUSER_VMREC When r382230
added an option to not denoise the MeetMe conference (if a user
had a channel whose format's sample rate changed frequently, for
example), the value added was the maximum allowed value for the
constants that define the options for MeetMe in 1.8. Not so in 11
- unfortunately, the option CONFFLAG_DONT_DENOISE conflicts with
CONFFLAG_INTROUESR_VMREC. This patch fixes that, and also tweaks
one of the way in which the constants was declared for
consistency. Thanks to Tony Mountifield for pointing out the
problem and solution. (closes issue ASTERISK-22269) Reported by:
Tony Mountifield ........ Merged revisions 396944 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-19 16:10 +0000 [r396930] Richard Mudgett <rmudgett@digium.com>
* main/bridge.c: Update BUGBUG comment.
2013-08-19 14:54 +0000 [r396923] Jonathan Rose <jrose@digium.com>
* main/bridge.c: attended transfers: Fix a bug affecting external
blond transfers Performing a blond transfer (attended transfer
that is completed before the transfer recipient picks up)
externally through chan_sip or chan_pjsip would result in lost
references to the channels involved with the transfer as well as
their bridge. (closes issue ASTERISK-22092) Reported by:
mmichelson Review: https://reviewboard.asterisk.org/r/2766/
2013-08-19 14:53 +0000 [r396915-396922] Matthew Jordan <mjordan@digium.com>
* channels/sip/include/sip.h: Whitespace cleanup Remove some
extraneous blobs
* main/data.c: Fix invalid access to disposed memory in main/data
unit test It is not safe to iterate over a macro'd list of ao2
objects, deref them such that the item's destructor is called,
and leave them in the list. The list macro to iterate over items
requires the item to be a valid allocated object in order to
proceed to the next item; with MALLOC_DEBUG on the corruption of
the linked list is caught in the crash. This patch fixes the
invalid access to free'd memory by removing the ao2 item from the
list before de-refing it.
2013-08-18 03:05 +0000 [r396908-396909] Kinsey Moore <kmoore@digium.com>
* channels/chan_mgcp.c: Update chan_mgcp to the modified parking
API
* res/res_corosync.c: Disable build of res_corosync until it is
back in a compiling state
2013-08-17 18:13 +0000 [r396899-396902] Rusty Newton <rnewton@digium.com>
* res/res_pjsip.c: xml doc changes for 'aor' config object and a
few of its options Added or modified text in the xml doc for the
'aor' config object to address a few issues: * help for the
'mailboxes' option didn't make it clear how the "list" should be
formatted. * AoR object's involvement in inbound registration
wasn't mentioned. * help for the 'contact' option didn't describe
how to specify multiple contacts. * help for the 'max_contacts'
option didn't tell whether it limited the amount of contacts
defined through static configuration. (issue ASTERISK-22118)
(closes issue ASTERISK-22118)
* res/res_pjsip.c: 'domain_alias' config object XML help doesn't
make it clear that the name used for the object is the domain
alias (issue ASTERISK-22114) (closes issue ASTERISK-22114)
* res/res_pjsip.c: xml doc changes for clarity - 'auth' config
object and auth's 'auth_type' config option (issue
ASTERISK-22108) (closes issue ASTERISK-22108)
* res/res_pjsip.c: xml doc change for transport config object -
remove non-applicable warning and add text regarding Asterisk
restart (closes issue ASTERISK-22105)
2013-08-17 15:01 +0000 [r396887-396890] Kinsey Moore <kmoore@digium.com>
* main/bridge.c, res/parking/parking_applications.c,
include/asterisk/parking.h, main/bridge_channel.c,
res/parking/parking_bridge_features.c, channels/chan_dahdi.c,
res/parking/res_parking.h, res/res_parking.c,
channels/sig_analog.c, channels/chan_skinny.c, main/parking.c:
Allow res_parking to be unloadable This change protects accesses
of res_parking such that it can unload safely once transient uses
of its registered functions are complete. The parking API has
been restructured such that its consumers do not have access to
the vtable exposed by the parking provider, but instead route
through stubs to prevent consumers from holding on to function
pointers. This adds calls to all the parking unload functions and
moves application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of
res_parking. Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)
* tests/test_event.c, include/asterisk/_private.h, main/cel.c,
cel/cel_odbc.c, include/asterisk/event.h,
include/asterisk/event_defs.h, cel/cel_manager.c,
cel/cel_custom.c, tests/test_cel.c, cel/cel_sqlite3_custom.c,
main/event.c, main/asterisk.c, cel/cel_pgsql.c, cel/cel_radius.c,
include/asterisk/cel.h, cel/cel_tds.c: Refactor CEL to avoid
using the event system core This removes usage of the event
system for CEL backend data distribution and strips unused pieces
out of the event system. Review:
https://reviewboard.asterisk.org/r/2732/
* main/presencestate.c, channels/sig_pri.h, res/res_parking.c,
channels/chan_dahdi.c, main/manager.c,
funcs/func_presencestate.c, include/asterisk/event.h,
include/asterisk/event_defs.h, channels/chan_skinny.c,
tests/test_cel.c, main/event.c,
include/asterisk/security_events_defs.h,
res/parking/parking_manager.c, channels/chan_mgcp.c,
res/res_security_log.c, apps/app_voicemail.c,
res/parking/parking_ui.c, channels/chan_unistim.c, main/pbx.c,
include/asterisk/devicestate.h, main/security_events.c,
channels/chan_sip.c, main/ccss.c, tests/test_event.c,
main/devicestate.c, res/parking/parking_applications.c,
res/res_xmpp.c, channels/sig_pri.c, channels/chan_iax2.c,
apps/app_queue.c, res/res_jabber.c: Strip down the old event
system This removes unused code, event types, IE pltypes, and
event IE types where possible and makes several functions private
that were once public. This includes a renumbering of the
remaining event and IE types which breaks binary compatibility
with previous versions. The last remaining consumers of the old
event system (or parts thereof) are main/security_events.c,
res/res_security_log.c, tests/test_cel.c, tests/test_event.c,
main/cel.c, and the CEL backends. Review:
https://reviewboard.asterisk.org/r/2703/ (closes issue
ASTERISK-22139)
2013-08-16 20:48 +0000 [r396849-396877] Richard Mudgett <rmudgett@digium.com>
* main/bridge_channel.c, include/asterisk/bridge.h, main/bridge.c,
include/asterisk/bridge_channel.h: Fix CLI "bridge kick <bridge>
<channel>" to check if the bridge needs dissolving. SIP/foo --
Local;1==Local;2 -- .... -- Local;1==Local;2 -- SIP/bar Kick a ;1
channel and the chain toward SIP/foo goes away. Kick a ;2 channel
and the chain toward SIP/bar goes away. This can leave a local
channel chain between the kicked ;1 and ;2 channels that are
orphaned until you manually request one of those channels to
hangup or request the bridge to dissolve. * Added
ast_bridge_kick() as a companion to ast_bridge_remove(). The
functional difference is that ast_bridge_kick() may dissolve the
bridge as a result of the channel leaving the bridge. * Made CLI
"bridge kick <bridge> <channel>" use ast_bridge_kick() instead of
ast_bridge_remove() so the bridge can dissolve if needed. *
Renamed bridge_channel_handle_hangup() to
ast_bridge_channel_kick() and made it accessible to other files.
* include/asterisk/doxygen/architecture.h,
include/asterisk/bridge_channel_internal.h: Fix some doxygen
bridging file references.
* res/parking/parking_bridge_features.c, main/cdr.c, main/data.c,
main/manager.c, tests/test_jitterbuf.c, main/features.c,
tests/test_voicemail_api.c, main/file.c, tests/test_cel.c,
main/stasis_channels.c, main/bridge_channel.c, main/message.c,
tests/test_cdr.c, main/db.c, main/xmldoc.c, main/format.c,
res/res_rtp_asterisk.c, main/pbx.c, main/rtp_engine.c,
tests/test_abstract_jb.c, channels/chan_sip.c, main/pickup.c,
apps/app_queue.c, main/indications.c: Doxygen comment tweaks.
* main/utils.c, main/hashtab.c: Fix utilities compilation/linking.
The horrid structure of the source in the utils directory strikes
again. Moved the _ast_mem_backtrace_buffer[] definition from the
logical location in utils.c to hashtab.c so the aelparse and
conf2ael utilities can link.
* include/asterisk/utils.h: utils.h: Minor formatting tweaks.
2013-08-16 16:03 +0000 [r396842] David M. Lee <dlee@digium.com>
* main/stasis.c, main/stasis_cache_pattern.c, main/stasis_cache.c,
include/asterisk/astobj2.h, main/stasis_channels.c,
tests/test_stasis.c: Stasis: address refcount races;
implementation comments Change r395954 reordered some stasis
object destruction, which should have been fine. Unfortunately,
it caused some hard to reproduce issues related to objects being
accessed after they had been destroyed. The patch in r396329
fixed the destruction order problem; this patch addresses the
underlying issue. A few other stasis-related fixes were also
added. * Add ref-bumps around areas where objects may get
transitively destroyed. (For example, where we lock a topic,
unref a subscription, which unrefs the topic, which explodes the
topic when we try to unlock it.) * Wrote an extensive doxygen
page about Stasis implementation, relationships between objects,
lifecycles of objects, how the refcounting works, etc. Many other
comments were added, corrected, or cleaned up. * Added an assert
to the topic dtor to catch extra ref decrements. * Fixed type
used after destruction errors for graceful shutdown in
stasis_channels.c. * I added two unit tests in an attempt to
catch destruction order issues. Since the underlying cause is a
race condition, though, the tests rarely failed even when the
code was wrong. * Fixed a leak in stasis_cache_pattern.c. (closes
issue ASTERISK-22243) Review:
https://reviewboard.asterisk.org/r/2746/
2013-08-16 12:20 +0000 [r396829] Kinsey Moore <kmoore@digium.com>
* main/utils.c, main/sounds_index.c, main/loader.c: Improve sounds
indexer CLI commands This reworks the CLI commands used to access
sounds information from "sounds show[ soundid]" to "core show
sounds" and "core show sound <soundid>". This also reworks the
"sounds reload" CLI command to fall under normal module reloading
("module reload sounds"). Also, make trunk build when
DEBUG_MALLOC is not enabled. Review:
https://reviewboard.asterisk.org/r/2745/ (closes issue
ASTERISK-22141)
2013-08-16 07:18 +0000 [r396822] Walter Doekes <walter+asterisk@wjd.nu>
* include/asterisk/utils.h, main/pbx.c, main/utils.c: Prevent heap
alloc functions from running out of stack space. When asterisk
has run out of memory (for whatever reason), the alloc function
logs a message. Logging requires memory. A recipe for infinite
recursion. Stop the recursion by comparing the function call
depth for sane values before attempting another OOM log message.
Review: https://reviewboard.asterisk.org/r/2743/
2013-08-15 22:10 +0000 [r396783-396814] Richard Mudgett <rmudgett@digium.com>
* main/bridge_channel.c: Bridge: Don't suspend/unspend the channel
for interception routines. By their nature, the connected line
and redirecting interception routines are not supposed to affect
the channel's media. Therefore, they should not suspend and
unsuspend the channel while running. The suspend/unsuspend
operations could be expensive depending upon the bridge and
channel technology involved.
* res/parking/res_parking.h, res/res_parking.c,
res/parking/parking_tests.c, main/features.c: Minor parking
cleanup.
* res/parking/parking_bridge_features.c: Parking: Eliminate local
channel name hack to get peer channel. (closes issue
ASTERISK-22034) Reported by: Matt Jordan
* main/bridge_channel.c, main/features.c: Remove early bridge
BUGBUG comments. Remove some unneeded features.c comments.
* configs/features.conf.sample: Update features.conf.sample
atxferdropcall option.
* main/bridge.c, include/asterisk/bridge_channel.h,
main/config_options.c, main/bridge_channel.c,
apps/confbridge/conf_config_parser.c: Changed some BUGBUG tags to
associated JIRA issue tags.
* main/bridge.c, main/features.c, bridges/bridge_softmix.c,
include/asterisk/bridge.h: Resolve some BUGBUG comments.
2013-08-15 16:37 +0000 [r396747] Kinsey Moore <kmoore@digium.com>
* main/asterisk.c, main/cli.c, /: Remove leading spaces from the
CLI command before parsing If you've mistakenly put a space
before typing in a command, the leading space will be included as
part of the command, and the command parser will not find the
corresponding command. This patch rectifies that situation by
stripping the leading spaces on commands. Review:
https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman
Lesher ........ Merged revisions 396745 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 396746 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-15 15:12 +0000 [r396732-396734] Richard Mudgett <rmudgett@digium.com>
* channels/chan_vpb.cc, main/features.c,
include/asterisk/channel.h, channels/chan_iax2.c: Remove some
dead code dealing with: AST_BRIDGE_REC_CHANNEL_0,
AST_BRIDGE_REC_CHANNEL_1, and AST_BRIDGE_IGNORE_SIGS.
* include/asterisk/bridge_channel_internal.h, main/manager.c,
main/bridge_channel.c: Fix Bridge API DTMF hook matching for
begin and end DTMF events. The Bridge API DTMF hook matching
would not deal with DTMF end events only. It required a DTMF
begin event to start matching the DTMF hooks. There are many
places in Asterisk where code only generates DTMF end events
without the corresponding begin event. One such place is the AMI
action Atxfer. * Fixed DTMF hook matching if there is a string of
DTMF frames in the read queue. We could potentially miss some of
them before. * Fixed AMI Atxfer action documentation. (closes
issue ASTERISK-22037) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2752/
2013-08-15 12:17 +0000 [r396722-396724] Kinsey Moore <kmoore@digium.com>
* apps/app_confbridge.c, main/bridge.c, main/features.c: Fix
feature_attended_transfer test The feature_attended_transfer test
is failing due to Asterisk not passing DTMF in the bridges
created for internal attended transfers. This sets the features
initialization routine to set this flag by default and adjusts
the basic bridge and confbridge's use of the bridging system
accordingly as per Richard's suggestion instead of adjusting this
individual case. This change allows the necessary DTMF to pass
through the attended transfer bridge and complete the test
successfully. Review: https://reviewboard.asterisk.org/r/2759/
(closes issue ASTERISK-22222)
* main/utils.c, include/asterisk/lock.h, channels/chan_sip.c: Fix
deadlocks in chan_sip in REFER and BYE handling This resolves
several deadlocks in chan_sip relating to usage of
ast_channel_bridge_peer and improves accessibility of lock
debugging function calls. Review:
https://reviewboard.asterisk.org/r/2756/ (closes issue
ASTERISK-22215)
* res/res_stasis.c: Prevent automagic things from happening to
Stasis application bridges This prevents swap optimization,
merges, and transfers involving Stasis application bridges. It
wouldn't be nice if the bridge you thought you owned disappeared
from under you. Reported-by: Richard Mudgett
2013-08-15 00:16 +0000 [r396695-396713] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, main/channel.c, channels/chan_vpb.cc:
Remove unsupported channel technology callbacks.
* channels/chan_vpb.cc: chan_vpb: Effectively remove native
support. Left enough bread crumbs to be able to convert later if
needed.
* channels/chan_iax2.c: chan_iax2: Conditionally remove native
support for now. (issue ASTERISK-21944)
* channels/chan_misdn.c: chan_misdn: Effectively remove native
support. Left enough bread crumbs to be able to convert later if
needed.
* apps/app_bridgewait.c: app_bridgewait: Inhibit local channel
optimizations to the bridge. Holding bridges can allow local
channel move/swap optimization to the bridge. However, we cannot
allow it for the BridgeWait holding bridge because the call will
lose the channel roles and dialplan location as a result.
2013-08-14 19:06 +0000 [r396621-396658] Joshua Colp <jcolp@digium.com>
* /, tests/test_hashtab_thrash.c: Tweak comment for why usleep is
used. ........ Merged revisions 396656 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 396657 from
http://svn.asterisk.org/svn/asterisk/branches/11
* tests/test_hashtab_thrash.c, /: Tweak test_hashtab_thrash test to
allow the critical threads to execute. Depending on certain
conditions it was possible for the hashtab counting thread to
starve other threads, preventing them from executing in the
expected fashion. This change adds a sleep to allow the others to
do what they need to do. While this doesn't thrash the hashtab as
much as previously, it at least works. (closes issue
ASTERISK-22276) Reported by: Matt Jordan ........ Merged
revisions 396619 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 396620 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-13 18:47 +0000 [r396581-396584] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: chan_sip: Convert 'just did sched_add
waitid...' from warning to debug message. Patches:
reviewboard-2377.patch uploaded by Paul Belanger Review:
https://reviewboard.asterisk.org/r/2377/ ........ Merged
revisions 396582 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 396583 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/
........ Merged revisions 396579 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 396580 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-13 15:27 +0000 [r396559-396568] David M. Lee <dlee@digium.com>
* include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c,
res/stasis/control.h, include/asterisk/bridge_internal.h,
include/asterisk/bridge_features.h, res/res_stasis.c,
res/ari/resource_bridges.c, res/res_stasis_bridge_add.c
(removed), res/res_stasis_playback.c, res/stasis/control.c,
res/res_stasis_bridge_add.exports.in (removed),
include/asterisk/stasis_app.h: ARI: allow other operations to
happen while bridged This patch changes ARI bridging to allow
other channel operations to happen while the channel is bridged.
ARI channel operations are designed to queue up and execute
sequentially. This meant, though, that while a channel was
bridged, any other channel operations would queue up and execute
only after the channel left the bridge. This patch changes ARI
bridging so that channel commands can execute while the channel
is bridged. For most operations, things simply work as expected.
The one thing that ended up being a bit odd is recording. The
current recording implementation will fail when one attempts to
record a channel that's in a bridge. Note that the bridge itself
may be recording; it's recording a specific channel in the bridge
that fails. While this is an annoying limitation, channel
recording is still very useful for use cases such as voice mail,
and bridge recording makes up much of the difference for other
use cases. (closes issue ASTERISK-22084) Review:
https://reviewboard.asterisk.org/r/2726/
* tests/test_hashtab_thrash.c: Missed a spot in r396559
* tests/test_hashtab_thrash.c: Fix build warnings when printf a
tv_usec. The debug logs added in r396528 neglected to account for
suseconds_t being an int. See r392076 for more info.
2013-08-12 22:05 +0000 [r396552] John Bigelow <jbigelow@digium.com>
* res/res_pjsip_registrar.c: Add test suite events for when
contacts are added or removed from an AOR These are needed by the
pjsip inbound registration test suite tests. (issue
ASTERISK-21833) (issue ASTERISK-21834) (issue ASTERISK-21835)
(issue ASTERISK-21837) Review:
https://reviewboard.asterisk.org/r/2700/ Review:
https://reviewboard.asterisk.org/r/2739/
2013-08-12 15:59 +0000 [r396542-396543] Matthew Jordan <mjordan@digium.com>
* main/bridge_channel.c, main/bridge.c, main/features.c: Fix two
race conditions and ref counting issue when joining a bridge
These problems were all caught by a test in the Asterisk Test
Suite that originated some Local channels and attempted to move
the ;2 half of the Local channel into a bridge using the Bridge
AMI action. (1) When originating a channel, the Newchannel event
is emitted quickly; however, the ;2 channel will not have a pbx
thread assigned to it until after the outbound 'dialing' for the
;1 is complete. Thus, there is a period of time where the outside
world "knows" of the channel's existence and can influence it but
Asterisk has not yet started the dialplan execution thread. If a
Bridge AMI action is taken on the channel, the channel appears to
be a Dialed channel with no PBX thread; hence, the channel will
be imparted into the Bridge by first 'yanking' the channel. At
the same time, a race condition can occur after the yank (but
before entering the bridge) when ;1 answers and starts a PBX on
the ;2. The end result currently is an assertion failure in the
Bridging API, as a channel with a PBX is imparted into the
Bridge. There's no way to prevent AMI from attempting to Bridge a
channel immediately after creation; likewise, holding the channel
lock through the entire Dial operation is unwise (and
impossible). Instead of treating the presence of a PBX thread as
an error, we simply bail out of the adding the channel to the
bridge through ast_bridge_impart. The Bridge action will then
fail - but we avoid a situation where the channel is both
executing a PBX thread and simultaneously being given a separate
thread in the bridging system (which would be a "bad thing").
Since imparting a channel with a PBX *can* occur and is not a
programming error, the asserts have been removed. (2) When the
first condition occurs, we have to take one of two actions:
either hangup the yanked channel as it did not enter the bridge,
or deref it because we don't own it. We can determine if we own
it or not by testing for the presence of the PBX thread. If we
hung it up directly, we'd crash. (3) bridge_find_channel does not
increase the reference count of the ast_bridge_channel object.
The RAII_VAR usage in ast_bridge_add_channel thus created a
ticking time bomb in whatever bridge the channel moved into, as
the destructor for the ast_bridge_channel object would be called.
Review: https://reviewboard.asterisk.org/r/2741/
* main/pbx.c: Unlock outgoing dial lock on off nominal path If the
thread servicing the dial request isn't created successfully, the
outgoing dial lock will still be held when the function returns.
This patch unlocks the lock on this off nominal path.
2013-08-10 20:29 +0000 [r396521-396535] Matthew Jordan <mjordan@digium.com>
* tests/test_hashtab_thrash.c: Pipe test output through test object
not stdout Otherwise, it doesn't show up in the automated test
failures
* tests/test_hashtab_thrash.c: Add some debugging when
test_hashtab_thrash fails Disabling DEBUG_THREADS caused this
test to fail on the 32-bit build agent. Adding some debugging to
see why it thinks the test is timing out.
* main/pbx.c: Unlock the dial operation lock on a failed dial If a
dial operation fails, the pbx_outgoing_attempt routine will exit
without first having unlocked the outgoing dial lock. This would
be a "bad thing".
2013-08-09 21:50 +0000 [r396512] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_native_rtp.c: bridge_native_rtp: Remove some
unnecessary NULL checks on c1.
2013-08-09 20:29 +0000 [r396505] Walter Doekes <walter+asterisk@wjd.nu>
* main/autoservice.c: Don't leak frames when memory is full in
autoservice_run. Review: https://reviewboard.asterisk.org/r/2566/
2013-08-09 17:28 +0000 [r396497-396498] Jonathan Rose <jrose@digium.com>
* main/pbx.c, channels/chan_sip.c: pbx: Make originate threads
indicate dial status when synchronous This makes it so that we
can detect failures to originate as with earlier versions of
Asterisk, which restores the Asterisk 11 behavior for the
originate manager action. This was causing the ACL tests for SIP
and IAX2 to fail since those tests expected originate failures
when ACLs would cause rejections. Also, this patch fixes crashes
in chan_sip when ACLs rejected peers during registration
verification. (closes issue ASTERISK-22212) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/2753/
* main/core_unreal.c, main/bridge_channel.c,
include/asterisk/bridge.h, res/ari/resource_bridges.c,
include/asterisk/core_unreal.h: bridge_channel: Support the
lonely flag and make ARI use it. The lonely flag is an optional
flag for bridge channels that will make them leave a bridge when
a channel leaves if only lonely channels are in the bridge at
that point. This is useful for things like ending recording and
playback channels when they cease to be interacting with other
channels in the bridge. (closes issue ASTERISK-22117) Reported
by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2721/
2013-08-09 13:58 +0000 [r396490] Matthew Jordan <mjordan@digium.com>
* apps/confbridge/conf_config_parser.c: Update documentation for
ConfBridge with some additional markup Add some additional markup
for items that needed it, e.g., replaceable tags, literal tags,
etc.
2013-08-08 22:57 +0000 [r396480] Richard Mudgett <rmudgett@digium.com>
* tests/test_stasis.c: Fix stasis/core unit test. Should have had
the CR/LF.
2013-08-08 22:09 +0000 [r396474] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: chan_dahdi: create channels at run-time
This code adds chan_dahdi the command 'dahdi create channels
<range>' (where <range> is a single <n>-<m> or 'new') and updates
'dahdi destroy channel' with a similar 'dahdi destroy channels'.
It allows DAHDI channels and spans to be added after the initial
channel load (without destroying all other channels as in 'dahdi
restart'). It also includes some fixes to the D-Channel / span
destruction code (r394552). This change is intended to provide a
hook for a script running from udev once a span has been assigned
("registered") / unassigned ("unregistered") for its channels.
The udev hook configures the span's channels with dahdi_cfg -S,
and can then ask Asterisk to create ethe channels. See the
scripts added to DAHDI-tools in 2.7.0. Review:
https://reviewboard.asterisk.org/r/1598/
2013-08-08 20:52 +0000 [r396417-396463] Richard Mudgett <rmudgett@digium.com>
* tests/test_stasis.c: Add missing CR/LF to FakeMI stasis test AMI
event.
* main/stasis_bridges.c: Remove extra CR/LF from AMI event.
* main/manager_bridges.c, apps/confbridge/confbridge_manager.c,
include/asterisk/manager.h, main/stasis_bridges.c: Make bridge
snapshots use prefixes. * Changed
ast_manager_build_bridge_state_string() to assume an empty prefix
string just like ast_manager_build_channel_state_string(). *
Created ast_manager_build_bridge_state_string_prefix() to work
just like ast_manager_build_channel_state_string_prefix(). * Made
BridgeMerge AMI event use To/From prefixes.
2013-08-08 18:40 +0000 [r396412] Matthew Jordan <mjordan@digium.com>
* formats/format_wav_gsm.c: Improve disk writes for wav49 format
Writing to a file in the wav49 format performs rather
inefficiently. The procedure is approximately: (1) Write GSM
frame to the end of the file (2) Seek to the end of the file (3)
Seek to the header (4) Update the file size (5) Seek (again) to
the end of the file (6) Repeat This pattern negates any attempt
to use the stdio buffering setup in ast_writefile. It also
results in many small writes that require a seek going to the
disk each second which translates to poor disk performance on
certain file systems, particularly when there are multiple wav49
files being written simultaneously. (closes issue ASTERISK-19595)
Reported by: Byron Clark Tested by: Byron Clark patches:
gsm_wav_only_update_header_on_close.patch uploaded by byronclark
(License 6157)
2013-08-08 17:51 +0000 [r396401] Richard Mudgett <rmudgett@digium.com>
* main/channel_internal_api.c, main/features.c,
include/asterisk/bridge_features.h, main/bridge.c: Remove some
resolved or obsolete BUGBUG comments.
2013-08-08 14:13 +0000 [r396391-396392] Matthew Jordan <mjordan@digium.com>
* apps/confbridge/conf_chan_announce.c, main/manager_channels.c,
main/channel.c, main/manager_bridges.c,
channels/chan_bridge_media.c, apps/confbridge/conf_chan_record.c,
main/channel_internal_api.c, include/asterisk/channel.h,
main/cel.c: Hide the Surrogate channels from external consumers;
kill Masquerade events This patch does three things: 1. It
provides a Surrogate channel technology with a consolidated
"implementation detail flag" on the channel technology. This
tells consumers of Stasis that the creation of this channel is an
implementation detail in Asterisk and can be ignored (if they so
choose). This consolidates the conference recorder/announcer
flags as well - these flags had no additional meaning beyond
"ignore this channel please". 2. It modifies allocation of a
channel in two ways: (a) If a channel technology can be
determined from the name, we set it directly in the allocation
routine. This prevents the initial publication of the message
from going out with a NULL channel technology where possible.
This lets Stasis consumers get the right channel technology on
the first publication. (b) It reorganizes allocation to make use
of the 'finalized' property on the channel. This was already used
to know that a channel had completely finished its construction
in the masquerade routine; now we also use it to know whether or
not the setting of certain channel properties is occurring during
or post construction. The various set routines were modified
accordingly as well. 3. The masquerade event is now dead, Jim. It
no longer served any purpose whatsoever - if you perform a call
pickup you'll get a Pickup event; if you perform an attended
transfer you will still get those events; if you steal a channel
to put it elsewhere you'll get the corresponding NewExten or
BridgeEnter events. Review:
https://reviewboard.asterisk.org/r/2740
* main/utils.c: Prevent spurious memory error when appending
backtrace with MALLOC_DEBUG Backtraces are allocated outside of
the usual memory tracking performed by MALLOC_DEBUG. This allows
them to be used by the memory tracking enabled by that build
option; however, it also means that when backtraces are disposed
of they have to be done so outside of the re-defined free. This
patch undef's free prior to disposing of the allocated backtrace
when a backtrace is appended as a result of 'core show locks'.
2013-08-08 12:38 +0000 [r396385] Kinsey Moore <kmoore@digium.com>
* main/bridge.c: Prevent unreal channels from optimizing during
DTMF emulation This prevents unreal channel optimization during
the prequalification phase when either channel is involved in
DTMF emulation. This prevents a situation where an emulated digit
would be missed because the emulation was never completed.
Review: https://reviewboard.asterisk.org/r/2747/ (closes issue
ASTERISK-22214)
2013-08-08 07:05 +0000 [r396378] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: - Fix different issues with call
transfer cancel. In case 3rd party busy or congestion call was
not returned. - Fix displaying soft button 'Redial' in case of no
redial number exists ........ Merged revisions 396377 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-08 02:58 +0000 [r396365-396371] Matthew Jordan <mjordan@digium.com>
* main/cdr.c: Handle Surrogate channels in Dial message processing
Depending on when a Surrogate channel replaces an existing
channel, it is possible to get a Dial message for the Surrogate
channel. When this occurs, no CDR will exist for the channel as
Surrogate channels are ignored. Safely handle the case when a CDR
doesn't exist for a Dial message.
* apps/app_queue.c: Perform Ring-No-Answer checks before processing
Hangup logic The rna() routine will raise a Stasis message
involving both the caller and the agent. This doesn't work so
well if we already hung up the agent channel, as the channel
doesn't quite exist. Not surprisingly, this will crash. This
patch properly runs the rna subroutine (performing all of the
Ring-No-Answer logic) prior to hanging up the agent channel.
(closes issue ASTERISK-22258) Reported by: Kiril Valchev Tested
by: Kiril Valchev
2013-08-06 21:20 +0000 [r396329-396347] David M. Lee <dlee@digium.com>
* apps/app_meetme.c: Fixed app_meetme for cache split changes
* include/asterisk/frame.h, rest-api/api-docs/recordings.json,
res/ari/resource_recordings.c, apps/app_voicemail.c,
main/channel.c, res/res_ari_recordings.c, include/asterisk/app.h,
include/asterisk/stasis_app_recording.h,
res/ari/resource_recordings.h, funcs/func_frame_trace.c,
apps/app_minivm.c, main/app.c, res/res_stasis_recording.c: ARI:
Add recording controls This patch implements the controls from
ARI recordings. The controls are: * DELETE
/recordings/live/{recordingName} - stop recording and discard it
* POST /recordings/live/{recordingName}/stop - stop recording *
POST /recordings/live/{recordingName}/pause - pause recording *
POST /recordings/live/{recordingName}/unpause - resume recording
* POST /recordings/live/{recordingName}/mute - mute recording
(record silence to the file) * POST
/recordings/live/{recordingName}/unmute - unmute recording. Since
this underlying functionality did not already exist, is was added
to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even
though the ARI controls are idempotent, to be consistent with the
playback control frames. (closes issue ASTERISK-22181) Review:
https://reviewboard.asterisk.org/r/2697/
* main/stasis_cache_pattern.c, main/stasis_cache.c,
include/asterisk/stasis.h, tests/test_stasis.c: Tweak caching
topics to fix CEL tests The Stasis changes in r395954 had an
unanticipated side effect: messages published directly to an _all
topic does not get forwarded to the corresponding caching topic.
This patch fixes that by changing how caching topics forward
messages, and how the caching pattern forwards are setup. For the
caching pattern, the all_topic is forwarded to the
all_topic_cached. This forwards messages published directly to
the all_topic to all_topic_cached. In order to avoid duplicate
messages on all_topic_cached, caching topics were changed to no
longer forward uncached messages. Subscribers to an individual
caching topic should only expect to receive cache updates, and
subscription change messages. Since individual caching topics are
new, this shouldn't be a problem. There are a few minor changes
to the pre-cache split behavior. * For topics changed to use the
caching pattern, the all_topic_cached will forward snapshots in
addition to cache updates. Since subscribers by design ignore
unexpected messages, this should be fine. * Caching topics that
don't use the caching pattern no longer forward non-cache
updates. This makes no difference for the current caching topics.
* mwi_topic_cached, channel_by_name_topic and
presence_state_topic_cached have no subscribers *
device_state_topic_cached's only subscriber only processes cache
udpates (issue ASTERISK-22243) Review:
https://reviewboard.asterisk.org/r/2738
2013-08-06 13:08 +0000 [r396320-396321] Kinsey Moore <kmoore@digium.com>
* res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c,
res/res_pjsip/config_system.c: Expose res_pjsip threadpool
options Expose initial size, automatic increment, maximum size,
and idle timeout as configurable parameters for the res_pjsip
thread pool. Review: https://reviewboard.asterisk.org/r/2704/
(closes issue ASTERISK-22143)
* main/cdr.c: Fix memory leaks in the CDR engine Fix refcount bugs
and a possible locking problem in the CDR engine relating to use
of ao2_iterators. Review:
https://reviewboard.asterisk.org/r/2724/ (closes issue
ASTERISK-22126)
2013-08-06 12:39 +0000 [r396319] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_notify.c, res/res_pjsip_outbound_registration.c,
res/res_pjsip_messaging.c, res/res_pjsip_exten_state.c: Fix crash
in res_pjsip_outbound_registration when the remote server can not
be resolved. This crash was caused by decrementing the reference
count of a newly created message when it should not be. This
change fixes that but also fixes all other cases where this was
incorrectly done. (closes issue ASTERISK-22188) Reported by:
Kinsey Moore
2013-08-06 08:43 +0000 [r396309-396311] Walter Doekes <walter+asterisk@wjd.nu>
* /, funcs/func_strings.c: Check result of ast_var_assign() calls
for memory allocation failure (2). Missed a spot in the previous
commit. ........ Merged revisions 396310 from
http://svn.asterisk.org/svn/asterisk/branches/11
* pbx/pbx_dundi.c, utils/extconf.c, apps/app_stack.c,
apps/app_playback.c, funcs/func_global.c, main/cdr.c,
pbx/pbx_loopback.c, main/pbx.c, /, funcs/func_strings.c: Check
result of ast_var_assign() calls for memory allocation failure.
We try to keep the system running even when all available memory
is spent. Review: https://reviewboard.asterisk.org/r/2734/
........ Merged revisions 396279 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 396287 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-05 20:20 +0000 [r396253] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix Registration Failure When A Peer And
TLS Are Used If a peer is used in a register line and TLS is
defined as the transport, the registration fails since the
transport on the dialog is never set properly resulting in UDP
being used instead of TLS. This patch sets the dialog's transport
based on the transport that was defined in the register line. If
the register line does not specify a transport, the parsing
function for the register line always defaults back to UDP.
(closes issue ASTERISK-21964) Reported by: Doug Bailey Tested by:
Doug Bailey Patches: asterisk-21964-set-reg-dialog-transport.diff
by Michael L. Young (license 5026) ........ Merged revisions
396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 396248 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-05 20:18 +0000 [r396245] Jonathan Rose <jrose@digium.com>
* main/bridge_basic.c, main/features.c,
include/asterisk/bridge_basic.h: bridge features: Dial and Queue
add features instead of replace them. Dial and Queue would
previously apply a new set of features whenever bridging. These
options would be based purely on the options supplied to the
dial/queue applications. This patch changes the function those
applications use to bridge calls so that the features will be
added to the set of existing features for each channel rather
than having them override the existing features. (closes issue
ASTERISK-22209) Reported by: Jonathan Rose Review:
https://reviewboard.asterisk.org/r/2713/
2013-08-05 19:01 +0000 [r396201] Matthew Jordan <mjordan@digium.com>
* res/res_pjsip_outbound_registration.c: Add AMI registration
events for PJSIP outbound registration attempts This patch adds
AMI events whenever an outbound registration attempt succeeds or
fails from res_pjsip_outbound_registration. This brings it inline
with the existing SIP channel driver and IAX channel driver.
Review: https://reviewboard.asterisk.org/r/2729/
2013-08-05 18:52 +0000 [r396198-396200] Michael L. Young <elgueromexicano@gmail.com>
* /, UPGRADE-11.txt: Change "from" to "From". (related to issue
ASTERISK-21903) ........ Merged revisions 396199 from
http://svn.asterisk.org/svn/asterisk/branches/11
* UPGRADE-11.txt, /: Adding a note to UPGRADE.txt about a change
made to res_agi in order to indicate when streaming an audio file
fails like it is done in other parts of the code to indicate an
error. Note was requested by Paul Belanger:
http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
(related to issue ASTERISK-21903) ........ Merged revisions
396196 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 396197 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-08-05 17:48 +0000 [r396175-396189] Jonathan Rose <jrose@digium.com>
* bridges/bridge_holding.c: bridge_holding: Add suspsend/unsuspend
callbacks Suspend and unsuspend callbacks are added to the
holding bridge so that entertainment can be disabled and
re-enabled when operations would suspend a channel on the bridge
(such as playback operations). This fixes entertainment so that
when those operations end, the entertainment can pick back up and
it also serves as an optimization. Also, this patch fixes a bug
caused by triggering ringing frames immediately instead of
pushing them to the queue which created a race condition where
sometimes parking with ringing during attended transfers would
cause the ringing to be interrupted by an unhold frame. (closes
issue ASTERISK-22006) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2711/
* res/res_ari_bridges.c, include/asterisk/bridge_roles.h,
res/ari/resource_bridges.h, res/stasis/control.c,
include/asterisk/stasis_app.h, main/bridge_roles.c,
rest-api/api-docs/bridges.json, res/ari/resource_bridges.c: ARI:
bridges/{bridgeID}/addChannel: add roles parameter Roles are now
cleared with each entry into a bridge with addChannel. If the
roles parameter is present, the role specified will be applied to
all channels being added with the addChannel command. (closes
issue ASTERISK-21973) Reported by: Matt Jordan
https://reviewboard.asterisk.org/r/2691/
* res/parking/res_parking.h, res/res_parking.c,
res/parking/parking_tests.c (added),
res/parking/parking_bridge.c: res_parking: Unit tests Adds the
following unit tests: * create_lot: tests adding and removal of a
new parking lot (baseline) * park_extensions: creates a parking
lot that registers extensions and then confirms that all of the
expected extensions exist * extensions_conflicts: creates
numerous parking lots to test that extension conflicts in parking
lots result in parking lot creation failing *
dynamic_parking_variables: Tests that the creation of dynamic
parking lots respects the related channel variables set on the
channel that requests them. * park_call: Tests adding a channel
to a parking lot's holding bridge by standard parking functions.
* retrieve_call: Tests pulling a channel out of a parking lot's
holding bridge via parked call retrieval functions. (closes issue
ASTERISK-22138) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2714/
2013-08-05 14:35 +0000 [r396166] David M. Lee <dlee@digium.com>
* main/asterisk.c, main/cli.c, main/channel.c, main/pbx.c,
main/manager.c, res/ari/resource_asterisk.c, utils/extconf.c,
include/asterisk/options.h: Fix res_ari_asterisk load issue The
new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the
module won't load on Linux. This patch renames the variables,
adding the ast_ prefix so they will be exported. Review:
https://reviewboard.asterisk.org/r/2737
2013-08-03 03:53 +0000 [r396158] Matthew Jordan <mjordan@digium.com>
* main/manager_bridges.c: Don't unsubscribe from the AMI message
router from manager_bridges The AMI message router is owned
wholly by manager.c. Previously, each of the manager_{item}
source files had their own message router and they unsubscribed
from each; once they moved over to using a single message router
only a single unsubscribe became necessary.
2013-08-02 17:50 +0000 [r396145] Mark Michelson <mmichelson@digium.com>
* channels/sig_pri.c: And get rid of another ast_bridged_channel()
2013-08-02 17:29 +0000 [r396136-396143] David M. Lee <dlee@digium.com>
* main/stasis_bridges.c: Clean up ast_json with ast_json_unref
* /: Removed svnmerge-integrated from trunk
2013-08-02 15:01 +0000 [r396126] Mark Michelson <mmichelson@digium.com>
* res/snmp/agent.c: Get the SNMP code to compile.
2013-08-02 14:46 +0000 [r396119-396125] David M. Lee <dlee@digium.com>
* res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
rest-api/api-docs/asterisk.json, res/ari/resource_asterisk.c: ARI
- GET /ari/asterisk/info This patch adds basic system information
access to ARI. The results are roughly what you get from 'core
show settings', with a few minor differences. * Data is
structured, with 'build', 'system', 'config' and 'status'
sub-objects. * Each sub-object is selectable, using the ?only=
parameter. A comma separated list can be provided to select
multiple sections. * A few config options are numeric, for which
0 means 'unlimited'. Instead of having a special interpretation
of those fields, they are simply omitted if they're 0. * The
information is limited to what might be useful to building
external applications. (closes issue ASTERISK-21575) Review:
https://reviewboard.asterisk.org/r/2702/
* rest-api-templates/param_cleanup.mustache (added),
rest-api/api-docs/events.json, /, res/ari/resource_events.c,
rest-api-templates/ari_resource.h.mustache,
res/res_ari_asterisk.c, res/res_ari_playback.c,
rest-api-templates/res_ari_resource.c.mustache,
res/ari/resource_events.h, rest-api/api-docs/sounds.json,
res/res_ari_channels.c, rest-api/api-docs/bridges.json,
rest-api-templates/param_parsing.mustache,
res/ari/resource_bridges.c, res/ari/resource_sounds.h,
res/res_ari_recordings.c, res/ari/resource_bridges.h,
res/res_ari_endpoints.c, res/res_ari_events.c,
res/ari/resource_asterisk.h, rest-api/api-docs/channels.json,
res/res_ari_sounds.c, res/res_ari_bridges.c: ARI - implement
allowMultiple for parameters Swagger allows parameters to be
specified as 'allowMultiple', meaning that the parameter may be
specified as a comma separated list of values. I had written some
of the API docs using that, but promptly forgot about
implementing it. This patch finally fills in that gap. The
codegen template was updated to represent 'allowMultiple' fields
as array/size fields in the _args structs. It also parses the
comma separated list using ast_app_separate_args(), so quoted
strings in the argument will be handled properly. Review:
https://reviewboard.asterisk.org/r/2698/
* tests/test_json.c, main/json.c, res/res_sorcery_astdb.c,
include/asterisk/json.h, main/cel.c, res/ari/ari_websockets.c:
Address JSON thread safety issues. In tracking down some unit
tests failures, I ended up reading the fine print[1] regarding
Jansson's thread safety. In short: 1. Ref-counting is non-atomic.
2. json_dumps() and friends are not thread safe. This patch adds
locking where necessary to our ast_json_* wrapper API, with
documentation in json.h describing the thread safety limitations
of the API. [1]:
http://www.digip.org/jansson/doc/2.4/portability.html#thread-safety
Review: https://reviewboard.asterisk.org/r/2716/
2013-08-02 14:13 +0000 [r396107] Mark Michelson <mmichelson@digium.com>
* main/cel.c, include/asterisk/parking.h, main/bridge_channel.c,
main/stasis_bridges.c, res/parking/parking_manager.c,
res/parking/parking_bridge.c, main/manager_bridges.c,
include/asterisk/stasis_bridges.h: Make a couple of changes to
help AMI events to be more clear in what is occurring. *
BridgeEnter now contains the unique ID of the channel that is to
be swapped out, if applicable. * There is a ParkedCallSwap event
that is sent when a parked channel has a new channel take its
place. (closes issue ASTERISK-22193) reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/2712
2013-08-02 14:08 +0000 [r396105] Kinsey Moore <kmoore@digium.com>
* include/asterisk/strings.h, main/astobj2.c, utils/Makefile,
utils/refcounter.c, main/strings.c, include/asterisk/astobj2.h:
Move ast_str_container_alloc and friends This moves
ast_str_container_alloc, ast_str_container_add,
ast_str_container_remove, and related private functions into
strings.c/h since they really don't belong in astobj2.c/h. As a
result of this move, utils also had to be updated. Review:
https://reviewboard.asterisk.org/r/2719/ (closes issue
ASTERISK-22041)
2013-08-02 14:05 +0000 [r396102-396103] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, channels/chan_skinny.c,
funcs/func_channel.c, main/channel_internal_api.c,
include/asterisk/channel.h, channels/chan_iax2.c,
apps/app_chanspy.c, channels/chan_oss.c, channels/chan_mgcp.c,
main/channel.c, channels/chan_dahdi.c, channels/chan_misdn.c,
main/rtp_engine.c: Get rid of ast_bridged_channel() and the
bridged_channel field on ast_channels. This commit is smaller
than the initial review placed on review board. This is because a
change to allow for channel drivers to access parking
functionality externally was committed and invalidated quite a
few of the changes initially made. (closes issue ASTERISK-22039)
reported by Matt Jordan Review:
https://reviewboard.asterisk.org/r/2717
* include/asterisk/pickup.h: Make sure that pickup.h does not use
an include guard name used elsewhere.
2013-08-02 13:29 +0000 [r396087-396099] Kinsey Moore <kmoore@digium.com>
* main/pickup.c: Correct the last of the Newchannel xi:includes
* res/res_pjsip_notify.c, res/res_pjsip_outbound_registration.c,
res/res_pjsip/include/res_pjsip_private.h,
res/res_pjsip/pjsip_options.c, res/res_pjsip.c: Add CLI/AMI
commands to force chan_pjsip actions For chan_pjsip, this
introduces CLI/AMI remote unregistration commands, reworks CLI
syntax for sending NOTIFYs, adds AMI qualification support, and
adds documentation for PJSIPNotify. This also fixes two
refcounting bugs in the outbound registration code. Review:
https://reviewboard.asterisk.org/r/2695/ (closes issue
ASTERISK-21939)
2013-08-02 04:48 +0000 [r396075] David M. Lee <dlee@digium.com>
* channels/sig_analog.c: Fixed chan_dahdi compilation failure
2013-08-02 03:12 +0000 [r396060-396062] Matthew Jordan <mjordan@digium.com>
* tests/test_cel.c, tests/test_cdr.c: Fix test modules More missing
include files. :-\
* channels/chan_dahdi.c, channels/chan_mgcp.c: Add pickup.h include
lines for chan_dahdi and chan_mgcp
* include/asterisk/parking.h, include/asterisk/pickup.h (added),
main/asterisk.c, res/parking/parking_manager.c, tests/test_cdr.c,
channels/chan_unistim.c, main/pbx.c, res/stasis/control.c,
main/pickup.c (added), channels/chan_sip.c, main/bridge.c,
UPGRADE.txt, res/parking/parking_applications.c,
include/asterisk/_private.h, channels/chan_gtalk.c, main/cel.c,
CHANGES, include/asterisk/features.h, main/cdr.c,
res/res_parking.c, channels/chan_skinny.c,
apps/app_directed_pickup.c, main/features.c, tests/test_cel.c:
Remove dead code from features.c; refactor pickup code into
pickup.c This patch does the following: * It moves the pickup
code out of features.c and into pickup.c * It removes the vast
majority of dead code out of features.c. In particular, this
includes the parking code. (issue ASTERISK-22134)
2013-08-01 23:38 +0000 [r396048] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_registrar.c: Fix a crash due to performing full URI
validation on a contact which only contains '*'. (closes issue
AST-1198) Reported by: John Bigelow
2013-08-01 21:19 +0000 [r396035] David M. Lee <dlee@digium.com>
* main/sorcery.c: Fix sorcery for some rather picky regex
implementations. Some regex implementations won't compile an
empty string. Assuming that it's equivalent of a regex that will
match anything, use ".?" instead.
2013-08-01 20:55 +0000 [r396010-396028] Matthew Jordan <mjordan@digium.com>
* channels/chan_skinny.c, main/parking.c, main/bridge.c,
main/features.c, channels/chan_iax2.c,
include/asterisk/parking.h, main/bridge_channel.c,
res/parking/parking_bridge_features.c, channels/chan_mgcp.c,
include/asterisk/features.h, channels/chan_dahdi.c,
res/res_parking.c, channels/sig_analog.c: Support externally
initiated parking requests; remove some dead code This patch does
the following: * It adds support for externally initiated parking
requests. In particular, chan_skinny has a protocol level message
that initiates a call park. This patch now supports that option,
as well as the protocol specific mechanisms in
chan_dahdi/sig_analog and chan_mgcp. * A parking bridge features
virtual table has been added that provides access to the parking
functionality that the Bridging API needs. This includes requests
to park an entire 'call' (with little or no additional
information, thank you chan_skinny), perform a blind transfer to
a parking extension, determine if an extension is a parking
extension, as well as the actual "do the parking" request from
the Bridging API. * Refactoring in chan_mgcp, chan_skinny, and
chan_dahdi to make use of the new functions * The removal of some
- but not all - dead parking code from features.c This also fixed
blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the
parking features kK might not have worked) Review:
https://reviewboard.asterisk.org/r/2710 (closes issue
ASTERISK-22134) Reported by: Matt Jordan
* CHANGES, apps/app_queue.c: Add queue member paused hints This
patch adds the ability in Queue to raise a hint when a member's
paused state changes. The hint uses the form
'Queue:{queue_name}_pause_{member_name}', where {queue_name} and
{member_name} are the name of the queue and the name of the
member to subscribe to, respectively. For example: exten =>
8501,hint,Queue:sales_pause_mark. Members will show as In Use
when paused. Note that the format of the queue pause hint was
changed slightly from what is on the issue to accomodate
suggestion on the code review. Review:
https://reviewboard.asterisk.org/r/2254 (closes issue
ASTERISK-20842) Reported by: Philippe Lindheimer patches:
qpause-10-378206.diff uploaded by Philippe Lindheimer (license
5519) qpause-11-378206.diff uploaded by Philippe Lindheimer
(license 5519) qpause-trunk-378206.diff uploaded by Philippe
Lindheimer (license 5519)
2013-08-01 17:23 +0000 [r395985-395998] Kinsey Moore <kmoore@digium.com>
* configure: Regenerate configure for configure.ac changes
* Makefile, apps/confbridge/confbridge_manager.c, makeopts.in,
doc/appdocsxml.dtd, apps/app_stack.c,
res/parking/parking_manager.c, main/manager_mwi.c,
main/rtp_engine.c, apps/app_meetme.c,
include/asterisk/autoconfig.h.in, main/xml.c,
main/stasis_bridges.c, contrib/scripts/install_prereq,
main/manager_bridges.c, channels/chan_dahdi.c, main/manager.c,
doc/snapshots.xslt (added), main/features.c, apps/app_minivm.c,
res/res_agi.c, main/stasis_channels.c, main/manager_channels.c,
channels/chan_sip.c, main/Makefile, configure.ac, UPGRADE.txt,
main/aoc.c, main/core_local.c, channels/sig_pri.c,
apps/app_queue.c, CHANGES, funcs/func_global.c,
apps/app_agent_pool.c: Fix documentation replication issues This
prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter
sets with a given prefix or defaulting to no prefix. This also
prevents documentation from becoming fractured and out of date by
keeping all variations of the documentation in template form such
that it only needs to be updated once and keeps maintenance to a
minimum. Review: https://reviewboard.asterisk.org/r/2708/
2013-08-01 16:56 +0000 [r395954-395984] David M. Lee <dlee@digium.com>
* utils/astman.c: Fixed warning in astman for gcc-4.8.
* res/res_pjsip_mwi.c, channels/chan_pjsip.c: Fixed compile errors
introduced in r395954. Just a merge error due to a file rename.
Grrr...
* main/manager.c, tests/test_devicestate.c, res/res_agi.c,
include/asterisk/stasis_cache_pattern.h (added), main/app.c,
main/stasis_channels.c, res/ari/resource_channels.c,
include/asterisk/stasis_endpoints.h, include/asterisk/bridge.h,
main/manager_channels.c, channels/chan_mgcp.c, main/pbx.c,
include/asterisk/devicestate.h, main/stasis_cache.c,
res/ari/resource_endpoints.c, channels/chan_sip.c,
main/channel_internal_api.c, include/asterisk/presencestate.h,
include/asterisk/stasis_bridges.h, include/asterisk/stasis.h,
include/asterisk/channel.h, channels/sig_pri.c, main/cel.c,
tests/test_stasis_endpoints.c, res/ari/resource_bridges.c,
include/asterisk/app.h, include/asterisk/stasis_channels.h,
apps/confbridge/confbridge_manager.c, tests/test_cel.c,
tests/test_stasis.c, res/res_stasis.c,
main/stasis_cache_pattern.c (added), apps/app_voicemail.c,
channels/chan_unistim.c, main/stasis_endpoints.c,
main/stasis_wait.c (added), apps/app_meetme.c,
res/stasis/control.c, main/bridge.c, main/manager_endpoints.c,
include/asterisk/channel_internal.h, main/devicestate.c,
res/res_xmpp.c, main/endpoints.c, channels/chan_iax2.c,
res/res_jabber.c, main/presencestate.c, main/stasis_bridges.c,
res/res_chan_stats.c, main/stasis.c, main/cli.c, main/cdr.c,
channels/chan_dahdi.c, main/manager_bridges.c: Split caching out
from the stasis_caching_topic. In working with res_stasis, I
discovered a significant limitation to the current structure of
stasis_caching_topics: you cannot subscribe to cache updates for
a single channel/bridge/endpoint/etc. To address this, this patch
splits the cache away from the stasis_caching_topic, making it a
first class object. The stasis_cache object is shared amongst
individual stasis_caching_topics that are created per
channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code
does not change. In making these changes, I noticed that we
frequently used a similar pattern for bridges, endpoints and
channels: single_topic ----------------> all_topic ^ |
single_topic_cached ----+----> all_topic_cached | +----> cache
This pattern was extracted as the 'Stasis Caching Pattern',
defined in stasis_caching_pattern.h. This avoids a lot of
duplicate code between the different domain objects. Since the
cache is now disassociated from its upstream caching topics, this
also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function,
which works for any stasis_topic. (closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
2013-08-01 11:21 +0000 [r395938] Joshua Colp <jcolp@digium.com>
* res/res_pjsip_session.c: Answer with multiple codecs if the
underlying pjproject supports it.
2013-08-01 00:07 +0000 [r395906-395907] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Raise Registry AMI events on registration
failures This patch makes it so that all registration attempts
that fail that also permanently modify the registration state
will raise an appropriate AMI event. Note that this patch was
forward ported to trunk and the Stasis Core message bus by
mjordan. (closes issue ASTERISK-21368) Reported by: Dmitriy Serov
patches: chan_sip.c.diff uploaded by Demon (license 6479)
* res/res_agi.c, CHANGES: Update CONTROL STREAM FILE to accept an
'offsetms' parameter This patch allows starting playback of audio
through the CONTROL STREAM FILE AGI command to start at a
particular offset. It will also return the final position of the
file in the 'endpos' attribute. (closes issue ASTERISK-17803)
Reported by: Murray Melvin patches: res_agi.c.r316293.diff
uploaded by murraytm (license 6221)
2013-07-31 15:43 +0000 [r395884] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip/pjsip_options.c: Found another missed "sip" ->
"pjsip" CLI command.
2013-07-31 15:27 +0000 [r395881] Kinsey Moore <kmoore@digium.com>
* tests/test_cel.c: Disable CEL tests that need rearchitecting to
operate properly
2013-07-31 14:45 +0000 [r395868] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_endpoint_identifier_constant.c (removed): Remove
"constant" endpoint identifier. This was created as a debugging
tool before proper endpoint identifiers were created. Using it
now can actually lead to harmful results.
2013-07-31 14:29 +0000 [r395866] Joshua Colp <jcolp@digium.com>
* bridges/bridge_native_rtp.c: Fix hold/unhold in
bridge_native_rtp, use tech_pvt instead of bridge_pvt, reduce
bridging attempts, and fix breaking native RTP bridges. (closes
issue ASTERISK-22128) (closes issue ASTERISK-22104)
2013-07-31 13:31 +0000 [r395837-395851] Kinsey Moore <kmoore@digium.com>
* channels/chan_pjsip.c, include/asterisk/res_pjsip.h,
include/asterisk/res_pjsip_pubsub.h,
include/asterisk/res_pjsip_exten_state.h,
include/asterisk/res_pjsip_session.h, configs/pjsip.conf.sample,
res/res_pjsip/include/res_pjsip_private.h: Fix remnants of the
pjsip renaming
* tests/test_cel.c: Enforce conference exit order for CEL tests
2013-07-30 22:41 +0000 [r395810-395824] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_endpoint_identifier_ip.c: Missed a conversion to
pjsip.conf in documentation and sorcery.
* main/abstract_jb.c: Remove ast_bridged_channel call from
abstract_jb.c Interestingly, this only happens in dead code.
2013-07-30 20:44 +0000 [r395793] David M. Lee <dlee@digium.com>
* res/res_pjsip: Setting svn:ignore for res/res_pjsip
2013-07-30 19:10 +0000 [r395748-395779] Mark Michelson <mmichelson@digium.com>
* res/res_pjsip_endpoint_identifier_constant.c: Update
res_pjsip_endpoint_identifier_constant.c to use reorganized
endpoint structure.
* res/res_sip_nat.c (removed),
res/res_pjsip_outbound_registration.c (added),
res/res_sip_session.c (removed),
res/res_pjsip_endpoint_identifier_anonymous.c (added),
res/res_sip_rfc3326.c (removed), res/res_pjsip_acl.c (added),
res/res_pjsip/pjsip_distributor.c (added),
res/res_sip_endpoint_identifier_constant.c (removed),
res/res_sip_mwi.c (removed), res/res_pjsip_diversion.c (added),
res/res_sip (removed), res/res_pjsip_dtmf_info.c (added),
res/res_sip_pubsub.c (removed),
include/asterisk/res_pjsip_exten_state.h (added),
res/res_pjsip_sdp_rtp.c (added), res/res_pjsip_messaging.c
(added), res/res_pjsip_registrar_expire.c (added),
res/res_pjsip_caller_id.c (added),
res/res_sip_authenticator_digest.c (removed),
res/res_sip_session.exports.in (removed),
res/res_pjsip_exten_state.c (added), res/res_sip_logger.c
(removed), res/res_sip.c (removed),
res/res_pjsip_pubsub.exports.in (added),
res/res_pjsip_endpoint_identifier_constant.c (added),
res/res_sip_outbound_registration.c (removed),
res/res_sip_endpoint_identifier_anonymous.c (removed),
res/res_pjsip_pubsub.c (added), res/res_pjsip/config_transport.c
(added), res/res_pjsip_transport_websocket.c (added),
res/res_pjsip_registrar.c (added), channels/chan_pjsip.c (added),
res/res_pjsip/pjsip_outbound_auth.c (added),
res/res_pjsip/config_global.c (added), res/res_sip_acl.c
(removed), res/res_sip_diversion.c (removed),
res/res_pjsip_authenticator_digest.c (added),
res/res_pjsip_session.exports.in (added), res/res_sip_dtmf_info.c
(removed), res/res_pjsip/config_domain_aliases.c (added),
include/asterisk/res_sip_session.h (removed), res/res_pjsip_t38.c
(added), res/res_sip_notify.c (removed), res/res_pjsip_logger.c
(added), res/res_pjsip/pjsip_options.c (added),
res/res_sip_endpoint_identifier_ip.c (removed),
res/res_sip_sdp_rtp.c (removed), res/res_sip_messaging.c
(removed), include/asterisk/res_pjsip_pubsub.h (added),
res/res_sip_caller_id.c (removed),
res/res_sip_endpoint_identifier_user.c (removed),
res/res_sip_pidf.c (removed),
res/res_pjsip_outbound_authenticator_digest.c (added),
res/res_sip_exten_state.c (removed),
res/res_pjsip_one_touch_record_info.c (added),
res/res_sip_pubsub.exports.in (removed), res/res_pjsip_refer.c
(added), include/asterisk/res_pjsip_session.h (added),
res/res_pjsip_notify.c (added), res/res_sip_transport_websocket.c
(removed), res/res_sip_registrar.c (removed),
res/res_pjsip_endpoint_identifier_ip.c (added),
include/asterisk/res_sip.h (removed),
res/res_pjsip/config_security.c (added), res/res_sip.exports.in
(removed), res/Makefile, res/res_sip_exten_state.exports.in
(removed), res/res_pjsip_endpoint_identifier_user.c (added),
res/res_pjsip/include (added), res/res_pjsip_pidf.c (added),
res/res_pjsip_nat.c (added), res/res_pjsip_session.c (added),
res/res_sip_t38.c (removed), channels/chan_gulp.c (removed),
res/res_pjsip/location.c (added), res/res_pjsip_rfc3326.c
(added), res/res_pjsip/config_system.c (added),
configs/pjsip.conf.sample (added),
include/asterisk/res_sip_pubsub.h (removed), res/res_pjsip_mwi.c
(added), res/res_pjsip/pjsip_configuration.c (added),
res/res_sip_outbound_authenticator_digest.c (removed),
res/res_pjsip (added), res/res_pjsip/include/res_pjsip_private.h
(added), res/res_sip_one_touch_record_info.c (removed),
include/asterisk/res_pjsip.h (added), res/res_pjsip/config_auth.c
(added), res/res_pjsip.exports.in (added),
configs/res_sip.conf.sample (removed), res/res_sip_refer.c
(removed), res/res_pjsip_exten_state.exports.in (added),
res/res_pjsip/security_events.c (added),
include/asterisk/res_sip_exten_state.h (removed),
res/res_pjsip/pjsip_global_headers.c (added), res/res_pjsip.c
(added), res/res_sip_registrar_expire.c (removed): The large
GULP->PJSIP renaming effort. The general gist is to have a clear
boundary between old SIP stuff and new SIP stuff by having the
word "SIP" for old stuff and "PJSIP" for new stuff. Here's a
brief rundown of the changes: * The word "Gulp" in dialstrings,
functions, and CLI commands is now "PJSIP" * chan_gulp.c is now
chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*"
are now "chan_pjsip_*" * All files that were "res_sip*" are now
"res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files
in the "res_pjsip" directory that began with "sip_*" are now
"pjsip_*" * The configuration file is now "pjsip.conf" instead of
"res_sip.conf" * The module info for all PJSIP-related files now
uses "PJSIP" instead of "SIP" * CLI and AMI commands created by
Asterisk's PJSIP modules now have "pjsip" as the starting word
instead of "sip"
* res/res_sip/sip_options.c,
res/res_sip_outbound_authenticator_digest.c,
res/res_sip_outbound_registration.c, res/res_sip_mwi.c,
res/res_sip_one_touch_record_info.c, res/res_sip_pubsub.c,
res/res_sip_diversion.c, res/res_sip/sip_configuration.c,
include/asterisk/res_sip.h, res/res_sip/sip_distributor.c,
res/res_sip.exports.in, res/res_sip_authenticator_digest.c,
res/res_sip/sip_outbound_auth.c, res/res_sip_sdp_rtp.c,
res/res_sip_messaging.c, res/res_sip_t38.c, channels/chan_gulp.c,
res/res_sip_caller_id.c, res/res_sip.c, res/res_sip_nat.c,
res/res_sip_session.c: Reorganize the ast_sip_endpoint structure
into substructures. (closes issue ASTERISK-22135) reported by
Matt Jordan Review: https://reviewboard.asterisk.org/r/2707
2013-07-30 14:16 +0000 [r395731] Joshua Colp <jcolp@digium.com>
* res/res_sip.c, res/res_sip/sip_configuration.c,
res/res_sip_session.c, include/asterisk/res_sip.h,
include/asterisk/res_sip_session.h,
res/res_sip_session.exports.in, channels/chan_gulp.c,
res/res_sip_t38.c (added): Add support for T.38 fax to
chan_pjsip. Review: https://reviewboard.asterisk.org/r/2692/
2013-07-30 13:46 +0000 [r395728] Kinsey Moore <kmoore@digium.com>
* res/res_pktccops.c: Fix compilation on gcc 4.8.1
2013-07-29 17:51 +0000 [r395686] David M. Lee <dlee@digium.com>
* res/parking/parking_devicestate.c, include/asterisk/mixmonitor.h,
main/mixmonitor.c: Removed quotes from svn:keywords props on a
few files. Subversion doesn't do quote processing, so it actually
thinks that the closing quote in 'Revision"' is a part of the
keyword.
2013-07-29 16:16 +0000 [r395674] Mark Michelson <mmichelson@digium.com>
* res/res_sip.c: Clarify documentation for trust of identification.
(closes issue ASTERISK-22023) Reported by Rusty Newton
2013-07-29 15:58 +0000 [r395672-395673] Matthew Jordan <mjordan@digium.com>
* main/loader.c: Put the include in there Mea culpa...
* main/loader.c: When performing a reload, reload the new
features_config and not the old Performing a module reload of
core components causes specific functions compiled into the
Asterisk binary to be reloaded. The table of said functions was
still pointing to the old features reload mechanism, and not the
new one.
2013-07-29 14:51 +0000 [r395653] Kinsey Moore <kmoore@digium.com>
* tests/test_cel.c: Clean up and improve test_cel Improve
reliability of attended transfer merge and link tests. Stop using
ast_log(LOG_ERROR, ...); in favor of ast_test_status_update
Remove fred and eve channel helpers since they are not necessary
2013-07-29 14:08 +0000 [r395636] David M. Lee <dlee@digium.com>
* res/ari: Set svn:ignore in res/ari directory
2013-07-29 12:10 +0000 [r395619] Kinsey Moore <kmoore@digium.com>
* res/res_sip.c: Remove comment that no longer applies The monitor
thread is already properly torn down on unload and load failure.
2013-07-27 23:11 +0000 [r395588-395603] Kinsey Moore <kmoore@digium.com>
* tests/test_ari_model.c, res/ari.make (added),
res/ari/resource_bridges.h (added), res/ari/resource_asterisk.c
(added), res/res_ari_endpoints.c (added),
res/res_stasis_http_sounds.c (removed),
res/ari/resource_asterisk.h (added), res/res_stasis_http.c
(removed), rest-api-templates/stasis_http_resource.h.mustache
(removed), res/res_ari.c (added),
rest-api-templates/make_ari_stubs.py,
rest-api-templates/ari_resource.h.mustache (added),
res/res_ari_asterisk.c (added), res/Makefile, res/ari/internal.h
(added), res/res_ari_model.c, res/res_stasis_http.exports.in
(removed), res/ari/resource_playback.c (added),
tests/test_stasis_http.c (removed), res/ari/resource_playback.h
(added), res/ari/resource_channels.c (added),
res/ari/ari_websockets.c (added), res/ari/resource_recordings.c
(added), res/ari/resource_channels.h (added), tests/test_ari.c
(added), res/ari/resource_endpoints.c (added),
res/ari/resource_events.c (added), res/ari/resource_recordings.h
(added), include/asterisk/stasis_http.h (removed),
res/res_ari_playback.c (added), res/ari/resource_endpoints.h
(added), res/ari/resource_events.h (added),
res/ari/resource_sounds.c (added), configs/ari.conf.sample,
include/asterisk/ari.h (added), res/res_ari_channels.c (added),
rest-api-templates/stasis_http.make.mustache (removed),
res/stasis_http.make (removed), res/ari/resource_sounds.h
(added), res/res_ari_recordings.c (added),
rest-api-templates/ari.make.mustache (added),
res/res_ari_events.c (added), res/res_statsd.c,
res/res_stasis_http_bridges.c (removed), res/res_ari_sounds.c
(added), rest-api-templates/ari_model_validators.c.mustache,
res/res_ari_bridges.c (added), res/res_stasis_http_asterisk.c
(removed), res/stasis_http (removed),
rest-api-templates/res_stasis_http_resource.c.mustache (removed),
main/stasis_config.c, rest-api-templates/rest_handler.mustache,
res/ari (added), rest-api-templates/res_ari_resource.c.mustache
(added), res/ari/ari_model_validators.c (added),
res/ari/ari_model_validators.h (added), res/res_ari.exports.in
(added), rest-api-templates/stasis_http_resource.c.mustache
(removed), res/ari/config.c (added),
rest-api-templates/ari_resource.c.mustache (added), res/ari/cli.c
(added), res/res_stasis_http_playback.c (removed),
rest-api-templates/ari_model_validators.h.mustache,
res/res_stasis_http_channels.c (removed),
res/res_ari_model.exports.in, res/res_stasis_http_recordings.c
(removed), res/res_stasis_http_endpoints.c (removed),
res/ari/resource_bridges.c (added), res/res_stasis_http_events.c
(removed): Rename everything Stasis-HTTP to ARI This renames all
files and API calls from several variants of Stasis-HTTP to ARI
including: * Stasis-HTTP -> ARI * STASIS_HTTP -> ARI *
stasis_http -> ari (ast_ari for global symbols, file names as
well) * stasis http -> ARI Review:
https://reviewboard.asterisk.org/r/2706/ (closes issue
ASTERISK-22136)
* tests/test_cel.c: Improve reliability of bridge merge CEL test
2013-07-26 21:34 +0000 [r395559-395574] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_builtin_features.c, main/parking.c, main/bridge.c,
main/bridge_basic.c, main/features.c,
bridges/bridge_builtin_interval_features.c,
apps/app_bridgewait.c, apps/app_confbridge.c,
include/asterisk/bridge_features.h, include/asterisk/parking.h,
main/bridge_channel.c, res/parking/parking_bridge_features.c,
apps/app_agent_pool.c, apps/confbridge/conf_config_parser.c:
Remove the unsafe bridge parameter from
ast_bridge_hook_callback's. Most hook callbacks did not need the
bridge parameter. The pointer value could become invalid if the
channel is moved to another bridge while it is executing. * Fixed
some issues in feature_attended_transfer() as a result. * Reduce
the bridge inhibit count in
attended_transfer_properties_shutdown() after it has restored the
bridge channel hooks. * Removed basic bridge requirement on
feature_blind_transfer(). It does not require the basic bridge
like feature_attended_transfer().
* include/asterisk/bridge_features.h,
res/parking/parking_bridge_features.c, main/bridge.c,
bridges/bridge_builtin_interval_features.c,
apps/app_bridgewait.c: Improved feature limits interval hook
implementaion. * Fixed feature limits to not use special members
of struct ast_bridge_features. * Fixed memory leak in off nominal
paths of bridge_builtin_set_limits(). * Fixed off nominal path in
ast_bridge_features_limits_construct() freeing unallocated memory
if it was not called by bridge_builtin_set_limits(). * Made
bridge_builtin_interval_features.so unloadable. * Simplified
parking's use of its duration interval hook. * Made BridgeWait S
option not depend upon another module being loaded. (closes issue
ASTERISK-22107) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2701/
2013-07-26 17:42 +0000 [r395527] David M. Lee <dlee@digium.com>
* res/stasis_http/resource_events.c, res/stasis/app.c: Fix
/stasis/res/app_replaced unit test. A typo in recent changes
caused the JSON ApplicationReplaced message to fail to build, so
the message wasn't being sent out the WebSocket. Related, the
replaced application would also unregister itself when it
disconnected, which would actually unregister the new
application. This was also fixed.
2013-07-26 16:34 +0000 [r395509] Jonathan Rose <jrose@digium.com>
* main/bridge_channel.c, include/asterisk/bridge.h,
include/asterisk/bridge_channel_internal.h, main/bridge.c,
apps/app_bridgewait.c: Add name argument to BridgeWait() so
multiple holding bridges may be used Changes arguments for
BridgeWait from BridgeWait(role, options) to
BridgeWait(bridge_name, role, options). Now multiple holding
bridges may be created and referenced by this application.
(closes issue ASTERISK-21922) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2642/
2013-07-26 00:03 +0000 [r395466-395477] Richard Mudgett <rmudgett@digium.com>
* apps/app_bridgewait.c: Remove some unnecessary parentheses.
* bridges/bridge_builtin_interval_features.c: Revision
2013-07-25 20:54 +0000 [r395439-395455] Joshua Colp <jcolp@digium.com>
* res/res_sip_session.c: Fix crash due to trying to send a
re-invite while in the incorrect state. This crash would occur if
a re-invite was queued while the initial INVITE transaction was
still occurring and the response to the INVITE was not ACKed.
This lack of ACK would cause the INVITE session state to never
reach confirmed. Once the transaction terminated, however, the
queued re-invite would occur and cause a crash due to this lack
of state change. This fix checks the INVITE session state before
performing the re-invite to ensure it is in the required
confirmed state.
* res/res_sip.c, res/res_sip/sip_configuration.c: Change the
default value for "allowsubscribe" to yes to match chan_sip.
2013-07-25 18:27 +0000 [r395430] Richard Mudgett <rmudgett@digium.com>
* main/stasis_bridges.c, include/asterisk/bridge_after.h,
include/asterisk/bridge_channel_internal.h,
main/manager_bridges.c, include/asterisk/bridge_channel.h,
main/bridge_after.c, include/asterisk/bridge_technology.h,
include/asterisk/bridge_internal.h,
include/asterisk/bridge_features.h, main/bridge_channel.c,
include/asterisk/bridge.h, include/asterisk/bridge_basic.h,
include/asterisk/bridge_roles.h, main/bridge.c,
main/bridge_basic.c, include/asterisk/stasis_bridges.h,
main/bridge_roles.c: Restore bridging files history.
2013-07-25 15:29 +0000 [r395367-395410] Matthew Jordan <mjordan@digium.com>
* main/features.c, include/asterisk/features.h: Remove some dead
parking call Since nothing is using these global parking
functions, remove them! The first of many.
* main/features.c: Remove dead bridging code from features This
removes the previously #if 0'd code. The functionality removed
has either been subsumed by the Bridging API or is no longer
applicable.
* main/cli.c, main/cdr.c, main/manager_bridges.c, main/manager.c,
res/stasis_http/resource_bridges.c, tests/test_cel.c,
res/res_stasis.c, main/stasis_bridges.c, tests/test_cdr.c: Fix
incorrect reference to stasis/bridging.h
* include/asterisk/stasis_bridges.h (added),
include/asterisk/bridging_after.h (removed),
bridges/bridge_simple.c, main/core_local.c,
res/parking/parking_bridge_features.c,
res/parking/parking_bridge.c, main/cli.c, main/manager_bridges.c
(added), include/asterisk/bridging_technology.h (removed),
apps/confbridge/include/confbridge.h, channels/chan_skinny.c,
include/asterisk/bridging_features.h (removed),
main/bridge_after.c (added), main/stasis_channels.c,
include/asterisk/bridge_features.h (added), main/bridge_channel.c
(added), res/parking/parking_manager.c, channels/chan_mgcp.c,
channels/chan_unistim.c, include/asterisk/bridge_roles.h (added),
channels/chan_bridge_media.c, main/bridge.c (added),
res/parking/parking_controller.c, apps/app_bridgewait.c,
res/stasis_http/resource_bridges.c,
res/parking/parking_applications.c,
include/asterisk/bridging_channel_internal.h (removed),
main/cel.c, apps/app_queue.c, include/asterisk/stasis_bridging.h
(removed), main/stasis_bridges.c (added), main/bridging_after.c
(removed), res/res_stasis_bridge_add.c,
include/asterisk/bridge_channel_internal.h (added),
channels/chan_dahdi.c, channels/sig_analog.c,
include/asterisk/bridging_internal.h (removed),
apps/confbridge/confbridge_manager.c, main/manager_bridging.c
(removed), tests/test_cel.c, include/asterisk/bridge_internal.h
(added), include/asterisk/bridging_roles.h (removed),
apps/confbridge/conf_chan_announce.c,
include/asterisk/bridge_basic.h (added),
include/asterisk/core_unreal.h, main/parking.c,
res/stasis/control.c, bridges/bridge_holding.c,
channels/chan_sip.c, bridges/bridge_softmix.c,
main/bridge_roles.c (added), channels/chan_iax2.c,
apps/app_agent_pool.c, include/asterisk/bridging_channel.h
(removed), apps/confbridge/conf_config_parser.c,
include/asterisk/features.h, main/channel.c,
res/parking/res_parking.h, main/manager.c, channels/chan_misdn.c,
main/stasis_bridging.c (removed), include/asterisk/bridging.h
(removed), bridges/bridge_builtin_interval_features.c,
include/asterisk/bridging_basic.h (removed),
include/asterisk/bridge_technology.h (added),
bridges/bridge_native_rtp.c, tests/test_cdr.c,
include/asterisk/doxygen/architecture.h, main/bridging_roles.c
(removed), res/res_sip_refer.c, main/bridge_basic.c (added),
apps/confbridge/conf_chan_record.c, main/core_unreal.c,
channels/sig_pri.c, include/asterisk/bridge_after.h (added),
bridges/bridge_builtin_features.c,
channels/dahdi/bridge_native_dahdi.c,
res/stasis_http/resource_channels.c,
include/asterisk/bridge_channel.h (added), funcs/func_channel.c,
main/bridging_channel.c (removed), apps/app_dumpchan.c,
main/features.c, apps/app_confbridge.c, include/asterisk/bridge.h
(added), main/bridging.c (removed), main/bridging_basic.c
(removed), apps/app_dial.c: A great big renaming patch This patch
renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk
naming conventions: * channel is not "channeling" * monitor is
not "monitoring" etc. A bridge is an object. It is a first class
citizen in Asterisk. "Bridging" is the act of using a bridge on a
set of channels - and the API that fulfills that role is more
than just the action. (closes issue ASTERISK-22130)
* include/asterisk/bridging_features.h, funcs/func_channel.c,
main/bridging_channel.c, main/features.c,
include/asterisk/bridging.h,
bridges/bridge_builtin_interval_features.c, main/bridging.c,
main/bridging_basic.c, apps/app_dial.c,
include/asterisk/bridging_after.h (added),
bridges/bridge_softmix.c,
include/asterisk/bridging_channel_internal.h, apps/app_queue.c,
res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
include/asterisk/bridging_channel.h, main/bridging_after.c
(added), include/asterisk/bridging_technology.h,
include/asterisk/bridging_internal.h,
bridges/bridge_builtin_features.c: Move after bridge callbacks
into their own file One more major refactoring to go.
2013-07-25 00:44 +0000 [r395351] Joshua Colp <jcolp@digium.com>
* res/res_sip/sip_distributor.c, channels/chan_gulp.c,
res/res_sip_session.c: Improve initial INVITE handling and fix
crash due to rapidly arriving CANCEL. (closes issue
ASTERISK-22150) Review: https://reviewboard.asterisk.org/r/2696/
2013-07-24 23:40 +0000 [r395316-395340] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c, include/asterisk/bridging_features.h,
main/bridging_channel.c,
include/asterisk/bridging_channel_internal.h: Simplify interval
hooks since there is only one bridge threading model now. *
Convert interval timers to use the ast_waitfor_nandfds() timeout.
* Remove bridge channel action for intervals. Now the main loop
handles running interval hooks.
* main/bridging.c, include/asterisk/bridging_features.h,
main/bridging_channel.c, apps/app_confbridge.c: Refactor
ast_bridge_features struct. * Reduced the number of hook
containers to just dtmf_hooks, interval_hooks, and other_hooks.
As a result, several functions dealing with the different hook
containers could be combined. * Extended the generic hook struct
for DTMF and interval hooks instead of using a variant record. *
Merged the special talk detector hook into the other_hooks
container. * Replaced ast_bridge_features_set_talk_detector()
with ast_bridge_talk_detector_hook(). (issue ASTERISK-22107)
* main/features.c: * Refactor setup_bridge_features_builtin(). *
Add an error message so you know when a feature is not available
and you tried to use it. It usually means the module has not been
loaded.
2013-07-24 19:32 +0000 [r395295-395298] Matthew Jordan <mjordan@digium.com>
* main/asterisk.exports.in: Export exports.in as well Because is is
rather needed.
* main/bridging.c, res/parking/parking_bridge_features.c,
apps/app_agent_pool.c, include/asterisk/bridging_channel.h,
main/bridging_basic.c, bridges/bridge_builtin_features.c,
include/asterisk/bridging_features.h, main/bridging_channel.c,
bridges/bridge_builtin_interval_features.c,
include/asterisk/bridging_channel_internal.h: Update
bridge_channel refactorings; export bridge_ symbol
2013-07-24 18:51 +0000 [r395283] Jason Parker <jparker@digium.com>
* contrib/scripts/install_prereq: Add pjproject to install_prereq.
Also fixes spacing, in passing. (closes issue ASTERISK-22131)
2013-07-24 18:08 +0000 [r395267-395271] Kinsey Moore <kmoore@digium.com>
* res/res_sip.c: Tweak another magic number
* main/manager_bridging.c: Make AMI BridgeInfo action more verbose
Ensure that the BridgeInfo command provides adequate state
information about channels by publishing the full channel
snapshot for BridgeInfoChannel subevents. This prevents a
two-stage lookup since most consumers will be keying on channel
names instead of uniqueids. (closes issue ASTERISK-22140)
* res/res_sip/sip_global_headers.c: Tweak a magic number (closes
issue ASTERISK-22146)
2013-07-24 16:01 +0000 [r395254-395255] Richard Mudgett <rmudgett@digium.com>
* main/bridging_channel.c,
include/asterisk/bridging_channel_internal.h,
include/asterisk/bridging_channel.h, main/channel.c: Add missing
end-of-file line terminators.
* bridges/bridge_native_rtp.c: Add missing line terminator to debug
message.
2013-07-24 15:38 +0000 [r395253] Matthew Jordan <mjordan@digium.com>
* include/asterisk/bridging_channel_internal.h (added),
res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
include/asterisk/bridging_channel.h (added),
res/parking/parking_bridge.c, include/asterisk/features.h,
main/channel.c, include/asterisk/bridging_technology.h,
include/asterisk/bridging_internal.h,
bridges/bridge_builtin_features.c, main/bridging_channel.c
(added), main/features.c, include/asterisk/bridging.h,
bridges/bridge_builtin_interval_features.c, main/bridging.c,
main/bridging_basic.c, include/asterisk/channel.h: Perform the
initial renaming of the Bridging API This patch does the
following: * It pulls out bridge_channel and puts it into its own
translation unit * It adds public and protected headers for
bridging_channel. Protected functions are appropriate only for
the Bridging API and sub-classes of a bridge. (issue
ASTERISK-22130)
2013-07-24 14:35 +0000 [r395243] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: Let the compiler do more type checking with
bridge hook callbacks.
2013-07-23 22:32 +0000 [r395227] Joshua Colp <jcolp@digium.com>
* bridges/bridge_native_rtp.c: Fix a check in bridge_native_rtp
which determined if attaching the framehook failed or not.
2013-07-23 21:32 +0000 [r395215] Jonathan Rose <jrose@digium.com>
* funcs/func_channel.c, include/asterisk/bridging_basic.h,
main/bridging_basic.c: func_channel: dtmf_features setting Allows
reading andsetting dtmf features via a channel function
CHANNEL(dtmf_features) (closes issue ASTERISK-21876) Reported by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/2648/
2013-07-23 21:14 +0000 [r395203-395205] Joshua Colp <jcolp@digium.com>
* bridges/bridge_native_rtp.c: Add some debug messages to make it
clear what RTP bridging functionality is in use.
* bridges/bridge_native_rtp.c: Fix some logic so native RTP bridge
will occur when monitor, audiohooks, or framehooks are not
present.
2013-07-23 19:14 +0000 [r395188] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_softmix.c, main/bridging.c,
include/asterisk/bridging.h: Pull softmix bridge parameters into
a sub structure.
2013-07-23 18:41 +0000 [r395183] Joshua Colp <jcolp@digium.com>
* channels/chan_gulp.c: Drop the reference count on the correct
object.
2013-07-23 18:41 +0000 [r395154-395182] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, main/utils.c: Reinclude sys/stat.h in
chan_dahdi.c and remove redundant include in utils.c
* channels/chan_dahdi.h, channels/chan_mgcp.c,
channels/chan_dahdi.c, channels/dahdi/bridge_native_dahdi.c: Some
chan_dahdi protected function renaming. analog_lib_handles -->
dahdi_analog_lib_handles enable_dtmf_detect -->
dahdi_dtmf_detect_enable disable_dtmf_detect -->
dahdi_dtmf_detect_disable dahdi_enable_ec --> dahdi_ec_enable
dahdi_disable_ec --> dahdi_ec_disable update_conf -->
dahdi_conf_update dahdi_link --> dahdi_master_slave_link
dahdi_unlink --> dahdi_master_slave_unlink (closes issue
ASTERISK-22129) Reported by: rmudgett
* channels/chan_dahdi.c, channels/dahdi/bridge_native_dahdi.c,
channels/chan_dahdi.h (added), channels/dahdi (added),
channels/dahdi/bridge_native_dahdi.h, bridges/bridge_softmix.c,
channels/Makefile, main/bridging.c: Restore chan_dahdi native
bridging and PRI tromboned call elimination. Created a
native_dahdi bridging technology for use with the new bridging
API. The new bridging technology is part of the chan_dahdi
channel driver because it is very specific to that driver. Rather
than include the new code directly into chan_dahdi.c the new
bridge technology is in its own file and linked into
chan_dahdi.so. A large part of this change is the mechanical
process of moving declarations around so chan_dahdi.c can be
split up into more files later. * Changed the bridging core to
pass NULL frames into the channel technologies instead of
discarding them. The channel technologies may need the proding to
determine if their configuration is still valid. (closes issue
ASTERISK-21886) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2681/
2013-07-23 15:28 +0000 [r395151] Mark Michelson <mmichelson@digium.com>
* main/bridging_roles.c, include/asterisk/bridging_internal.h
(added), bridges/bridge_builtin_features.c,
main/stasis_bridging.c, include/asterisk/bridging_features.h,
include/asterisk/features_config.h, include/asterisk/bridging.h,
main/features.c, include/asterisk/bridging_roles.h, main/cel.c,
main/features_config.c, include/asterisk/stasis_bridging.h,
main/bridging.c, main/bridging_basic.c: Make DTMF attended
transfer support feature-complete. This greatly modifies the
operation of DTMF attended transfers so that the full range of
options from features.conf applies. In addition, a new option has
been added that allows for a transferer to switch between bridges
during a transfer before completing the transfer. (closes issue
ASTERISK-21543) reported by Matt Jordan Review:
https://reviewboard.asterisk.org/r/2654
2013-07-23 14:57 +0000 [r395136] David M. Lee <dlee@digium.com>
* res/res_stasis_http_channels.c, res/res_stasis_http_sounds.c,
res/res_stasis_http_bridges.c, res/res_stasis_http_recordings.c,
res/res_stasis_http.c, res/res_stasis_http_endpoints.c,
res/res_stasis_http_asterisk.c, res/res_stasis_http_playback.c,
rest-api-templates/res_stasis_http_resource.c.mustache: No more
teapots. Now that the ARI implementation is nearing some
definition of completeness, we should properly respond with 501's
for unimplemented functionality, instead of the almost humorous
418.
2013-07-23 14:49 +0000 [r395135] Matthew Jordan <mjordan@digium.com>
* main/channel.c: Kill the zombies In previous versions of
Asterisk, the zombies roamed freely, unchecked and uncontrolled.
They ravaged Asterisk systems with their biting and their nashing
and their pointy teeth. Sometimes, you couldn't even hang them
up. Now, zombies are rare. They still *technically* exist in
certain places, but they are controlled. Kind of like a zombie
zoo: you can see them, but you can't touch them, and they can't
touch you. Bring your kids! Because zombies are now population
controlled with a very short lifespan, there's no reason to
rename the channels to '%s<ZOMBIE>'. The channels are guaranteed
to die off quickly; the rename really is just confusing at this
point. This patch finally removes the renaming. On the plus side:
this made my life easier in CDRs during call pickup and attended
transfers to an Asterisk application. It will make other folks
lives easier as well! Review:
https://reviewboard.astierks.org/r/2690/ (closes issue
ASTERISK-21699) Reported by: Matt Jordan
2013-07-23 13:52 +0000 [r395121] Kinsey Moore <kmoore@digium.com>
* res/res_sip_sdp_rtp.c, channels/chan_gulp.c, res/res_sip.c,
channels/chan_sip.c, res/res_sip/sip_configuration.c,
res/res_sip_session.c, include/asterisk/res_sip.h,
include/asterisk/res_sip_session.h: Add DTLS-SRTP support to
chan_pjsip This patch introduces DTLS-SRTP support to chan_pjsip
and the options necessary to configure it including an option to
allow choosing between 32 and 80 byte SRTP tag lengths. During
the implementation and testing of this patch, three other bugs
were found and their fixes are included with this patch. The two
in chan_sip were a segfault relating to DTLS setup and mistaken
call rejection. The third bug fix prevents chan_pjsip from
attempting to perform bridge optimization between two endpoints
if either of them is running any form of SRTP. Review:
https://reviewboard.asterisk.org/r/2683/ (closes issue
ASTERISK-21419)
2013-07-23 13:42 +0000 [r395118-395120] David M. Lee <dlee@digium.com>
* res/stasis/app.h, res/res_stasis.c, res/stasis/app.c: Continue
events when ARI WebSocket reconnects This patch addresses a bug
in the /ari/events WebSocket in handling reconnects. When a
Stasis application's associated WebSocket was disconnected and
reconnected, it would not receive events for any channels or
bridges it was subscribed to. The fix was to lazily clean up
Stasis application registrations, instead of removing them as
soon as the WebSocket goes away. When an application is
unregistered at the WebSocket level, the underlying application
is simply deactivated. If the application WebSocket is
reconnected, the application is reactivated for the new
connection. To avoid memory leaks from lingering, unused
application, the application list is cleaned up whenever new
applications are registered/unregistered. (closes issue
ASTERISK-21970) Review: https://reviewboard.asterisk.org/r/2678/
* main/manager_bridging.c,
include/asterisk/stasis_message_router.h, tests/test_stasis.c,
main/manager_channels.c, main/cdr.c,
main/stasis_message_router.c: Fix bridge/channel AMI event
ordering issues The stasis_cache_update messages are somewhat
cumbersome to handle with the stasis_message_router. Since all
updates have the same message type, they are normally handled
with the same route. Since caching itself is a first class
component of stasis-core, it makes sense for the router to handle
the cache update messages itself. This patch adds
stasis_message_router_add_cache_update() and
stasis_message_router_remove_cache_update() to handle the routing
of stasis_cache_update messages. This patch also corrects an
issue with manager_{bridging,channels}.c, where events might be
reordered. The reordering occurs because the components use
different message routers, which they needed because they both
needed to route cache update messages. They now both use
manager's router, and add cache routes for just the cache updates
they are interested in. (closes issue ASTERISK-22038) Review:
https://reviewboard.asterisk.org/r/2677/
2013-07-23 12:56 +0000 [r395107] Kinsey Moore <kmoore@digium.com>
* res/res_sip/sip_options.c: Add missing newline
2013-07-23 12:27 +0000 [r395102] Joshua Colp <jcolp@digium.com>
* channels/chan_gulp.c, res/res_sip_session.c,
include/asterisk/res_sip_session.h,
res/res_sip_session.exports.in: Expose the chan_pjsip
implementation pvt and session in a defined manner. This allows
modules outside of chan_pjsip itself to get the session given
only an Asterisk channel. Review:
https://reviewboard.asterisk.org/r/2674/
2013-07-23 00:16 +0000 [r395089] Matthew Jordan <mjordan@digium.com>
* main/cdr.c: Fix unbalanced lock when serializing CDR variables
I'm only surprised that this didn't cause larger problems.
2013-07-23 00:02 +0000 [r395088] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: Remove some BUGBUG notes that have been handled.
2013-07-22 20:42 +0000 [r395074] Kinsey Moore <kmoore@digium.com>
* tests/test_cel.c: Make the CEL blind transfer test pass
consistently
2013-07-22 13:52 +0000 [r394881-395034] Matthew Jordan <mjordan@digium.com>
* /, main/asterisk.c: Update copyright year to 2013 in asterisk.c;
some whitespace fixes (closes issue ASTERISK-22179) Reported by:
Malcolm Davenport ........ Merged revisions 395032 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 395033 from
http://svn.asterisk.org/svn/asterisk/branches/11
* funcs/func_channel.c, /: Clean up documentation This patch cleans
up documentation in func_channel for the following items: *
rtpsource * secure_signaling * secure_media * various OOH323
parameters (closes issue ASTERISK-20969) Reported by: snuffy
patches: func_chan-update.diff uploaded by snuffy (License 5024)
........ Merged revisions 394980 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394981 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, configs/indications.conf.sample: Provide proper ring tone in
indications.conf for Malaysia The ring tone provided in the
sample indications.conf was incorrect. This patch modifies the
sample ring tone to be what it should: ring =
425/400,0/200,425/400,0/2000 This brings it in line with the tone
definition in DAHDI 2.7.0. (zonedata.c) (closes issue
ASTERISK-21997) Reported by: Filip Jenicek patches:
malaysia_ring.patch uploaded by phill (License 6277) ........
Merged revisions 394940 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394941 from
http://svn.asterisk.org/svn/asterisk/branches/11
* contrib/scripts/safe_asterisk, Makefile,
configs/safe_asterisk.conf.sample (added), CHANGES: Always
install safe_asterisk; add configuration file support This patch
modifies the behavior of safe_asterisk in two ways: (1) It
modifies the Asterisk Makefile such that safe_asterisk is always
installed on a 'make install'. This was done as bugfixes in the
safe_asterisk script were not applied in previous version of
Asterisk without first removing the old version of the script.
(2) In order to keep a newly installed version of safe_asterisk
from impacting local modifications, a new config file -
safe_asterisk.conf.sample - has been provided. Settings that were
previously modified in safe_asterisk can be set there instead.
(closes issue ASTERISK-21965) Reported by: Jeremy Kister patches:
safe_asterisk.patch uploaded by jkister (License 6232)
* /, main/http.c: Tolerate presence of RFC2965 Cookie2 header by
ignoring it This patch modifies parsing of cookies in Asterisk's
http server by doing an explicit comparison of the "Cookie"
header instead of looking at the first 6 characters to determine
if the header is a cookie header. This avoids parsing "Cookie2"
headers and overwriting the previously parsed "Cookie" header.
Note that we probably should be appending the cookies in each
"Cookie" header to the parsed results; however, while clients can
send multiple cookie headers they never really do. While this
patch doesn't improve Asterisk's behavior in that regard, it
shouldn't make it any worse either. Note that the solution in
this patch was pointed out on the issue by the issue reporter,
Stuart Henderson. (closes issue ASTERISK-21789) Reported by:
Stuart Henderson Tested by: mjordan, Stuart Henderson ........
Merged revisions 394899 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394900 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, contrib/realtime/postgresql/realtime.sql: Update PostgreSQL
realtime scripts with schema for queue_log table This patch
updates the realtime SQL scripts with an entry that will create
the queue_log table. This brings the PostgreSQL scripts inline
with the MySQL scripts, with respect to what tables they will
create. (closes issue ASTERISK-21021) Reported by: Eugene
patches: queue_log.sql uploaded by varnav (license 6360) ........
Merged revisions 394896 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394897 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/iax2/parser.c: Add additional control frame types to the
IAX2 parser for debug messages This patch adds some of the more
recent control frame types to the IAX2 parser. When IAX2
debugging is enabled, it will now show more of the control frame
types. (closes issue ASTERISK-22120) Reported by: Birger "WIMPy"
Harzenetter patches: iaxcmds.diff uploaded by wimpy
* /, configs/iax.conf.sample: Document connectedline parameter for
chan_iax2 The connectedline parameter for a chan_iax2 peer was
undocumented. This patch documents the options in the sample
configuration file. (closes issue ASTERISK-21953) Reported by:
Birger "WIMPy" Harzenetter ........ Merged revisions 394886 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394890 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/manager.c, CHANGES: Allow setting allowmultiplelogin on an
account basis This patch modifies manager to allow the
allowmultiplelogin setting to be set on an account by account
basis. When set in the general context, it will act as the
default for the defined accounts. Setting it in the account will
override the general setting. (closes issue ASTERISK-21324)
Reported by: vldmr patches:
asterisk-manager-per-user-allowmultiplelogin.patch uploaded by
vldmr (License 6487)
2013-07-20 13:25 +0000 [r394858-394870] Kinsey Moore <kmoore@digium.com>
* include/asterisk/cel.h, tests/test_cel.c, CHANGES, main/cel.c,
main/asterisk.c: Add CEL local optimization record type This adds
a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent local
channel optimizations. Local channel optimizations were one of
several things conveyed by the now defunct BRIDGE_UPDATE event
type. This also adds a unit test to test generation of this new
CEL event. Review: https://reviewboard.asterisk.org/r/2676/
* tests/test_cel.c, CHANGES, apps/app_queue.c, main/cel.c,
apps/app_dial.c, main/channel.c, channels/chan_dahdi.c,
main/pbx.c, channels/sig_analog.c, channels/chan_sip.c,
include/asterisk/cel.h, apps/app_celgenuserevent.c,
apps/app_directed_pickup.c, main/features.c: Add transfer support
to CEL This adds CEL support for blind and attended transfers and
call pickup. During the course of adding this functionality I
noticed that CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are
particularly useless without a bridge identifier, so I added that
as well. This adds tests for blind transfers, several types of
attended transfers, and call pickup. The extra field in CEL
records now consists of a JSON blob whose fields are defined on a
per-event basis. Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
2013-07-20 01:11 +0000 [r394825-394846] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h: Regroup the ao2 search_flags. Moved
the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags together into
a field and renamed them to OBJ_SEARCH_OBJECT, OBJ_SEARCH_KEY,
and OBJ_SEARCH_PARTIAL_KEY respectively. The values were selected
to keep existing code compiling and working until the codebase
can be changed to stop using these values as bit flags and use
them as an enum field. The old names are defined to the new names
for backward compatibility.
* main/audiohook.c, main/channel.c, include/asterisk/audiohook.h:
Minor optimizations. * Made ast_audiohook_detach_list() and
ast_audiohook_write_list_empty() NULL tolerant. * Made
ast_audiohook_detach_list() return void since it is a destructor.
* main/bridging.c, main/channel.c, include/asterisk/channel.h,
bridges/bridge_native_rtp.c: Extract a repeated test into
ast_channel_has_audio_frame_or_monitor().
2013-07-19 19:40 +0000 [r394809-394810] Jonathan Rose <jrose@digium.com>
* res/stasis/control.c, res/stasis_http/resource_channels.c,
res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
res/stasis_http/resource_channels.h,
rest-api/api-docs/channels.json: ARI: MOH start and stop for a
channel (issue ASTERISK-21974) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2680/
* rest-api/api-docs/channels.json, res/res_stasis_http_bridges.c,
res/res_stasis.c, rest-api/api-docs/recordings.json,
include/asterisk/core_unreal.h, res/res_stasis_http_playback.c,
res/res_stasis_playback.c, channels/chan_bridge_media.c (added),
res/stasis/control.c, res/stasis_http/ari_model_validators.c,
res/res_stasis_http_channels.c, main/core_unreal.c,
include/asterisk/stasis_app.h,
res/stasis_http/resource_bridges.c,
res/stasis_http/ari_model_validators.h,
res/stasis_http/resource_bridges.h,
include/asterisk/stasis_app_playback.h,
rest-api/api-docs/bridges.json, include/asterisk/logger.h,
res/stasis_http/resource_channels.c,
rest-api/api-docs/playback.json: ARI: Bridge Playback, Bridge
Record Adds a new channel driver for creating channels for
specific purposes in bridges, primarily to act as either
recorders or announcers. Adds ARI commands for playing
announcements to ever participant in a bridge as well as for
recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as
playback controls. (closes issue ASTERISK-21592) Reported by:
Matt Jordan (closes issue ASTERISK-21593) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/2670/
2013-07-19 19:23 +0000 [r394795-394808] Kinsey Moore <kmoore@digium.com>
* include/asterisk/channel.h, main/stasis_channels.c, main/cel.c,
apps/confbridge/conf_chan_announce.c, main/manager_channels.c,
res/parking/parking_manager.c, main/cdr.c,
include/asterisk/stasis_channels.h,
apps/confbridge/conf_chan_record.c,
apps/confbridge/confbridge_manager.c, main/manager_bridging.c:
Filter channels used as internal mechanisms This adds new flags
to the channel tech properties that flag it as different types of
implementation detail used exclusively to provide a feature.
Examples of channels that would have these flags include the
announcement and recording channels used by confbridge which are
the only two marked as such by this patch. Review:
https://reviewboard.asterisk.org/r/2633/ (closes issue
ASTERISK-21873)
* channels/chan_sip.c: Fix crash when using temporary peers
Temporary peers do not have an associated Stasis endpoint and
quite a bit of code in chan_sip assumes that all peers have a
Stasis endpoint. All endpoint accesses in chan_sip are now
wrapped in an endpoint NULL-check.
2013-07-19 18:00 +0000 [r394793] Jason Parker <jparker@digium.com>
* main/stasis_system.c, main/ccss.c,
include/asterisk/stasis_system.h: Convert CCSS manager events to
stasis. (closes issue ASTERISK-21473) Review:
https://reviewboard.asterisk.org/r/2682/
2013-07-19 17:55 +0000 [r394776-394791] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: Made audiohooks, framehooks, and monitor prevent
local channel optimization. Audiohooks, framehooks, and monitor
represent state on a local channel that will go away if it is
optimized out. (closes issue ASTERISK-21954) Reported by:
rmudgett Review: https://reviewboard.asterisk.org/r/2685/
* include/asterisk/channel.h: Fixup doxygen on ast_hangup().
2013-07-18 19:25 +0000 [r394759] Mark Michelson <mmichelson@digium.com>
* res/res_sip_session.c, res/res_sip/sip_global_headers.c (added),
res/res_sip/config_system.c (added),
res/res_sip_one_touch_record_info.c, res/res_sip_mwi.c,
res/res_sip_pubsub.c, res/res_sip/config_transport.c,
res/res_sip/sip_configuration.c, res/res_sip_refer.c,
include/asterisk/res_sip.h, res/res_sip/config_global.c (added),
res/res_sip/include/res_sip_private.h, res/res_sip.exports.in,
res/res_sip_sdp_rtp.c, channels/chan_gulp.c,
res/res_sip_caller_id.c, res/res_sip.c: Add a bunch of options
from sip.conf to res_sip.conf For a complete list of the options
added, see the review linked at the bottom of this commit
message. (closes issue ASTERISK-21506) reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2671
2013-07-18 18:05 +0000 [r394744] David M. Lee <dlee@digium.com>
* res/res_http_websocket.c: Fixed null dereference when WebSocket
subprotocol isn't specified
2013-07-18 16:49 +0000 [r394731] Jonathan Rose <jrose@digium.com>
* apps/app_bridgewait.c, main/bridging_roles.c,
bridges/bridge_holding.c: bridge_holding/app_bridgewait: Add new
entertainment options This patch adds more entertainment options
to holding bridges and the bridge_wait application. Also, holding
bridges will now use music on hold as the default entertainment
option instead of none. The parameters for app_bridgewait have
changed to (role, options) from the previous (options) and the
options themselves have changed as well (entertainment options
are now contained in an enumerator, role specification is handled
by the role parameter, etc) (closes issue ASTERISK-21923)
Reported by: Matthew Jordan Review:
https://reviewboard.asterisk.org/r/2679/
2013-07-18 16:03 +0000 [r394715] Jason Parker <jparker@digium.com>
* res/stasis_http/resource_channels.c,
include/asterisk/stasis_app.h, include/asterisk/channel.h,
res/res_mutestream.c, main/channel.c, res/stasis/control.c: ARI:
Add support for suppressing media streams. Also convert
res_mutestream to use the core feature behind this. (closes issue
ASTERISK-21618) Review: https://reviewboard.asterisk.org/r/2652/
2013-07-18 14:50 +0000 [r394701] Matthew Jordan <mjordan@digium.com>
* main/http.c: Tweak debug statements This patch does two things:
1. It moves the debug statement that shows the HTTP sub-protocols
being compared after the string length calculation such that it
shows the correct string length in the output 2. It adds some
additional debug that displays when it matches on a sub-protocol
and when it fails
2013-07-18 14:08 +0000 [r394686] David M. Lee <dlee@digium.com>
* main/stasis_cache.c: Fix caching topic shutdown assertions The
recent changes to update stasis_cache_topics directly from the
publisher thread uncovered a race condition, which was causing
asserts in the /stasis/core tests. If the caching topic's
subscription is the last reference to the caching topic, it will
destroy the caching topic after the final message has been
processed. When dispatching to a different thread, this usually
gave the unsubscribe enough time to finish before destruction
happened. Now, however, it consistently destroys before
unsubscription is complete. This patch adds an extra reference to
the caching topic, to hold it for the duration of the
unsubscription. This patch also removes an extra unref that was
happening when the final message was received by the caching
topic. It was put there because of an extra ref that was put into
the caching topic's constructor. Both have been removed, which
makes the destructor a bit less confusing. Review:
https://reviewboard.asterisk.org/r/2675/
2013-07-18 12:54 +0000 [r394642] Michael L. Young <elgueromexicano@gmail.com>
* /, res/res_agi.c: Properly indicate failure to open an audio
stream in res_agi If there is an error streaming an audio file,
the current return status makes it difficult for an AGI script to
determine that there was an error with the audio file. This
patches changes the result to return -1 and the function returns
RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other
parts of res_agi, this would appear to be the proper way to
handle an error. (closes issue ASTERISK-21903) Reported by: Ariel
Wainer Tested by: Ariel Wainer Patches:
asterisk-21903-return-stream-res_1.8.diff by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2625/
........ Merged revisions 394640 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394641 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-07-17 22:30 +0000 [r394600-394623] Richard Mudgett <rmudgett@digium.com>
* tests/test_app.c, main/features.c, tests/test_voicemail_api.c,
tests/test_cel.c, include/asterisk/channel.h,
addons/chan_mobile.c, tests/test_cdr.c,
tests/test_stasis_endpoints.c, apps/app_voicemail.c,
main/channel.c, main/dial.c, apps/app_meetme.c: Change
ast_hangup() to return void and be NULL safe. Since ast_hangup()
is effectively a channel destructor, it should be a void
function. * Make the few silly callers checking the return value
no longer do so. Only the CDR and CEL unit tests checked the
return value. * Make all callers take advantage of the NULL safe
change and remove the NULL check before the call.
* main/features.c: Remove some completed and no longer relevant
BUGBUG notes.
2013-07-17 18:26 +0000 [r394583] Jonathan Rose <jrose@digium.com>
* apps/confbridge/conf_chan_announce.c: app_confbridge: Eliminate a
reference leak for confbridge announcer channels
2013-07-17 17:49 +0000 [r394552-394567] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: Left over spacing issues of review 726.
* channels/chan_dahdi.c: handle DAHDI_EVENT_REMOVED on a pri
D-Channel When a DAHDI device is removed at run-time it sends the
event DAHDI_EVENT_REMOVED on each channel. This is intended to
signal the userspace program to close the respective file handle,
as the driver of the device will need all of them closed to
properly clean-up. This event has long since been handled in
chan_dahdi (chan_zap at the time). However the event that is sent
on a D-Channel of a "PRI" (ISDN) span simply gets ignored. This
commit adds handling for closing the file descriptor (and
shutting down the span, while we're at it). It also adds a CLI
command 'pri destroy span <N>' to destroy the span and its DAHDI
channels. Review: https://reviewboard.asterisk.org/r/726/
2013-07-16 22:33 +0000 [r394530-394531] Matthew Jordan <mjordan@digium.com>
* apps/app_confbridge.c, CHANGES: Add 'kick all' capability to
ConfBridge CLI command This patch adds the ability to kick all
users out of a conference from the ConfBridge kick CLI command.
It is invoked by passing 'all' as the channel parameter to the
CLI command, i.e., "confbridge kick <conf> all". Note that this
patch was modified slightly to conform to trunk. (closes issue
ASTERISK-21827) Reported by: dorianlogan patches:
kickall-patch_v2.diff uploaded by dorianlogan (License 6504)
* main/cel.c: Re-order handlers in CEL to ensure that HANGUP events
happen after APP_END When a channel is hungup, both an APP_END
event and a HANGUP event can be fired. To ensure that HANGUP
events occur after APP_END events, the method callbacks for the
APP_END event should be processed prior to the callbacks for the
HANGUP event.
2013-07-16 21:44 +0000 [r394513] David M. Lee <dlee@digium.com>
* res/stasis_http/ari_websockets.c: Debug logging to help with
WebSocket connection problems
2013-07-16 20:00 +0000 [r394489] Richard Mudgett <rmudgett@digium.com>
* channels/chan_gulp.c: chan_gulp: Fix gulp_indicate() handling of
AST_CONTROL_PVT_CAUSE_CODE.
2013-07-16 19:13 +0000 [r394473] Mark Michelson <mmichelson@digium.com>
* res/res_sip_session.c: Prevent crash from trying to end a session
in an invalid way. This ensures that code that was only meant to
be run on a reinvite failure only runs on a reinvite failure.
(closes issue ASTERISK-22061) reported by Rusty Newton
2013-07-16 18:49 +0000 [r394470-394471] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, channels/chan_sip.c: Remove some dead code
dealing with old bridging method.
* bridges/bridge_simple.c: Simplify bridge_simple chan join code.
2013-07-16 18:22 +0000 [r394469] Matthew Jordan <mjordan@digium.com>
* main/cdr.c: Re-order cleanup This patch attempts to fix some
possible race conditions in shutdown of the CDR engine. It: *
Adds a cleanup handler to only unsubscribe and join on stasis
messages during graceful shutdown. The cleanup handler should
execute before the regular atexit handler, as we want to
unsubscribe for any further messages before dispatching the CDRs.
* The CDRs are now locked when we dispatch them on shutdown.
2013-07-16 15:30 +0000 [r394442] David M. Lee <dlee@digium.com>
* res/res_http_websocket.c: Fixed null dereference when WebSocket
protocol is omitted
2013-07-15 23:20 +0000 [r394417] Richard Mudgett <rmudgett@digium.com>
* configs/agents.conf.sample, include/asterisk/config_options.h,
include/asterisk/stasis_channels.h, channels/chan_agent.c
(removed), configs/queues.conf.sample,
include/asterisk/bridging.h, UPGRADE.txt, main/stasis_channels.c,
CHANGES, main/bridging.c, apps/app_agent_pool.c (added): Replace
chan_agent with app_agent_pool. The ill conceived chan_agent is
no more. It is now replaced by app_agent_pool. Agents login using
the AgentLogin() application as before. The AgentLogin()
application no longer does any authentication. Authentication is
now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice
prompts asked for.) Sample extensions.conf [login] ; Sample agent
1001 login ; Set COLP for in between calls so the agent does not
see the last caller COLP. exten =>
1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>) ; Give the
agent DTMF transfer and disconnect features when connected to a
caller. same => n,Set(CHANNEL(dtmf-features)=TX) same =>
n,AgentLogin(1001) same => n,NoOp(AGENT_STATUS is
${AGENT_STATUS}) same => n,Hangup() [caller] ; Sample caller
direct connect to agent 1001 exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same =>
n,Hangup() ; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q) same => n,Hangup() Sample
queues.conf [agent_q] member =>
Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation
overview: 1) Logged in agents wait for callers in an agents
holding bridge. 2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller
joins the basic bridge to wait for the agent. 4) The agent is
either automatically connected to the caller or must ack the call
to connect. 5) The agent is moved from the agents holding bridge
to the basic bridge. 6) The agent and caller talk. 7) The
connection is ended by either party. 8) The agent goes back to
the agents holding bridge. To avoid some locking issues with the
agent holding bridge, I needed to make some changes to the after
bridge callback support. The after bridge callback is now a list
of requested callbacks with the last to be added the only active
callback. The after bridge callback for failed callbacks will
always happen in the channel thread when the channel leaves the
bridging system or is destroyed. (closes issue ASTERISK-21554)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2657/
2013-07-15 22:05 +0000 [r394402] Mark Michelson <mmichelson@digium.com>
* include/asterisk/stasis_channels.h: Remove misleading
documentation for channel snapshot creation.
2013-07-15 21:22 +0000 [r394397] David M. Lee <dlee@digium.com>
* res/res_stasis_http.c: Document the ari.conf allowed_origins
setting
2013-07-15 13:43 +0000 [r394370] Joshua Colp <jcolp@digium.com>
* res/res_sip_session.c, include/asterisk/res_sip_session.h: Remove
some callbacks and functions which are not needed.
2013-07-14 02:41 +0000 [r394278-394346] Matthew Jordan <mjordan@digium.com>
* /, apps/app_queue.c: Provide error message for QUEUE_MEMBER when
member is not in queue When QUEUE_MEMBER is used and the member
specified is not in the queue, Asterisk provides an ERROR message
that indicates that the option specified is not valid. This patch
now properly displays an ERROR message that the member is not in
the queue if an interface is specified. (closes issue
ASTERISK-21980) Reported by: Avraam David ........ Merged
revisions 394345 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/dns.c: Remove redundant code in dns.c Peter J Philipp
pointed out that there are two checks that ensure that len is not
less than 0. If len is less than 0, the function returns. Having
both of them is clearly redundant. This removes the second and
attempts to clarify (slightly) the error condition. (closes issue
ASTERISK-21772) Reported by: Peter J Philipp
* /, funcs/func_strings.c: Clarify documentation for function
PASSTHRU It is not apparent to the average user that the PASSTHRU
function should not be passed as ${PASSTHRU(string)} but just as
PASSTHRU(string) to functions which take a variable name and not
its contents. This patch clarifies the behavior in the
documentation and provides an example. (closes issue
ASTERISK-21717) Reported by: Richard Miller patches:
func_strings.diff uploaded by Richard Miller (license 5685)
........ Merged revisions 394302 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394303 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/bridging.c, main/cdr.c: Fix FRACK message from external
redirects; handle outbound channels better This patch does the
following: * It simplifies the Dial handling in CDRs. As a rule,
the caller in a dial relationship is always the Party A. There
was some logic present in the handling of the dial message that
could, conceivably, pick the caller as Party A for the beginning
of the dial and the peer as Party A for the end of the dial. This
shouldn't have happened if the code in the bridging framework was
doing its job; however, that was broken and it led to the FRACK.
As it is, this code was overly ocmplex and not needed: the
caller, if present, should always be Party A. Period. * It
properly checks to see if a channel will continue on in the
dialplan. ast_check_hangup - much like cake at the end - is a
lie. It will tell you that you are hungup when you are not. Do
not believe it. I would make this function tell the truth, but
I'm nervous that we've been depending on it sitting on its throne
of lies for far too long, and it would probably break lots of
things. So I'm just checking the "internal" soft hangup flags,
like everyone else. (closes issue ASTERISK-22060) Reported by:
Mark Michelson (issue ASTERISK-21831) Reported by: Matt Jordan
* channels/chan_sip.c: Pretty up a debug message if the
referred-by-uri isn't available Instead of formatting a NULL
pointer into a "%s" format string (which is usually not a good
thing to do), we instead print "Unknown".
2013-07-12 22:35 +0000 [r394263] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Fix a longstanding issue with MFC-R2
configuration that prevented users from mixing different variants
or general MFC-R2 settings within the same E1 line. Most users do
not have a problem with this since MFC-R2 lines are usually
fractional E1s, or the whole E1 has the same country variant and
R2 settings. In Venezuela however is common to have inbound
MFC-R2 and outbound DTMF-R2 within the same E1. This fix now
properly parses the chan_dahdi.conf file to generate a new openr2
context every time a new channel => section is found and the
configuration was changed. (closes issue ASTERISK-21117) Reported
by: Rafael Angulo Related Elastix issue:
http://bugs.elastix.org/view.php?id=1612 ........ Merged
revisions 394106 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 394173 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-07-12 21:42 +0000 [r394249] Joshua Colp <jcolp@digium.com>
* main/channel.c, main/channel_internal_api.c,
include/asterisk/channel.h, main/bridging.c: Add support to the
bridging core for performing COLP updates when channels join a 2
party bridge. (closes issue ASTERISK-21829) Review:
https://reviewboard.asterisk.org/r/2636/
2013-07-12 21:01 +0000 [r394232] Mark Michelson <mmichelson@digium.com>
* main/bridging_basic.c: Prevent potential race condition in
multiparty basic bridges. For more details about the race
condition see the linked review at the bottom of this commit
(closes issue ASTERISK-21882) Reported by Matt Jordan Review:
https://reviewboard.asterisk.org/r/2663
2013-07-12 19:35 +0000 [r394216] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix a compiler warning.
2013-07-12 18:23 +0000 [r394203] David M. Lee <dlee@digium.com>
* tests/test_json.c: Fixed intermittent crash when loading
test_json.so The JSON test attempted an overly clever use of
RAII_VAR to run code at the beginning and end of each test, in
order to validate that no JSON objects were leaked during the
test. The problem is that the validation code would run during
the initial load, when the tests were initialized. This happens
during startup, when other parts of the system might actively be
allocating and freeing JSON objects. This patch changes the
RAII_VAR to use the new ast_test_register_{init,cleanup}
functions to run the validations properly. (closes issue
ASTERISK-21978) Review: https://reviewboard.asterisk.org/r/2669/
2013-07-12 17:52 +0000 [r394189] Jason Parker <jparker@digium.com>
* res/stasis_http/internal.h, res/stasis_http/config.c,
res/stasis_http/cli.c, res/res_stasis_http.c: ARI: Add support
for Cross-Origin Resource Sharing (CORS), origin headers This
rejects requests from any unknown origins. (closes issue
ASTERISK-21278) Review: https://reviewboard.asterisk.org/r/2667/
2013-07-11 21:01 +0000 [r394158] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridging_technology.h: Fix bridge tech write
callback parameter name.
2013-07-11 20:59 +0000 [r394156] David M. Lee <dlee@digium.com>
* channels/chan_skinny.c: Fixed chan_skinny for systems were
pthread_t isn't an int. I'm looking at you, OS X.
2013-07-11 20:17 +0000 [r394147] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Refactor and cleanup of skinny session
handling. Major changes are to pull all packet reading functions
into skinny_session and move timeout handling to scheduling
arrangements. Thread cancelling is now undertaken directly rather
than waiting for the read to timeout (cleanup is popped on thread
cancel). Also added some keepalive timings in debugging messages.
Keepalive timeout has been increased from 1.1 by keepalive to 3
times keepalive. This seems to align (after keepalives stabilise)
with when devices reset after not receiving keepalives. Probably
needs more work, especially around the first and/or second
keepalives that vary significantly by device and firmware
version. Review: https://reviewboard.asterisk.org/r/2611/
2013-07-11 16:23 +0000 [r394103] Joshua Colp <jcolp@digium.com>
* res/res_sip_exten_state.c: Tweak the subscription failure warning
message to include endpoint name and context.
2013-07-11 15:37 +0000 [r394037-394089] David M. Lee <dlee@digium.com>
* tests/test_cel.c: Correct test_cel cleanup. When I corrected the
CEL test crash in r394037, I didn't quite pay attention to how
the globals and locals were being shuffled around in the cleanup
callback. I removed the nulling of the global variables, which
caused them to be double cleaned. This patch puts the global
nulling code back (since the vars are cleaned up by RAII_VARs),
and removes the explicit ao2_cleanup() (since they were no-ops,
because the variables had just been nulled).
* res/stasis_http/config.c, configs/ari.conf.sample,
res/res_stasis_http.c: Change ARI user config to use a type field
When I initially wrote the configuration support for ARI users, I
determined the section type by a category prefix (i.e.,
[user-admin]). This is neither idiomatic Asterisk configuration,
nor is it really that user friendly. This patch replaces the
category prefix with a type field in the section, which is much
cleaner. Review: https://reviewboard.asterisk.org/r/2664/
* res/stasis_http/config.c: Apply defaults to ari.conf's general
section
* tests/test_voicemail_api.c: test_voicemail_api: fix warning found
by gcc-4.8 The voicemail_api test had code like strncmp(a, b,
sizeof(a)), but a was a char pointer, instead of a literal or
char array. This meant that sizeof was the size of the pointer,
not the length of the string. Since the string is in a
stringfield and should be null terminated, I just changed it to a
plain strcmp.
* tests/test_cel.c: Fixed some CEL test crashes
2013-07-10 22:26 +0000 [r394024] Kevin Harwell <kharwell@digium.com>
* contrib/scripts/sip_to_res_sip (added),
contrib/scripts/sip_to_res_sip/astconfigparser.py (added),
contrib/scripts/sip_to_res_sip/astdicts.py (added),
contrib/scripts/sip_to_res_sip/sip_to_res_sip.py (added): PSJIP -
sip.conf to res_sip.conf script ** This script is in no way
finished. Started the initial "cut" at converting a sip.conf file
to a res_sip.conf file. Hopefully the bulk of the framework is in
place and only a few minor adjustments need to be made when an
option mapping is added that "doesn't fit". This script and
supporting files should be executable against python version 2.5.
An OrderedDict class (backported from a newer version of python)
is included. A MultiOrderedDict class is implemented so options,
when added, should be able to be added in order and allowed to
have multiple values. Currently the scripts supports the majority
of endpoint options found in res_sip.conf. Support has also been
added for Aor(s) and the ACL/security sections. Inside the
sip_to_res_sip.py file one can see a list of options that still
need to be mapped. Also items that still need to be done:
templates, includes, parsing '=>' delimiter. Note that some code
is hopefully in place already to support templates (e.g.
lookup/retrieving defaults from them). However, the parsing of
and adding of the section needs to be done.
2013-07-10 20:02 +0000 [r394004] Joshua Colp <jcolp@digium.com>
* res/res_sip_outbound_registration.c: Handle outbound registration
failures that do not occur as a result of a real response.
(closes issue ASTERISK-22064) Reported by: Rusty Newton
2013-07-10 17:13 +0000 [r393968-393987] David M. Lee <dlee@digium.com>
* res/res_stasis_http_channels.c, rest-api/api-docs/channels.json:
Document the 400 error response for originate
* res/res_stasis_http_asterisk.c, rest-api/api-docs/asterisk.json,
res/stasis_http/ari_model_validators.c,
res/res_stasis_http_channels.c, rest-api/api-docs/channels.json,
res/stasis_http/ari_model_validators.h: Corrected api-docs for
channel variables
2013-07-10 01:56 +0000 [r393930] Russell Bryant <russell@russellbryant.com>
* configs/sla.conf.sample, /, apps/app_meetme.c: astobj2-ify the
SLA code The SLA code within app_meetme was written before
asotbj2 had been merged into Asterisk. Worse, support for reloads
did not exist at first and was added later as a bolt-on feature.
I knew at the time that reloading was not safe at all while SLA
was in use, so the reload would be queued up to execute when the
system was idle. Unfortunately, this approach was still prone to
errors beyond the fact that this was the only place in Asterisk
where configuration was not reloaded instantly when requested.
This patch converts various SLA objects to be reference counted
objects using astobj2. This allows reloads to be processed while
the system is in use. The code ensures that the objects will not
disappear while one of the other threads is using them. However,
they will be immediately removed from the global trunk and
station containers so no new calls will use them if removed from
configuration. Review: https://reviewboard.asterisk.org/r/2581/
........ Merged revisions 393928 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 393929 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-07-09 21:40 +0000 [r393919] Jason Parker <jparker@digium.com>
* include/asterisk/lock.h: Make SCOPED_LOCK use RAII_VAR. This
fixes an issue with requiring SCOPED_LOCK to be the last variable
declaration and removes duplicate code in the process. Review:
https://reviewboard.asterisk.org/r/2665/
2013-07-09 21:06 +0000 [r393910] Richard Mudgett <rmudgett@digium.com>
* main/xmldoc.c: Fix printf NULL string (null) substituion for NULL
config framework default.
2013-07-09 20:07 +0000 [r393897] Mark Michelson <mmichelson@digium.com>
* channels/chan_gulp.c: Use correct function for getting bridged
peer when doing direct media checks. (closes issue
ASTERISK-21947) reported by Matt Jordan
2013-07-09 19:38 +0000 [r393896] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/manager.h, include/asterisk/stasis_channels.h:
Fix some stasis doxygen comments.
2013-07-09 11:05 +0000 [r393857-393870] Joshua Colp <jcolp@digium.com>
* res/res_sip_outbound_registration.c: Ensure all pjsip_regc_*
access occurs within a pjlib thread. (closes issue
ASTERISK-22054) Reported by: Rusty Newton
* res/res_sip/config_auth.c: Tweak log message slightly.
* res/res_sip/config_auth.c: Treat the authentication object as
invalid if digest configuration is chosen and the digest is not
of the correct length. (closes issue ASTERISK-22003) Reported by:
Rusty Newton
2013-07-08 20:31 +0000 [r393834-393843] David M. Lee <dlee@digium.com>
* res/res_stasis_recording.c: Oh menuconfig, why do you hate
margins?
* res/stasis_http/ari_websockets.c: Better structure for the
WebSocket validation failure message
2013-07-08 19:53 +0000 [r393831-393833] Joshua Colp <jcolp@digium.com>
* res/res_sip/config_transport.c: Ensure that a valid bind host is
specified for transports. (closes issue ASTERISK-22017) Reported
by: Rusty Newton
* main/channel_internal_api.c, res/res_agi.c,
main/manager_bridging.c, include/asterisk/channel.h,
main/stasis_channels.c, main/bridging.c, main/manager_channels.c,
main/cli.c, main/channel.c, build_tools/cflags-devmode.xml,
main/pbx.c, include/asterisk/stasis_channels.h, main/manager.c:
Refactor operations to access the stasis cache instead of objects
directly when retrieving information. (closes issue
ASTERISK-21883) Review: https://reviewboard.asterisk.org/r/2645/
2013-07-08 16:04 +0000 [r393816] David M. Lee <dlee@digium.com>
* res/res_stasis_http.c: res_stasis_http doesn't depend on
res_stasis any more
2013-07-08 15:59 +0000 [r393815] Jonathan Rose <jrose@digium.com>
* res/parking/parking_controller.c, main/bridging.c,
res/parking/parking_bridge.c, res/parking/res_parking.h:
res_parking: Apply ringing role option on swap with a channel
that rings (closes issue ASTERISK-21877) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2656/
2013-07-08 15:11 +0000 [r393807] Joshua Colp <jcolp@digium.com>
* res/stasis/control.c: Fix building.
2013-07-08 14:46 +0000 [r393804-393806] Jason Parker <jparker@digium.com>
* res/res_stasis_http_asterisk.c, res/stasis/control.c,
res/stasis_http/resource_asterisk.h,
rest-api/api-docs/asterisk.json,
res/stasis_http/resource_channels.c,
res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
res/stasis_http/resource_channels.h,
rest-api/api-docs/channels.json,
res/stasis_http/resource_asterisk.c: ARI: Add support for
getting/setting channel and global variables. This allows for
reading and writing of functions on channels. (closes issue
ASTERISK-21868) Review: https://reviewboard.asterisk.org/r/2641/
* main/manager_system.c (added), res/res_stun_monitor.c,
main/file.c, main/sounds_index.c,
include/asterisk/stasis_system.h (added), channels/chan_iax2.c,
include/asterisk/manager.h, main/asterisk.c, include/asterisk.h,
main/stasis_system.c (added), main/manager.c,
channels/chan_sip.c: Move channel driver Registry manager events
to core. This also shuffles the stasis system topic and related
handling. (closes issue ASTERISK-21488) Review:
https://reviewboard.asterisk.org/r/2631/
2013-07-08 14:26 +0000 [r393801] Matthew Jordan <mjordan@digium.com>
* include/asterisk/core_local.h, include/asterisk/bridging.h,
main/core_unreal.c, main/core_local.c, CHANGES, main/bridging.c,
include/asterisk/core_unreal.h: Create Local channel messages on
the Stasis message bus and produce AMI events This patch does the
following: * It adds a virtual table of callbacks to core_unreal.
These callbacks can be supplied by concrete implementations of
"unreal" channel drivers, which lets the unreal channel driver
call specific functionality when it performs some action.
Currently, this is done to notify implementations when an
optimization operation has begun, and when an optimization
operation has succeeded. * It adds Stasis-Core messages for Local
channel bridging and Local channel optimization. Local channel
optimization is now two events: a Begin and an End. Some
consumers of Stasis-Core may want to know when an operation is
beginning so that they can 'prepare' their information; others
will be more concerned about when the operation has completed, so
that they can 'fix up' information. Stasis-Core allows for both,
as does AMI. Review: https://reviewboard.asterisk.org/r/2552
2013-07-08 13:57 +0000 [r393793] Mark Michelson <mmichelson@digium.com>
* res/res_sip_caller_id.c: Fix some broken logic in sending
outbound caller ID. * trust_id_outbound was required even when
the caller ID was not marked private. This is against intentions
and documentation. * We now check both name and number privacy
instead of checking name privacy twice.
2013-07-07 21:29 +0000 [r393777-393785] Matthew Jordan <mjordan@digium.com>
* main/channel.c: In a channel destructor dispose of items that
raise Stasis message properly This patch reorders certain actions
that may raise Stasis messages in the channel destructor such
that they occur before the Stasis cache is cleared. Once the
Stasis cache is cleared, its rather a bad idea to be trying to
publish information about a channel. (closes issue
ASTERISK-22001) Reported by: Jonathan Rose
* main/cdr.c, main/channel.c, main/pbx.c,
include/asterisk/stasis_channels.h, main/channel_internal_api.c,
include/asterisk/cdr.h, include/asterisk/channel.h,
main/stasis_channels.c, CHANGES, main/cel.c,
main/manager_channels.c: Handle hangup logic in the Stasis
message bus and consumers of Stasis messages This patch does the
following: * It adds a new soft hangup flag
AST_SOFTHANGUP_HANGUP_EXEC that is set when a channel is
executing dialplan hangup logic, i.e., the 'h' extension or a
hangup handler. Stasis messages now also convey the soft hangup
flag so consumers of the messages can know when a channel is
executing said hangup logic. * It adds a new channel flag,
AST_FLAG_DEAD, which is set when a channel is well and truly
dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs, and
other consumers of Stasis have been updated to look for this flag
to know when the channel should by lying six feet under. * The
CDR engine has been updated to better handle a channel entering
and leaving a bridge. Previously, a new CDR was automatically
created when a channel left a bridge and put into the 'Pending'
state; however, this way of handling CDRs made it difficult for
the 'endbeforehexten' logic to work correctly - there was always
a new CDR waiting in the hangup logic and, even if 'ended',
wouldn't be the CDR people wanted to inspect in the hangup
routine. This patch completely removes the Pending state and
instead defers creation of the new CDR until it gets a new
message that requires a new CDR.
2013-07-05 22:08 +0000 [r393749-393768] David M. Lee <dlee@digium.com>
* res/res_stasis_http.c: ARI: return a 503 if Asterisk isn't fully
booted
* res/stasis_http/ari_websockets.c: Print error details when set
nonblock fails
* res/stasis_http/ari_model_validators.c,
res/stasis_http/ari_model_validators.h,
res/stasis_http/resource_events.c, res/res_stasis_http_events.c,
rest-api/api-docs/events.json: Document MissingParams error
message for /ari/events
2013-07-05 17:33 +0000 [r393740] Matthew Jordan <mjordan@digium.com>
* include/asterisk/cdr.h, include/asterisk/channel.h,
channels/chan_gtalk.c, include/asterisk/json.h,
channels/chan_gulp.c, channels/chan_jingle.c, main/json.c,
main/manager.c, channels/chan_skinny.c, channels/chan_motif.c,
channels/chan_h323.c, include/asterisk/rtp_engine.h,
main/asterisk.c, channels/chan_mgcp.c, channels/chan_unistim.c,
res/res_rtp_asterisk.c, channels/chan_multicast_rtp.c,
main/rtp_engine.c, channels/chan_sip.c: Refactor RTCP events over
to Stasis; associate with channels This patch does the following:
* It merges Jaco Kroon's patch from ASTERISK-20754, which
provides channel information in the RTCP events. Because Stasis
provides a cache, Jaco's patch was modified to pass the channel
uniqueid to the RTP layer as opposed to a pointer to the channel.
This has the following benefits: (1) It keeps the RTP engine
'clean' of references back to channels (2) It prevents circular
dependencies and other potential ref counting issues * The RTP
engine now allows any RTP implementation to raise RTCP messages.
Potentially, other implementations (such as res_rtp_multicast)
could also raise RTCP information. The engine provides structs to
represent RTCP headers and RTCP SR/RR reports. * Some general
refactoring in res_rtp_asterisk was done to try and tame the RTCP
code. It isn't perfect - that's *way* beyond the scope of this
work - but it does feel marginally better. * A few random bugs
were fixed in the RTCP statistics. (Example: performing an
assignment of a = a is probably not correct) * We now raise RTCP
events for each SR/RR sent/received. Previously we wouldn't raise
an event when we sent a RR report. Note that this work will be of
use to others who want to monitor call quality or build modules
that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to
accomplish. It is also a first step (though by no means the last
step) towards getting Olle's pinefrog work incorporated. Again:
note that the patch by Jaco Kroon was modified slightly for this
work; however, he did all of the hard work in finding the right
places to set the channel in the RTP engine across the channel
drivers. Much thanks goes to Jaco for his hard work here. Review:
https://reviewboard.asterisk.org/r/2603/ (closes issue
ASTERISK-20574) Reported by: Jaco Kroon patches:
asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)
(closes issue ASTERISK-21471) Reported by: Matt Jordan
2013-07-05 14:54 +0000 [r393729] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: OneTouchRecord: Add function defined earlier:
ast_bridge_features_do()
2013-07-05 03:08 +0000 [r393716] Matthew Jordan <mjordan@digium.com>
* main/stasis_channels.c, include/asterisk/stasis_channels.h:
Remove parkinglot from the channel snapshot Legacy channel
drivers often include the ability to set a default parking lot on
an endpoint basis; when channels are created for that endpoint,
they inherit the parkinglot option. Parking used to use this
option more frequently; while it is still supported, other
options (such as using channel variables or creation of a custom
parkinglot) are supported. More importantly, conveying the
parkinglot information through a channel snapshot isn't terribly
useful - it is rarely (if ever) changed on a channel and some
consumers of channel snapshots, such as ARI, will never use the
information. (closes issue ASTERISK-21968) Reported by: Matt
Jordan
2013-07-04 18:46 +0000 [r393704] Jonathan Rose <jrose@digium.com>
* res/parking/parking_ui.c, main/parking.c,
res/parking/parking_controller.c, UPGRADE.txt,
res/parking/parking_applications.c, include/asterisk/channel.h,
main/cel.c, CHANGES, res/parking/parking_bridge_features.c,
res/parking/parking_bridge.c, main/channel.c,
res/parking/res_parking.h, bridges/bridge_builtin_features.c,
main/features.c, include/asterisk/parking.h, main/bridging.c,
res/parking/parking_manager.c: res_parking: Replace Parker
snapshots with ParkerDialString This process also involved a
large amount of rework regarding how to redial the Parker when a
channel leaves a parking lot due to timeout. An attended transfer
channel variable has been added to attended transfers to
extensions that will eventually park (but haven't at the time of
transfer) as well. This resolves one of the two BUGBUG comments
remaining in res_parking. (issues ASTERISK-21877) Reported by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/2638/
2013-07-04 13:37 +0000 [r393675-393687] David M. Lee <dlee@digium.com>
* res/res_ari_model.c: Fix int width problem for 32-bit... again
* tests/test_ari_model.c: Fix int width problem for 32-bit
* main/utils.c, main/crypt.c (added), main/Makefile: Fix utils
directory breakage.
2013-07-03 23:59 +0000 [r393600-393633] Richard Mudgett <rmudgett@digium.com>
* main/config_options.c: Add BUGBUG note for ASTERISK-22009
* channels/chan_agent.c (added), configs/queues.conf.sample,
include/asterisk/bridging.h, UPGRADE.txt, main/config_options.c,
main/stasis_channels.c, CHANGES, main/bridging.c,
apps/app_agent_pool.c (removed), configs/agents.conf.sample,
include/asterisk/config_options.h,
include/asterisk/stasis_channels.h: Revert accidental overcommit.
* channels/chan_agent.c (removed), configs/queues.conf.sample,
include/asterisk/bridging.h, UPGRADE.txt, main/config_options.c,
main/stasis_channels.c, CHANGES, main/bridging.c,
apps/app_agent_pool.c (added), configs/agents.conf.sample,
include/asterisk/config_options.h,
include/asterisk/stasis_channels.h: Add BUGBUG note for
ASTERISK-22009
* channels/chan_dahdi.c, /: chan_dahdi: Fix segfault reloading
chan_dahdi when round robin is used. * Clear round_robin[] in
dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo
Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621)
patch uploaded by rmudgett ........ Merged revisions 393627 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 393628 from
http://svn.asterisk.org/svn/asterisk/branches/11
* bridges/bridge_builtin_features.c,
include/asterisk/bridging_features.h: OneTouchRecord: Make so
Monitor/MixMonitor can be toggled/started/stopped. The
OneTouchRecord feature has historically been a toggle. This patch
adds the ability to make the OneTouchRecord hook optionally
start/stop recording only. If OneTouchRecord is already doing
what is requested then only the invoker hears the courtesy tone
and/or start/stop recording message. The new feature is written
so we could easily add explicit start/stop recording DTMF hooks
for Monitor and MixMonitor. The majority of the changes in
bridge_builtin_features.c is a refactoring of the OneTouchRecord
code (Monitor and MixMonitor versions) so it is easy to direct
the toggle/start/stop functionality. Review:
https://reviewboard.asterisk.org/r/2655/
* main/bridging.c: Move when bridge channel enter is published so
it does not interrupt the thought of some lines of code.
* main/stasis_config.c: Fix some indentation in stasis_config.c.
2013-07-03 22:04 +0000 [r393589-393599] Matthew Jordan <mjordan@digium.com>
* main/cdr.c: Fix some bugs in CDRs; add some CLI commands to help
debugging This patch fixes a few minor bugs and one major one:
the CDR by bridge container was less than helpful. The mechanism
previously used to try and find all of the CDRs in a particular
bridge ended up missing CDRs, resulting in incorrect records.
When looking up CDRs in a bridge, we now just bite the bullet and
do a selection across all existing CDRs.
* main/stasis_config.c: Let Stasis load itself with default values
While a Stasis configuration file is nice, it shouldn't be
mandatory. We can carry on with default values.
2013-07-03 20:41 +0000 [r393586] Mark Michelson <mmichelson@digium.com>
* main/bridging.c: Publish a bridge enter before pulling on a
push-and-swap operation. Prior to this patch, the order of
procedures on a bridge push was * Add new bridge channel to
bridge's array. * Pull the swap channel out of the bridge *
Publish a bridge enter event. The problem is that when the swap
channel was pulled from the bridge, a bridge leave event would be
published. The bridge snapshot published during the bridge leave
showed the new channel that had been added to the bridge, but
there had been no bridge enter event for that channel. The fix
provided here was to change the order a bit * Add new bridge
channel to bridge's array. * Publish bridge enter event. * Pull
the swap channel out of the bridge. This makes it so that the
bridge snapshots during the stasis events are accurate.
2013-07-03 19:46 +0000 [r393528-393576] David M. Lee <dlee@digium.com>
* res/res_stasis_http_bridges.c, res/res_stasis_http_recordings.c,
res/stasis_http/ari_model_validators.h,
res/res_stasis_http_endpoints.c, res/res_stasis_http_events.c,
rest-api-templates/ari_model_validators.c.mustache,
rest-api-templates/res_stasis_http_resource.c.mustache,
rest-api-templates/ari_model_validators.h.mustache,
res/stasis_http/ari_model_validators.c,
res/res_stasis_http_channels.c, res/res_stasis_http_sounds.c: Fix
load errors related to the new ari_model_validators. The Asterisk
strategy of loading modules with RTLD_LAZY to extract metadata
from the module works well enough, until you try to take the
address of a function. If a module takes the address of a
function, that function needs to be resolved at load time. That
kinda defeats RTLD_LAZY. This patch adds some
ari_validator_{id}_fn() wrapper functions for safely getting the
function pointer from a different module.
* res/res_ari_model.c: Violating the margins to make menuconfig
happy
* res/res_stasis_recording.exports.in (added), Makefile,
include/asterisk/file.h, include/asterisk/paths.h,
main/channel.c, include/asterisk/app.h,
res/stasis_http/resource_channels.c, tests/test_utils.c,
apps/app_minivm.c, main/file.c,
res/stasis_http/resource_recordings.c, main/app.c,
res/res_stasis_recording.c (added),
rest-api-templates/swagger_model.py,
rest-api/api-docs/channels.json,
res/stasis_http/resource_channels.h,
res/res_stasis_http_bridges.c, rest-api/api-docs/recordings.json,
res/stasis_http/resource_recordings.h, main/asterisk.c,
rest-api-templates/asterisk_processor.py, apps/app_voicemail.c,
include/asterisk/utils.h, res/res_stasis_playback.c,
include/asterisk/stasis_app_recording.h (added),
res/res_stasis_http_channels.c, main/utils.c,
include/asterisk/channel.h, res/res_stasis_http_recordings.c: ARI
- channel recording support This patch is the first step in
adding recording support to the Asterisk REST Interface.
Recordings are stored in /var/spool/recording. Since recordings
may be destructive (overwriting existing files), the API rejects
attempts to escape the recording directory (avoiding issues if
someone attempts to record to ../../lib/sounds/greeting, for
example). (closes issue ASTERISK-21594) (closes issue
ASTERISK-21581) Review: https://reviewboard.asterisk.org/r/2612/
* include/asterisk/stasis.h, configs/stasis_core.conf.sample
(removed), main/asterisk.c, main/stasis.c, main/stasis_config.c
(added), configs/stasis.conf.sample (added): Configuration for
Stasis threadpool The appropriate settings for the Stasis
threadpool is very system specific, depending upon both workload
and system configuration. This patch adds a stasis.conf file
which can be used to configure the key attributes of the
threadpool for the Stasis message bus. (closes issue
ASTERISK-21280) Review: https://reviewboard.asterisk.org/r/2651/
* res/stasis_http/cli.c (added), res/Makefile,
configs/ari.conf.sample (added), makeopts.in,
res/res_stasis_http.c, res/stasis_http/internal.h (added),
configs/stasis_http.conf.sample (removed), main/Makefile,
res/stasis_http/config.c (added), main/http.c, main/utils.c: No
message for rev 393530 found
* main/json.c, rest-api/api-docs/asterisk.json,
rest-api/api-docs/playback.json,
res/stasis_http/ari_websockets.c, main/stasis_channels.c,
rest-api-templates/swagger_model.py,
res/res_stasis_http_bridges.c,
rest-api-templates/res_stasis_json_resource.c.mustache (removed),
res/res_stasis_json_recordings.exports.in (removed),
rest-api/api-docs/endpoints.json, main/stasis_endpoints.c,
rest-api/api-docs/events.json, tests/test_res_stasis.c,
tests/test_stasis_channels.c, include/asterisk/stasis_http.h,
res/res_stasis_json_sounds.exports.in (removed),
res/res_ari_model.exports.in (added),
res/res_stasis_http_recordings.c,
rest-api-templates/res_stasis_json_resource.exports.mustache
(removed), rest-api/api-docs/bridges.json,
res/res_stasis_http_events.c, res/res_ari_model.c (added),
res/res_stasis_json_playback.exports.in (removed),
res/res_stasis_http_sounds.c, res/stasis_json (removed),
rest-api/api-docs/recordings.json,
rest-api-templates/ari_model_validators.c.mustache (added),
res/res_stasis_json_endpoints.exports.in (removed),
res/res_stasis_json_events.exports.in (removed),
res/res_stasis_http_asterisk.c,
rest-api-templates/res_stasis_http_resource.c.mustache,
rest-api-templates/make_ari_stubs.py (added),
res/res_stasis_json_recordings.c (removed),
rest-api-templates/api.wiki.mustache (added),
rest-api/api-docs/sounds.json, res/Makefile,
res/res_stasis_json_events.c (removed),
res/res_stasis_json_bridges.exports.in (removed),
res/res_stasis_json_sounds.c (removed),
rest-api-templates/models.wiki.mustache (added),
main/stasis_bridging.c, rest-api-templates/transform.py,
rest-api-templates/stasis_json_resource.h.mustache (removed),
res/res_stasis_json_channels.exports.in (removed),
res/res_stasis_json_asterisk.c (removed), res/res_stasis_http.c,
rest-api-templates/asterisk_processor.py,
res/res_stasis_http_playback.c,
rest-api-templates/ari_model_validators.h.mustache (added),
res/res_stasis_http_channels.c, res/res_stasis_json_endpoints.c
(removed), include/asterisk/json.h, tests/test_ari_model.c
(added), Makefile, res/res_stasis_json_asterisk.exports.in
(removed), res/res_stasis_json_bridges.c (removed),
res/stasis_http/resource_recordings.c,
rest-api/api-docs/channels.json, res/res_stasis_json_playback.c
(removed), res/res_stasis.c, doc/rest-api (added),
rest-api-templates/make_stasis_http_stubs.py (removed),
res/stasis_http/resource_recordings.h,
res/res_stasis_json_channels.c (removed),
res/stasis_http/ari_model_validators.c (added),
rest-api-templates/event_function_decl.mustache (removed),
res/stasis_http/ari_model_validators.h (added),
res/res_stasis_http_endpoints.c: No message for rev 393529 found
* res/Makefile, res/res_http_websocket.c,
res/res_stasis_http.exports.in, configure, tests/test_utils.c,
res/stasis_http/ari_websockets.c (added),
rest-api-templates/stasis_http_resource.c.mustache,
tests/test_stasis_http.c, res/stasis_http/resource_events.c,
rest-api-templates/asterisk_processor.py,
include/asterisk/utils.h, res/res_stasis_http_playback.c,
res/res_http_websocket.exports.in,
res/stasis_http/resource_events.h,
res/res_stasis_http_channels.c, include/asterisk/stasis_http.h,
configure.ac, res/res_stasis_http_recordings.c,
rest-api-templates/param_parsing.mustache (added),
res/res_stasis_http_endpoints.c, res/res_stasis_http_events.c,
include/asterisk/http.h, res/res_stasis_http_sounds.c,
rest-api-templates/swagger_model.py,
res/res_stasis_http_bridges.c, res/res_stasis_http.c,
rest-api-templates/stasis_http_resource.h.mustache,
res/res_stasis_http_asterisk.c,
rest-api-templates/res_stasis_http_resource.c.mustache,
rest-api/api-docs/events.json, res/res_stasis_websocket.c
(removed), include/asterisk/autoconfig.h.in,
rest-api-templates/rest_handler.mustache: No message for rev
393528 found
2013-07-02 22:01 +0000 [r393508] Jason Parker <jparker@digium.com>
* main/manager.c, CHANGES: Add a SystemName field to all AMI
events. This only gets sent out if configured in asterisk.conf
(closes issue ASTERISK-21494)
2013-07-02 21:19 +0000 [r393485-393500] Richard Mudgett <rmudgett@digium.com>
* apps/app_mixmonitor.c: MixMonitor: Minor code cleanup.
* apps/app_mixmonitor.c: MixMonitor: Make
start_mixmonitor_callback() options parameter NULL tolerant. *
Removed some unnecessary code in start_mixmonitor_callback().
* apps/app_mixmonitor.c: MixMonitor: Don't use ast_strdupa() in a
loop.
* apps/app_mixmonitor.c: MixMonitor: Update XML documentation and
CLI "mixmonitor {start|stop|list}" help.
* apps/app_mixmonitor.c: MixMonitor: Fix refleak in
manager_stop_mixmonitor() if could not stop monitoring.
* apps/app_mixmonitor.c: MixMonitor: Remove some unnecessary
channel locking.
* apps/app_mixmonitor.c: Fix MixMonitor b option. The option had
not been converted to use the replacement for
ast_bridged_channel(). One touch mixmonitor now records files
again.
* channels/chan_gtalk.c: Fix chan_gtalk.c compile error.
2013-07-02 20:34 +0000 [r393484] David M. Lee <dlee@digium.com>
* res/res_sip_notify.c: Add pjproject dependency to res_sip_notify
2013-07-02 18:28 +0000 [r393463] Mark Michelson <mmichelson@digium.com>
* include/asterisk/stasis_bridging.h: Remove unused blind transfer
publication structure. I ended up using a bridge blob, so this
structure was unused. Keeping it in the header would just cause
confusion.
2013-07-02 17:20 +0000 [r393442-393449] Kevin Harwell <kharwell@digium.com>
* main/aoc.c, main/manager.c: Stasis - Refactor AOC Events
Refactored the AMI events in AOC onto Stasis-Core. The
ast_aoc_manager_event function now publishes a channel snapshot,
along with a JSON blob describing the advice of charge. A
"to_ami" handler has also been added that converts the channel
snapshot and AOC event data back into the appropriate data
structure for use with AMI. (closes issue ASTERISK-21472)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2643/
* res/res_sip/sip_configuration.c, include/asterisk/res_sip.h,
res/res_sip/sip_distributor.c, res/res_sip/config_auth.c,
res/res_sip.exports.in,
res/res_sip_outbound_authenticator_digest.c,
res/res_sip_authenticator_digest.c, res/res_sip/config_security.c
(added), res/res_sip_acl.c, res/res_sip.c: New SIP Channel
driver: Always Auth Reject If no matching endpoint is found for
the incoming request Asterisk will respond with a 401
Unauthorized (rejecting the request), but will first challenge if
no authorization creditials are given. Changes also included
moving ACL options into a new global 'security' configuration
section in res_sip.conf. (closes issue ASTERISK-21433) Reported
by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2554/
2013-07-02 16:11 +0000 [r393410-393429] Kinsey Moore <kmoore@digium.com>
* main/stasis_bridging.c: Fix transfer AMI event parameter naming
* tests/test_cel.c (added), main/cel.c, include/asterisk/cel.h: Add
CEL unit tests and do some cleanup This adds several unit tests
for CEL functionality and provides the requisite framework for
creating additional unit tests. This also cleans up some
reference leaks that were occurring in Stasis-Core message
callback code. Review: https://reviewboard.asterisk.org/r/2646/
2013-07-02 10:16 +0000 [r393396] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: Fix issue with inability to cancell
call transfer made by on-sceen menus. Reported by: Igor Olhovskiy
........ Merged revisions 393395 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-07-02 08:23 +0000 [r393383] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/scripts/ast_tls_cert: ast_tls_cert: don't recreate
generated files Don't regenrate cat.cfg, ca.crt and ca.key if
they were already created on a previous run. (closes issue
ASTERISK-21932)
2013-07-01 21:28 +0000 [r393364] Kevin Harwell <kharwell@digium.com>
* res/res_sip/sip_configuration.c, include/asterisk/res_sip.h,
res/res_sip/include/res_sip_private.h, res/res_sip/sip_options.c,
res/res_sip.exports.in, res/res_sip_notify.c (added): New SIP
Channel Driver - Add CLI/AMI initiated NOTIFY requests Added the
ability to send unsolicited NOTIFY requests to a particular
endpoint with a configured payload. Added both CLI and AMI
support. For a given endpoint, this module will iterate over all
its contacts sending the appropriate NOTIFY request to each.
(closes issue ASTERISK-21436) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2623/
2013-07-01 21:24 +0000 [r393361] Matthew Jordan <mjordan@digium.com>
* include/asterisk/pbx.h, main/pbx.c, main/manager.c: Prevent crash
during synchronous AMI origination by ref bumping returned
channel The originate APIs allow callers to provide a pointer to
a channel that will point to the originated channel if the
function call succeeds. This is used by AMI to provide channel
information when the originate is performed synchronously.
Unfortunately, if the originate fails in certain ways, the
outbound channel is already disposed of during the dialing
itself. This results in the channel being improperly dereferenced
by the internal originate function in pbx.c. This patch ref bumps
the channel to prevent this from occurring. Callers must now
unlock and unref the channel (which is more in line with general
channel management guidelines anyway). This only affects manager,
as it is the only consumer of this API function that actually
passes in a channel pointer. Review:
https://reviewboard.asterisk.org/r/2617/
2013-07-01 18:56 +0000 [r393326-393332] Jason Parker <jparker@digium.com>
* res/stasis/control.c, res/stasis_http/resource_channels.c,
include/asterisk/stasis_app.h: ARI: Implement channel
hold/unhold. This puts the channel on hold (rather than queueing
a frame from the channel). (closes issue ASTERISK-21619) Review:
https://reviewboard.asterisk.org/r/2647/
* res/stasis_http/resource_channels.c,
res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
res/stasis_http/resource_channels.h,
rest-api/api-docs/channels.json, res/stasis/control.c: ARI:
Implement channel dial. This creates a new outbound channel, and
bridges it to a channel already in the Stasis application.
(closes issue ASTERISK-21620) Review:
https://reviewboard.asterisk.org/r/2634/
2013-07-01 16:01 +0000 [r393309] Jonathan Rose <jrose@digium.com>
* bridges/bridge_builtin_features.c,
include/asterisk/features_config.h, include/asterisk/mixmonitor.h
(added), include/asterisk/channel.h, CHANGES,
main/features_config.c, apps/app_mixmonitor.c,
configs/features.conf.sample, main/mixmonitor.c (added):
bridge_features: Support One touch Monitor/MixMonitor In addition
to porting those features, they now enjoy greater feature parity
with one another. Specifically, AutoMixMon now has a start and
stop message that can be specified with
TOUCH_MIXMONITOR_MESSAGE_START and TOUCH_MIXMONITOR_MESSAGE_STOP.
(closes issue ASTERISK-21553) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2620/
2013-07-01 13:16 +0000 [r393284] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c, apps/app_meetme.c,
include/asterisk/stasis.h, main/core_local.c,
include/asterisk/json.h, channels/chan_gtalk.c,
channels/sig_pri.c, channels/chan_iax2.c, apps/app_queue.c,
CHANGES, main/json.c, channels/chan_dahdi.c,
channels/sig_analog.c, res/res_agi.c, configs/sip.conf.sample,
channels/sip/include/sip.h: Refactor extraneous channel events
This change removes JitterBufStats, ChannelReload, and
ChannelUpdate and refactors the following events to travel over
Stasis-Core: * LocalBridge * DAHDIChannel * AlarmClear *
SpanAlarmClear * Alarm * SpanAlarm * DNDState * MCID *
SIPQualifyPeerDone * SessionTimeout Review:
https://reviewboard.asterisk.org/r/2627/ (closes issue
ASTERISK-21476)
2013-06-29 13:47 +0000 [r393262-393264] Joshua Colp <jcolp@digium.com>
* res/res_sip_pubsub.c: Nothing to see here, move along.
* res/res_sip_pubsub.c, include/asterisk/res_sip_pubsub.h,
res/res_sip_pubsub.exports.in: Implement the defined PUBLISH ESC
API within res_sip_pubsub. (closes issue ASTERISK-21452) Review:
https://reviewboard.asterisk.org/r/2630/
2013-06-29 00:31 +0000 [r393219-393241] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c, include/asterisk/bridging.h: Tweak after bridge
callback reason to string strings.
* main/bridging.c: Fix after bridge callback datastore data memory
leak.
* main/datastore.c: This is no longer needed.
* main/bridging.c: Promote local channel optimizing debug messages
to verbose 3 messages.
2013-06-28 19:22 +0000 [r393190-393197] Jonathan Rose <jrose@digium.com>
* res/parking/parking_applications.c, CHANGES,
res/parking/parking_ui.c, res/parking/res_parking.h,
res/res_parking.c: res_parking: Dynamic Parking Lots (closes
issue ASTERISK-21644) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2615/
* main/features.c, include/asterisk/features.h: features: call
pickup stasis refactoring (issue ASTERISK-21544) Reported by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/2588/
2013-06-28 19:05 +0000 [r393184] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridging_features.h: Fix overlapping enum
ast_bridge_feature_flags. Things may no longer behave in an
unexpected fashion. Local channel optimization to holding bridges
will work again.
2013-06-28 18:42 +0000 [r393182] Mark Michelson <mmichelson@digium.com>
* main/manager.c, bridges/bridge_builtin_features.c,
channels/chan_sip.c, channels/chan_skinny.c,
main/stasis_bridging.c, res/res_sip_refer.c,
include/asterisk/bridging.h, main/manager_bridging.c,
channels/chan_iax2.c, include/asterisk/stasis_bridging.h,
main/bridging.c: Add stasis publications for blind and attended
transfers. This creates stasis messages that are sent during a
blind or attended transfer. The stasis messages also are
converted to AMI events. Review:
https://reviewboard.asterisk.org/r/2619 (closes issue
ASTERISK-21337) Reported by Matt Jordan
2013-06-28 17:31 +0000 [r393164] Matthew Jordan <mjordan@digium.com>
* tests/test_cdr.c, main/cdr.c: Handle an originated channel being
sent into a non-empty bridge Originated channels are a bit odd -
they are technically a dialed channel (thus the party B or peer)
but, since there is no caller, they are treated as the party A.
When entering into a bridge that already contains participants,
the CDR engine - if the CDR record is in the Dial state -
attempts to match the person entering the bridge with an existing
participant. The idea is that if you dialed someone and the
person you dialed is already in the bridge, you don't need a new
CDR record, the existing CDR record describes the relationship.
Unfortunately, for an originated channel, there is no Party B. If
no one was in the bridge this didn't cause any issues; however,
if participants were in the bridge the CDR engine would attempt
to match a non-existant Party B on the channel's CDR record and
explode. This patch fixes that, and a unit test has been added to
cover this case.
2013-06-28 16:23 +0000 [r393144] Jason Parker <jparker@digium.com>
* res/res_stasis_http_channels.c,
res/stasis_http/resource_channels.h,
rest-api/api-docs/channels.json,
res/stasis_http/resource_channels.c: Change ARI originate to also
allow dialing an exten/context/priority. The old way didn't make
much sense, so some of the fields were repurposed. (closes issue
ASTERISK-21658) Review: https://reviewboard.asterisk.org/r/2626/
2013-06-28 15:50 +0000 [r393130] Matthew Jordan <mjordan@digium.com>
* include/asterisk/parking.h, main/asterisk.c, main/bridging.c,
main/cdr.c, include/asterisk/cdr.h: Better handle parking in CDRs
Parking typically occurs when a channel is transferred to a
parking extension. When this occurs, the channel never actually
hits the dialplan if the extension it was transferred to was a
"parking extension", that is, the extension in the first priority
calls the Park application. Instead, the channel is immediately
sent into the holding bridge acting as the parking bridge. This
is problematic. Because we never go out to the dialplan, the CDRs
won't transition properly and the application field will not be
set to "Park". CDRs typically swallow holding bridges, so the CDR
itself won't even be generated. This patch handles this by
pulling out the holding bridge handling into its own CDR state.
CDRs now have an explicit parking state that accounts for this
specific subclass of the holding bridge. In addition, we handle
the parking stasis message to set application specific data on
the CDR such that the last known application for the CDR properly
reflects "Park". This is a bit sad since we're working around the
odd internal implementation of parking that exists in Asterisk
(and that we had to maintain in order to continue to meet some
odd use cases of parking), but at least the code to handle that
is where it belongs: in CDRs as opposed to sprinkled liberally
throughout the codebase. This patch also properly clears the
OUTBOUND channel flag from a channel when it leaves a bridge, and
tweaks up dialing handling to properly compare the correct CDR
with the channel calling/being dialed.
2013-06-28 15:36 +0000 [r393128] Jason Parker <jparker@digium.com>
* res/stasis_http/resource_channels.c: Change some 500 errors to
400.
2013-06-28 02:14 +0000 [r393083-393100] David M. Lee <dlee@digium.com>
* res/res_stasis_http.c: Removed stray apostrophe. Apparently the
pluralization of an acronym does not use an apostophe, according
to most modern style guides. I feel like I've been living a lie
this whole time.
* res/res_stasis_http.c: Removed the automatic 302 redirects for
ARI URL's that end with a slash. There were some problems
redirecting RESTful API requests; notably the client would change
the request method to GET on the redirected requests. After some
looking into, I decided that a 404 would be simpler and have more
consistent behavior.
2013-06-27 21:01 +0000 [r393034-393066] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: Change the name of some local variables in
bridging.c to reflect what they really mean.
* main/config_options.c, include/asterisk/config_options.h: Add
config framework non-empty string validation requirement option.
Add config framework OPT_CHAR_ARRAY_T and OPT_STRINGFIELD_T
non-empty requirement option. There are cases were you don't want
a config option string to be empty. To require the option string
to be non-empty, just set the aco_option_register() flags
parameter to non-zero. * Updated some config framework enum
aco_option_type comments.
2013-06-26 20:59 +0000 [r393005] Jonathan Rose <jrose@digium.com>
* funcs/func_channel.c, include/asterisk/bridging.h,
main/bridging.c: func_channel: Read/Write after_bridge_goto
option Allows reading and setting of a channel's
after_bridge_goto datastore (closes issue ASTERISK-21875)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2628/
2013-06-26 19:29 +0000 [r392987] Jason Parker <jparker@digium.com>
* res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
res/stasis_http/resource_channels.h,
rest-api/api-docs/channels.json, res/stasis/control.c,
res/stasis_http/resource_channels.c: ARI: Add support for
continuing to a different location in dialplan. This allows going
elsewhere in the dialplan, so that the location can be specified
after exiting the Stasis application. (closes issue
ASTERISK-21870) Review: https://reviewboard.asterisk.org/r/2644/
2013-06-26 19:15 +0000 [r392933-392972] Richard Mudgett <rmudgett@digium.com>
* res/res_parking.c: Remove some redundant parking config error
messages.
* main/bridging.c: Fix several problems with
ast_bridge_add_channel(). * Fix locking problems.
ast_bridge_move() locks two bridges. To do that, deadlock
avoidance must be done. Called bridge_move_locked() instead. *
Fix inconsistency in the bridge dissolve check callers. The
original caller has already removed the channel from the bridge.
The new caller has not removed the channel from the bridge.
Reverted bridge_dissolve_check() and added
bridge_dissolve_check_stolen() to be used by the new caller on
the original bridge after the channel is moved to the new bridge.
* Fix memory leak of features if the added channel was already in
a bridge. * Fix incorrect call to ast_bridge_impart(). * Renamed
bridge_chan to yanked_chan.
* channels/chan_sip.c, include/asterisk/bridging.h,
apps/confbridge/conf_chan_announce.c: Fix incorrect calls to
ast_bridge_impart(). There was a misunderstanding about
ast_bridge_impart()'s handling of the imparted channel's
reference. The channel reference is passed by the caller unless
ast_bridge_impart() returns an error. * Fixed a memory leak in
conf_announce_channel_push() if the impart failed.
* main/features.c: AMI Bridge action: Get channel xfer config after
we have found the second channel.
2013-06-25 22:28 +0000 [r392915] Jonathan Rose <jrose@digium.com>
* res/parking/parking_applications.c, CHANGES, main/bridging.c,
res/parking/parking_bridge_features.c,
res/parking/parking_manager.c, include/asterisk/features.h,
res/parking/parking_bridge.c, res/parking/res_parking.h,
main/features.c, res/parking/parking_controller.c: res_parking:
Add Parking manager action to the new parking system (closes
issue ASTERISK-21641) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2573/
2013-06-25 20:25 +0000 [r392898] Jason Parker <jparker@digium.com>
* Makefile: Fix typo with XML docs.
2013-06-25 19:22 +0000 [r392864-392879] Joshua Colp <jcolp@digium.com>
* include/asterisk/sorcery.h: Add a note about being ready to
accept observer invocations before adding an observer.
* res/res_sip/sip_options.c: Move where the sorcery observer is
added for qualify to guarantee the sched_qualifies container
exists.
2013-06-25 13:03 +0000 [r392829] Kinsey Moore <kmoore@digium.com>
* apps/app_queue.c, main/cel.c, apps/app_dial.c,
include/asterisk/stasis_channels.h, include/asterisk/cel.h,
apps/app_celgenuserevent.c, main/stasis_channels.c: CEL
refactoring cleanup This change removes AST_CEL_BRIDGE_UPDATE
since it should no longer be used because masquerade situations
are now accounted for in other ways. This also refactors usage of
AST_CEL_FORWARD to be produced by a Dial message which has been
extended with a "forward" field. (closes issue ASTERISK-21566)
Review: https://reviewboard.asterisk.org/r/2635/
2013-06-25 01:12 +0000 [r392797-392812] Matthew Jordan <mjordan@digium.com>
* main/named_acl.c, res/res_calendar.c, /, channels/chan_motif.c,
main/http.c, main/config_options.c: Fix memory/ref counting leaks
in a variety of locations This patch fixes the following memory
leaks: * http.c: The structure containing the addresses to bind
to was not being deallocated when no longer used * named_acl.c:
The global configuration information was not disposed of *
config_options.c: An invalid read was occurring for certain
option types. * res_calendar.c: The loaded calendars on module
unload were not being properly disposed of. * chan_motif.c: The
format capabilities needed to be disposed of on module unload. In
addition, this now specifies the default options for the
maxpayloads and maxicecandidates in such a way that it doesn't
cause the invalid read in config_options.c to occur. (issue
ASTERISK-21906) Reported by: John Hardin patches: http.patch
uploaded by jhardin (license 6512) named_acl.patch uploaded by
jhardin (license 6512) config_options.patch uploaded by jhardin
(license 6512) res_calendar.patch uploaded by jhardin (license
6512) chan_motif.patch uploaded by jhardin (license 6512)
........ Merged revisions 392810 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/presencestate.c, main/sorcery.c,
res/parking/parking_bridge.c, main/cdr.c, main/manager.c,
main/parking.c, main/devicestate.c, main/cel.c: Fix a variety of
memory leaks This patch addresses the following memory/ref
counting leaks: * main/devicestate.c - unsubscribe and join our
devicestate message subscription * main/cel.c - clean up the
datastore and config objects on exist * main/parking.c - cleanup
memory leak of retriever snapshot on message payload destruction
* res/parking/parking_bridge.c - cleanup memory leak of retrieve
snapshot on message payload destruction * main/presencestate.c -
unsubscribe and join the caching topic on exit * manager.c -
properly unregister the manager action "BlindTransfer" *
sorcery.c - shutdown the threadpool on exit and dispose of any
wizards (issue ASTERISK-21906) Reported by: John Hardin patches:
cel.patch uploaded by jhardin (license #6512) devicestate.patch
uploaded by jhardin (license #6512) manager.patch uploaded by
jardin (license #6512) presencestate.patch uploaded by jhardin
(license #6512) retriever-channel-snapshot.patch uploaded by
jhardin (license #6512) sorcery.patch uploaded by jhardin
(license #6512)
2013-06-24 22:05 +0000 [r392778-392779] David M. Lee <dlee@digium.com>
* tests/test_endpoints.c, tests/test_stasis_endpoints.c: Few more
menuselect fixes missed in r392777
* res/stasis_json/resource_sounds.h,
rest-api-templates/res_stasis_json_resource.c.mustache,
rest-api-templates/res_stasis_http_resource.c.mustache: Fixed
templates so that the changes from r392777 won't be overwritten
the next time we run the generators.
2013-06-24 21:40 +0000 [r392777] Richard Mudgett <rmudgett@digium.com>
* res/res_stasis_http_playback.c, res/res_stasis_playback.c,
res/res_stasis_websocket.c, res/res_stasis_json_recordings.c,
res/res_stasis_http_channels.c, res/res_stasis_json_endpoints.c,
res/res_stasis_json_events.c, res/res_stasis_http_recordings.c,
res/res_stasis_answer.c, res/res_chan_stats.c,
res/res_stasis_http_endpoints.c, res/res_stasis_http_events.c,
res/res_stasis_json_sounds.c, res/res_stasis_bridge_add.c,
res/res_stasis_json_bridges.c, res/res_stasis_http_sounds.c,
res/res_statsd.c, res/res_stasis_http_bridges.c,
res/res_stasis_json_asterisk.c, res/res_stasis_test.c,
res/res_stasis_json_playback.c, res/res_stasis_http.c,
res/res_stasis.c, apps/app_stasis.c,
res/res_stasis_http_asterisk.c, res/res_stasis_json_channels.c:
Fix menuselect display for stasis modules. The menuselect parser
is very simple. It looks for AST_MODULE_INFO and uses any quoted
string on that line as the module summary display.
2013-06-24 19:28 +0000 [r392729-392747] Mark Michelson <mmichelson@digium.com>
* /: Remove stray properties from merge.
* /, main/features_config.c, doc/appdocsxml.dtd: Add documentation
for features configuration. Review:
https://reviewboard.asterisk.org/r/2616 (closes issue
ASTERISK-21542) Reported by Matt Jordan
2013-06-24 13:49 +0000 [r392700] Kinsey Moore <kmoore@digium.com>
* include/asterisk/media_index.h (added), main/file.c, main/http.c,
include/asterisk/format.h, rest-api/api-docs/sounds.json,
include/asterisk/_private.h, main/sounds_index.c (added),
res/res_stasis_http.c, main/asterisk.c, main/media_index.c
(added), include/asterisk/file.h, include/asterisk/http.h,
include/asterisk/sounds_index.h (added),
res/stasis_http/resource_sounds.c: Index installed sounds and
implement ARI sounds queries This adds support for stasis/sounds
and stasis/sounds/{ID} queries via the Asterisk RESTful Interface
(ARI, formerly Stasis-HTTP). The following changes have been made
to accomplish this: * A modular indexer was created for local
media. * A new function to get an ast_format associated with a
file extension was added. * Modifications were made to the
built-in HTTP server so that URI decoding could be deferred to
the URI handler when necessary. * The Stasis-HTTP sounds JSON
documentation was modified to handle cases where multiple
languages are installed in different formats. * Register and
Unregister events for formats were added to the system topic.
(closes issue ASTERISK-21584) (closes issue ASTERISK-21585)
Review: https://reviewboard.asterisk.org/r/2507/
2013-06-23 19:19 +0000 [r392676] Matthew Jordan <mjordan@digium.com>
* res/res_fax.c: Properly pack the parameters into ast_json_pack
when sending a send fax message This patch properly packs the
parameters into the send fax message so that it actually work.
Missing a ',' between two string fields can be difficult to
debug, particularly when the actual packing succeeds.
Interestingly enough, this didn't actually crash until the JSON
blob we deref'd and disposed of. Since that happened in a
different thread, it was pretty tough to track down.
2013-06-23 18:59 +0000 [r392627-392667] Joshua Colp <jcolp@digium.com>
* res/res_sip_outbound_registration.c,
res/res_sip_endpoint_identifier_ip.c, res/res_sip_acl.c: Add some
more missing ast_sorcery_generic_alloc conversions.
* tests/test_sorcery_realtime.c, tests/test_sorcery_astdb.c: Add
missing ast_sorcery_generic_alloc conversions.
* main/manager_endpoints.c: Fix a bug where messages were getting
duplicated on AMI. This was caused by forwarding all endpoint
messages to manager which includes channel messages that are
related to the endpoint. This change causes only the PeerStatus
messages to be forwarded to manager thus eliminating the
duplicate channel messages.
2013-06-22 22:42 +0000 [r392607] Matthew Jordan <mjordan@digium.com>
* res/res_fax.c: Properly extract channel variables for the
SendFAX/ReceiveFAX Stasis messages By the time something extracts
the pointers from ast_json_pack, the channels will already be
disposed of. This patch properly pulls the information out of the
variables and packs them into the JSON blob.
2013-06-22 14:26 +0000 [r392565-392586] Joshua Colp <jcolp@digium.com>
* include/asterisk/sorcery.h, res/res_sip/config_auth.c,
res/res_sip/sip_options.c, res/res_sip/location.c,
tests/test_sorcery.c, main/sorcery.c,
res/res_sip/config_domain_aliases.c,
res/res_sip/config_transport.c, res/res_sip/sip_configuration.c:
Make sorcery details opaque and add extended fields. Sorcery
specific object information is now opaque and allocated with the
object. This means that modules do not need to be recompiled if
the sorcery specific part is changed. It also means that sorcery
can store additional information on objects and ensure it is
freed or the reference count decreased when the object goes away.
To facilitate the above a generic sorcery allocator function has
been added which also ensures that allocated objects do not have
a lock. Extended fields have been added thanks to all of the
above which allows specific fields to be marked as extended, and
thus simply stored as-is within the object. Type safety is *NOT*
enforced on these fields. A consumer of them has to query and
ultimately perform their own safety check. What does this mean?
Extra modules can extend already defined structures without
having to modify them. Tests have also been included to verify
extended field functionality. Review:
https://reviewboard.asterisk.org/r/2585/
* res/res_sip_exten_state.exports.in (added),
res/res_sip_session.exports.in, res/res_sip_sdp_rtp.c,
res/res_sip_messaging.c (added), res/res_sip_caller_id.c,
channels/chan_gulp.c, res/res_sip_session.c,
res/res_sip_exten_state.c (added), res/res_sip/sip_options.c,
res/res_sip_pubsub.exports.in, channels/sip/include/sip.h,
include/asterisk/sdp_srtp.h (added), channels/sip/sdp_crypto.c
(removed), main/pbx.c, channels/sip/srtp.c (removed),
res/res_sip_transport_websocket.c (added), channels/chan_sip.c,
res/res_sip_registrar.c, res/res_sip/sip_distributor.c,
include/asterisk/res_sip_session.h,
include/asterisk/res_sip_exten_state.h (added),
res/res_sip/security_events.c (added),
res/res_sip_registrar_expire.c (added), res/res_sip.c,
res/res_sip_pidf.c (added), include/asterisk/res_sip_pubsub.h,
channels/sip/include/sdp_crypto.h (removed),
res/res_sip/location.c, res/res_sip_outbound_registration.c,
channels/sip/include/srtp.h (removed),
res/res_sip_endpoint_identifier_anonymous.c (added),
res/res_sip_one_touch_record_info.c (added),
res/res_sip_pubsub.c, res/res_sip/config_transport.c,
configs/res_sip.conf.sample, res/res_sip/sip_configuration.c,
res/res_sip_diversion.c (added), res/res_sip_refer.c (added),
include/asterisk/res_sip.h, res/res_sip_dtmf_info.c,
main/sdp_srtp.c (added), res/res_sip/include/res_sip_private.h,
res/res_sip.exports.in: Merge in current pimp_my_sip work,
including: 1. Security events 2. Websocket support 3. Diversion
header + redirecting support 4. An anonymous endpoint identifier
5. Inbound extension state subscription support 6. PIDF notify
generation 7. One touch recording support (special thanks Sean
Bright!) 8. Blind and attended transfer support 9. Automatic
inbound registration expiration 10. SRTP support 11. Media offer
control dialplan function 12. Connected line support 13.
SendText() support 14. Qualify support 15. Inband DTMF detection
16. Call and pickup groups 17. Messaging support Thanks everyone!
Side note: I'm reminded of the song "How Far We've Come" by
Matchbox Twenty.
2013-06-22 13:58 +0000 [r392564] Matthew Jordan <mjordan@digium.com>
* res/res_fax.c: Fix a deadlock and possible crash in res_fax This
patch fixes two bugs. (1) It unlocks the channel in the framehook
handlers before attempting to grab the peer from the bridge. The
locking order for the bridging framework is bridge first, then
channel - having the channel locked while attempting to obtain
the bridge lock causes a locking inversion and a deadlock. This
patch bumps the channel ref count prior to releasing the lock in
the framehook to avoid lifetime issues. Note that this does
expose a subtle problem in framehooks; that is, something could
modify the framehook list while we are executing, causing issues
in the framehook list traversal that the callback executes in.
Fixing this is a much larger problem that is beyond the scope of
this patch - (a) we already unlock the channel in this particular
framehook and we haven't run into a problem yet (as modifying the
framehook list when a channel is about to perform a fax gateway
would be a very odd operation) and (b) migrating to an ao2
container of framehooks would be more invasive at this point. See
the referenced ASTERISK issue for more information. (2) Directly
packing channel variables into a JSON object turned out to be
unsafe. A condition existed where the strings in the JSON blob
were no longer safe to be accessed if the channel object itself
was disposed of. (issue ASTERISK-21951)
2013-06-22 12:40 +0000 [r392538] Joshua Colp <jcolp@digium.com>
* include/asterisk/res_sip.h, main/manager_endpoints.c (added),
include/asterisk/stasis_endpoints.h, channels/chan_iax2.c,
include/asterisk/manager.h, channels/chan_gulp.c,
main/stasis_endpoints.c, res/res_sip.c, main/manager.c,
channels/chan_sip.c, channels/chan_skinny.c,
res/res_sip/sip_configuration.c: Migrate PeerStatus events to
stasis, add stasis endpoints, and add chan_pjsip device state.
(closes issue ASTERISK-21489) (closes issue ASTERISK-21503)
Review: https://reviewboard.asterisk.org/r/2601/
2013-06-21 22:39 +0000 [r392514] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_simple.c, bridges/bridge_softmix.c,
bridges/bridge_native_rtp.c, main/bridging.c,
include/asterisk/bridging_technology.h, bridges/bridge_holding.c,
include/asterisk/bridging.h: Extract a useful routine from the
softmix bridge technology. * Extract a useful routine from the
softmix bridge technology for other technologies. Make other
technologies use it if they can. * Made native and 1-1 bridges
write to all parties if the bridge channel writing the frame into
the bridge is NULL. Softmix will also do the same for frame types
that make sense. * Tweak the bridge write routine return value
meaning and adjust the bridge technologies to match.
2013-06-21 21:22 +0000 [r392489] Matthew Jordan <mjordan@digium.com>
* channels/chan_gulp.c: Add BUGBUG for broken direct media in
chan_gulp (issue ASTERISK-21947)
2013-06-21 18:54 +0000 [r392464] Jason Parker <jparker@digium.com>
* rest-api/api-docs/channels.json: Fix typo.
2013-06-21 18:10 +0000 [r392437] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: Add channel optimization interaction with frame
hooks BUGBUG comments.
2013-06-21 18:05 +0000 [r392436] Mark Michelson <mmichelson@digium.com>
* channels/chan_unistim.c: Change chan_unistim to use core transfer
API. Review: https://reviewboard.asterisk.org/r/2553 (closes
issue ASTERISK-21527) Reported by Matt Jordan
2013-06-21 17:48 +0000 [r392435] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_softmix.c, main/bridging.c,
include/asterisk/bridging_technology.h,
include/asterisk/bridging.h, main/features.c: Change several
bridge functions to return error status. The bridge frame queue
functions need to return an error status if the frame failed to
be queued because of an error condition. The main calls that
needed to return the status are:
ast_bridge_channel_queue_action_data() and
ast_bridge_channel_write_action_data(). The other return changes
are ripple effects.
2013-06-21 14:21 +0000 [r392409] Matthew Jordan <mjordan@digium.com>
* contrib/scripts/autosupport: Update autosupport script This patch
updates the autosupport script to collect all information
available to the Asterisk CLI command "digium_phones". It also
makes minor improvements in options handling. (closes issue
AST-1163) Reported by: Trey Blancher patches:
390347_autosupport.diff uploaded by tblancher (License 5821)
390348_autosupport.diff uploaded by tblancher (License 5821)
2013-06-20 21:13 +0000 [r392364] Joshua Colp <jcolp@digium.com>
* res/res_sip_session.c: Add a log message for when an incoming
session is rejected due to the extension not being found.
2013-06-20 17:21 +0000 [r392335] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridging_features.h, main/features.c,
main/bridging.c, res/parking/parking_bridge_features.c,
apps/confbridge/conf_config_parser.c: Fix potential bridge hook
resource leak if the hook install fails.
2013-06-20 16:29 +0000 [r392318] Mark Michelson <mmichelson@digium.com>
* main/threadpool.c: Fix threadpool rapid growth problem. When a
threadpool is set to autoincrement its threadcount, an issue may
arise when multiple tasks are queued at once into the threadpool.
Since threads start active, each new task would result in
autoincrementing the thread count. So if all threads were active,
and a thread's autoincrement value were 5, then 3 new tasks would
result in 15 threads being created even though the initial
autoincrement was sufficient to handle the number of tasks. This
change introduces three behavior changes: 1) New threads in the
threadpool start idle instead of active. 2) When a threadpool
autoincrements, one thread is activated after the growth. 3) When
a threadpool's size is incremented manually, all added threads
are activated. For a more detailed explanation about the changes,
please see the Review Board link at the bottom of this commit.
Review: https://reviewboard.asterisk.org/r/2629
2013-06-19 22:52 +0000 [r392279] David M. Lee <dlee@digium.com>
* Makefile, main/Makefile: Fix build problem on OS X Mountain Lion
(10.8) For about forever, our build flags for OS X have been
slightly off, but good enough to build and run. Apparently they
aren't good enough any more. Previously, we would compile with
macosx-version-min unset and link with it set. This combination,
using GCC 4.8, on Mountain Lion, would create a bad executable
("Illegal Instruction: 4", or something like that) This patch
consistently sets macosx-version-min for both compiling and
linking, which makes everything happy enough to build and run.
2013-06-19 12:55 +0000 [r392241] Kinsey Moore <kmoore@digium.com>
* include/asterisk/cel.h, main/cel.c: Pull CEL linkedid
manipulation into cel.c This finishes moving all CEL linkedid
tracking entirely within cel.c since that is now possible with
channel snapshots. This also removes another CEL linkedid
manipulation function from cel.h that has already been
internalized and is neither called nor available to link against.
Review: https://reviewboard.asterisk.org/r/2632/
2013-06-19 01:28 +0000 [r392190-392214] Matthew Jordan <mjordan@digium.com>
* funcs/func_cdr.c: Handle variable substitution in dummy variables
When func_cdr is used for variable substitution, there is no
channel name and hence no run-time information available for CDR
variable substitution. In that case, the correct thing to do is
to use the CDR object on the channel passed to the function. This
patch checks to see if the channel passed in has a name - if not,
it uses ast_cdr_format_var instead of ast_cdr_get_var. This
allows CDR backends to continue to use variable substitution in
order to resolve ast_cdr object properties.
* tests/test_substitution.c: Fix the test_substitution test In
r391947, the CDR function was modified such that it will return a
value for the start,answer, and end times if asked. That time
will just be 0 if it hasn't happened yet.
2013-06-18 19:31 +0000 [r392139-392166] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c, include/asterisk/bridging.h: Bridging: Fix crash
on destruction of a partially constructed bridge. * Promoted some
bridge construction debug messages to warnings.
* main/bridging.c: Add some safety cleanup for a failed push into a
bridge.
* main/bridging_basic.c: Remove stub comment on function that is
not a stub.
2013-06-18 14:30 +0000 [r392116] Kinsey Moore <kmoore@digium.com>
* include/asterisk/stasis_bridging.h,
rest-api/api-docs/bridges.json, main/stasis_bridging.c: Fix
bridge snapshot conversion to JSON This makes
ast_bridge_snapshot_to_json conform to the swagger Bridge model
by adding the two fields it required. Review:
https://reviewboard.asterisk.org/r/2583/
2013-06-17 18:58 +0000 [r392076] David M. Lee <dlee@digium.com>
* funcs/func_cdr.c, main/cdr.c: Fix build warnings related to
printf/scanf of tv_usec. The type of tv_usec is suseconds_t. On
Linux, this is usually a long int, but the specification is
actually pretty lax on what it might actually be. And, sadly,
there's no printf/scanf width specifier for suseconds_t. So it
could bit an int or a long, but there's not a great way to tell
which it is. This patch fixes scanf by reading into a long
temporary variable that's then stored into the tv_usec. It fixes
printf by casting the tv_usec to a long first. This patch also
adds some missing width specifiers for some debug statements,
which would cause ".000001" to be displayed at ".1".
2013-06-17 18:37 +0000 [r392053-392073] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, channels/chan_vpb.cc: chan_vpb: Fix compile error
and __ast_channel_alloc() prototype const inconsistency.
* channels/chan_misdn.c: chan_misdn: Fix compile error after CDR
merge.
2013-06-17 16:59 +0000 [r392032] Jason Parker <jparker@digium.com>
* include/asterisk/app.h: Fix a build warning with stasis messages.
2013-06-17 14:40 +0000 [r392004-392005] Matthew Jordan <mjordan@digium.com>
* main/manager_channels.c: Prevent sending a NewExten event after a
Hangup during a stack restore When a channel is originated, its
application is typically set to AppDial2, indicating that it was
a dialed channel through the Dial API. Asterisk during an
originate will perform a stack execute to direct the outgoing
channel to a particular place in the dialplan or application.
When the stack returns, the previous application (AppDial2) is
restored. Unfortunately, in the case of an originated channel,
the stack restore happens after hangup. A stasis message is sent
notifying everyone that the application was restored, and this
causes a NewExten event to go out after the Hangup event,
violating the basic contract consumers have of the channel
lifetime. While we could preclude the message from going out,
restoring the channel's state before it executed the next higher
frame in the stack has to occur, and other places in the code
depend on this behavior. Since we know that channel hung up (it's
a ZOMBIE!), this patch simply checks to see if the channel has
been zombified before sending a NewExten event. Note that this
will fix a number of bouncing tests in the Test Suite. Go tests.
* CHANGES: Restore bad merge on CHANGES The patch for CDRs moved
around a lot of content in CHANGES to try and organize the areas
that were affected. This missed some changes that went in with a
merge and removed some updates - this patch adds them back in.
2013-06-17 12:28 +0000 [r391982] Joshua Colp <jcolp@digium.com>
* main/cdr.c: Fix build warning (which is transmogrified into an
error) with my compiler due to uninitialized variable.
2013-06-17 03:31 +0000 [r391947-391964] Matthew Jordan <mjordan@digium.com>
* addons/cdr_mysql.c: Make cdr_mysql compile again by not directly
setting the run-time CDR object A stray ast_cdr_setvar was missed
in cdr_mysql (silly addons). This has now been refactored to not
set the property, as the property would have been set on a
run-time object that was already dispatched to the backend. The
module simply remembers the value it wanted to set and writes it
to MySQL later in the processing.
* apps/app_forkcdr.c, include/asterisk/stasis_channels.h,
main/test.c, channels/chan_h323.c, main/asterisk.c,
channels/chan_unistim.c, addons/chan_ooh323.c,
include/asterisk/cel.h, apps/app_authenticate.c, cdr/cdr_pgsql.c,
apps/app_followme.c, channels/chan_iax2.c,
res/res_config_sqlite.c, main/stasis.c, cdr/cdr_csv.c,
main/cli.c, main/dial.c, channels/chan_skinny.c,
cel/cel_manager.c, res/res_agi.c, main/stasis_channels.c,
cdr/cdr_odbc.c, tests/test_cdr.c (added), main/bridging_basic.c,
main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c,
UPGRADE.txt, include/asterisk/cdr.h, include/asterisk/channel.h,
res/res_stasis_answer.c, main/cel.c, cdr/cdr_tds.c,
funcs/func_channel.c, funcs/func_cdr.c,
include/asterisk/bridging.h, addons/cdr_mysql.c,
funcs/func_callerid.c, apps/app_cdr.c, include/asterisk/time.h,
cel/cel_radius.c, include/asterisk/stasis_internal.h (added),
include/asterisk/channel_internal.h, main/utils.c,
cdr/cdr_adaptive_odbc.c, cdr/cdr_radius.c, main/channel.c,
main/cdr.c, include/asterisk/test.h, channels/chan_dahdi.c,
main/manager.c, apps/app_osplookup.c, main/features.c,
apps/app_dumpchan.c, main/manager_channels.c, main/bridging.c,
cdr/cdr_custom.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
apps/app_dial.c, main/stasis_cache.c, cdr/cdr_syslog.c,
cel/cel_tds.c, channels/chan_agent.c, apps/app_disa.c,
apps/app_queue.c, CHANGES, res/res_monitor.c: Update Asterisk's
CDRs for the new bridging framework This patch is the initial
push to update Asterisk's CDR engine for the new bridging
framework. This patch guts the existing CDR engine and builds the
new on top of messages coming across Stasis. As changes in
channel state and bridge state are detected, CDRs are built and
dispatched accordingly. This fundamentally changes CDRs in a few
ways. (1) CDRs are now *very* reflective of the actual state of
channels and bridges. This means CDRs track well with what an
actual channel is doing - which is useful in transfer scenarios
(which were previously difficult to pin down). It does, however,
mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk
allowed for CDR applications, channels, and other properties to
be spoofed in parts of the code - this no longer works. (2) CDRs
have defined behavior in multi-party scenarios. This behavior
will not be what everyone wants, but it is a defined behavior and
as such, it is predictable. (3) The CDR manipulation functions
and applications have been overhauled. Major changes have been
made to ResetCDR and ForkCDR in particular. Many of the options
for these two applications no longer made any sense with the new
framework and the (slightly) more immutable nature of CDRs. There
are a plethora of other changes. For a full description of CDR
behavior, see the CDR specification on the Asterisk wiki. (closes
issue ASTERISK-21196) Review:
https://reviewboard.asterisk.org/r/2486/
2013-06-14 23:26 +0000 [r391921] Mark Michelson <mmichelson@digium.com>
* main/app.c: Fix regression in MWI stasis handling. In revision
389733, mwi state allocation was placed into its own function
instead of performing the allocation in-line when required. The
issue was that in ast_publish_mwi_state_full(), the local
variable "uniqueid" was no longer being set, but it was still
being used as the topic for MWI. This meant that all MWI
publications ended up being published to the "" (empty string)
mailbox topic. Thus MWI subscriptions for specific mailboxes were
never notified of mailbox state changes. This change fixes the
issue by removing the local uniqueid variable from
ast_publish_mwi_state_full() and instead referencing the
mwi_state->uniqueid field since it has been properly set. (closes
issue ASTERISK-21913) Reported by Malcolm Davenport
2013-06-14 21:57 +0000 [r391902] Joshua Colp <jcolp@digium.com>
* res/res_sip_registrar.c: Ensure that the number of added contacts
never goes below 0. This can happen when a REGISTER request is
removing a contact. (closes issue ASTERISK-21911) Reported by:
mdavenport
2013-06-14 18:50 +0000 [r391855-391856] Kinsey Moore <kmoore@digium.com>
* main/stasis_bridging.c, include/asterisk/stasis_bridging.h,
rest-api/api-docs/bridges.json: Revert parts of r391855 that were
not ready to go in to trunk
* main/cel.c, include/asterisk/stasis_bridging.h,
rest-api/api-docs/bridges.json, main/stasis_bridging.c: Fix two
more possible crashes in CEL These are locations that should
return valid snapshots, but need to be handled if not.
2013-06-14 16:32 +0000 [r391828] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c, /: app_mixmonitor: Fix crashes caused by
unloading app_mixmonitor Unloading app_mixmonitor while active
mixmonitors were running would cause a segfault. This patch fixes
that by making it impossible to unload app_mixmonitor while
mixmonitors are active. Review:
https://reviewboard.asterisk.org/r/2624/ ........ Merged
revisions 391778 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 391794 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-06-14 16:12 +0000 [r391776-391777] Kinsey Moore <kmoore@digium.com>
* main/cel.c: Fix a crash in CEL bridge snapshot handling Properly
search for bridge association structures so that they are found
when expected and handle cases where they don't exist.
* main/bridging.c: Publish bridge snapshots more often Bridge
snapshot events were missing some important transitions that were
noticed in subsequent snapshots. Snapshots will now be published
on all bridge reconfigurations.
2013-06-13 21:53 +0000 [r391732] Matthew Jordan <mjordan@digium.com>
* utils/check_expr.c, utils/refcounter.c, utils/ael_main.c,
utils/conf2ael.c: Make the utils directory compile... again.
Utils is a source folder that lies, eventually all developers
will cry, "I know I must maintain it, But really with this last
commit I can kiss my software ethics good-bye."
2013-06-13 19:04 +0000 [r391701] Richard Mudgett <rmudgett@digium.com>
* apps/app_confbridge.c, apps/confbridge/conf_config_parser.c, /,
apps/confbridge/include/confbridge.h: app_confbridge: Fix memory
leak on reload. The config framework options should not be
registered multiple times. Instead the configuration just needs
to be reprocessed by the config framework. ........ Merged
revisions 391700 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-06-13 18:26 +0000 [r391699] Mark Michelson <mmichelson@digium.com>
* main/features_config.c: Just return outright on a reload since we
have already processed configuration.
2013-06-13 18:20 +0000 [r391689] Kinsey Moore <kmoore@digium.com>
* main/cel.c: Ensure that Asterisk still starts up when cel.conf is
missing
2013-06-13 18:17 +0000 [r391676] Mark Michelson <mmichelson@digium.com>
* main/features_config.c: Fix memory leak in features_config.c The
options should not be registered multiple times. Instead, the
configuration just needs to be reprocessed by the config
framework. This also exposed that we were not properly telling
the config framework to treat the configuration processing with
the "reload" semantics when a reload occurred. Both of these
errors are fixed now. Thanks to Richard Mudgett for discovering
the leak.
2013-06-13 18:14 +0000 [r391675] Matthew Jordan <mjordan@digium.com>
* main/json.c, main/manager.c, include/asterisk/json.h: Blow away
usage of libjansson's foreach macro While very handy, this macro
didn't occur until a later version of libjansson. We'd prefer to
be compatible with older versions still - as such, iteration over
key/value pairs in a JSON object have to be done with a little
bit more manual work.
2013-06-13 13:46 +0000 [r391622-391643] Kinsey Moore <kmoore@digium.com>
* main/parking.c, include/asterisk/cel.h, main/features.c,
include/asterisk/_private.h, main/cel.c,
include/asterisk/parking.h, main/asterisk.c,
res/parking/parking_manager.c: Refactor CEL bridge events on top
of Stasis-Core This pulls bridge-related CEL event triggers out
of the code in which they were residing and pulls them into cel.c
where they are now triggered by changes in bridge snapshots. To
get access to the Stasis-Core parking topic in cel.c, the
Stasis-Core portions of parking init have been pulled into core
Asterisk init. This also adds a new CEL event
(AST_CEL_BRIDGE_TO_CONF) that indicates a two-party bridge has
transitioned to a multi-party conference. The reverse cannot
occur in CEL terms even though it may occur in actuality and two
party bridges which receive a AST_CEL_BRIDGE_TO_CONF will be
treated as multi-party conferences for the duration of the
bridge. Review: https://reviewboard.asterisk.org/r/2563/ (closes
issue ASTERISK-21564)
* include/asterisk/strings.h, main/cel.c,
include/asterisk/stasis_bridging.h, main/asterisk.c,
main/channel.c, include/asterisk/config_options.h, main/pbx.c,
include/asterisk/stasis_channels.h, main/stasis_bridging.c,
main/config_options.c, main/stasis_channels.c: Refactor CEL
channel events on top of Stasis-Core This uses the channel state
change events from Stasis-Core to determine when channel-related
CEL events should be raised. Those refactored in this patch are:
* AST_CEL_CHANNEL_START * AST_CEL_ANSWER * AST_CEL_APP_START *
AST_CEL_APP_END * AST_CEL_HANGUP * AST_CEL_CHANNEL_END Retirement
of Linked IDs is also refactored. CEL configuration has been
refactored to use the config framework. Note: Some HANGUP events
are not generated correctly because the bridge layer does not
propagate hangupcause/hangupsource information yet. Review:
https://reviewboard.asterisk.org/r/2544/ (closes issue
ASTERISK-21563)
2013-06-13 11:02 +0000 [r391596] Joshua Colp <jcolp@digium.com>
* main/endpoints.c, res/stasis_http/resource_endpoints.c,
main/stasis_cache.c, main/stasis_endpoints.c,
main/channel_internal_api.c, include/asterisk/stasis.h,
include/asterisk/channel.h, include/asterisk/stasis_endpoints.h:
Add support for requiring that all queued messages on a caching
topic have been handled before retrieving from the cache and also
change adding channels to an endpoint to be an immediate
operation. Review: https://reviewboard.asterisk.org/r/2599/
2013-06-12 21:08 +0000 [r391561] David M. Lee <dlee@digium.com>
* res/res_http_websocket.c, /: Fix segfault for certain invalid
WebSocket input. The WebSocket code would allocate, on the stack,
a string large enough to hold a key provided by the client, and
the WEBSOCKET_GUID. If the key is NULL, this causes a segfault.
If the key is too large, it could overflow the stack. This patch
checks the key for NULL and checks the length of the key to avoid
stack smashing nastiness. (closes issue ASTERISK-21825) Reported
by: Alfred Farrugia Tested by: Alfred Farrugia, David M. Lee
Patches: issueA21825_check_if_key_is_sent.patch uploaded by
Walter Doekes (license 5674) ........ Merged revisions 391560
from http://svn.asterisk.org/svn/asterisk/branches/11
2013-06-12 02:29 +0000 [r391479-391521] Matthew Jordan <mjordan@digium.com>
* main/loader.c, main/format.c, /, main/endpoints.c: Fix memory
leak while loading modules, adding formats, and destroying
endpoints This patch fixes three memory leaks * When we load a
module with the LOAD_PRIORITY flag, we remove its entry from the
load order list. Unfortunately, we don't free the memory
associated with entry in the list. This patch corrects that and
properly frees the memory for the module in the list. * When
adding a custom format (such as SILK or CELT), the routine for
adding the format was leaking a reference. RAII_VAR cleans this
up properly. * We now de-ref the channel_snapshot appropriately
when an endpoint is disposed of ........ Merged revisions 391489
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 391507 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/stasis_channels.c, bridges/bridge_native_rtp.c: Fix memory
leaks in stasis_channels and bridge_native_rtp This patch fixes
two memory leaks: * A memory leak in packing channels into a
multi-channel blob payload when publishing dial messages. The
multi-channel blob payload does not steal the references - this
approach was chosen because it works well with the RAII_VAR
macro. Unfortunately, this does mean that you actually have to
use the RAII_VAR macro (or manually deref it yourself) * RTP
instances returned as a result of one of the glue operations are
ref counted and have to be de-ref'd appropriately. We now do
that, as saying that we should do it and then not would be silly.
2013-06-11 22:57 +0000 [r391455] Mark Michelson <mmichelson@digium.com>
* main/bridging.c: Remove incorrect comment about local channel
optimization occurring when performing an attended transfer on an
entire bridge.
2013-06-11 22:21 +0000 [r391430-391453] Jonathan Rose <jrose@digium.com>
* bridges/bridge_native_rtp.c, include/asterisk/framehook.h,
main/framehook.c: bridge_native_rtp: Fix native bridge tech being
incompatible when it should be. When checking compatability for
the native RTP bridge technology there is a race condition
between clearing framehooks that are destroyed when leaving
certain bridges with certain technologies (such as
bridge_native_rtp) and joining bridges with the bridge_native_rtp
technology. Yes, that means a channel in a native RTP bridge
could move to another native RTP bridge and be considered
incompatible with the new native RTP bridge causing it to revert
to a simple bridge technology0. This fixes that bug by ignoring
framehooks that have been marked for destruction when checking
for compatibility with the bridge_native_rtp technology.
* bridges/bridge_native_rtp.c: bridge_native_rtp: Fix possible
segfaults on leaves/joins native_rtp_bridge_get can return any
result from the ast_rtp_glue_result enumerator and the join/leave
functions for bridge_native_rtp seem to assume that if the result
wasn't local that it was remote. Meanwhile forbid can be returned
by that function which can mean certain glue pointers are NULL.
Then when the join/leave functions try to use members of that
pointer, boom. Segfault.
2013-06-11 15:46 +0000 [r391403] David M. Lee <dlee@digium.com>
* main/manager.c, main/stasis_message.c, main/parking.c,
tests/test_stasis_channels.c, include/asterisk/stasis.h,
main/stasis_channels.c, tests/test_stasis.c,
main/manager_channels.c: Add vtable and methods for to_json and
to_ami for Stasis messages When a Stasis message type is defined
in a loadable module, handling those messages for AMI and
res_stasis events can be cumbersome. This patch adds a vtable to
stasis_message_type, with to_ami and to_json virtual functions.
These allow messages to be handled abstractly without putting
module-specific code in core. As an example, the VarSet AMI event
was refactored to use the to_ami virtual function. (closes issue
ASTERISK-21817) Review: https://reviewboard.asterisk.org/r/2579/
2013-06-11 10:24 +0000 [r391380] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: Fix issue with no sound in both way
in case of previous call to chan_unistim phone was canceled.
(related to ASTERISK-20183) ........ Merged revisions 391379 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-06-11 08:13 +0000 [r391335] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_iax2.c, /: IAX2: Transfer Reject: Lock bridgecallno
before touching it, refactor 1). When touching the bridgecallno,
we need to lock it. 2). Remove magic number '0' and replace with
TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce
indentation. Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2613/ ........ Merged
revisions 391333 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 391334 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-06-10 22:38 +0000 [r391314] Matthew Jordan <mjordan@digium.com>
* main/loader.c: Make the reload stasis message bump the ref count
of its sub-object JSON objects are reference stealing. Hence, if
you've RAII_VAR'd some subobject and want to pack it into another
JSON object, you have to bump the reference count. Using the 'O'
option during the pack will bump the reference count for you.
2013-06-10 21:04 +0000 [r391297] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Change chan_skinny to use core transfer
API. Changes for both attended and blind transfers in chan_skinny
to use the new transfer API instead of masquerade. (closes issue
ASTERISK-21526) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2557/
2013-06-10 16:03 +0000 [r391271] Kinsey Moore <kmoore@digium.com>
* res/res_agi.c: Add AGI command arguments to AsyncAGI event This
makes the AGI AsyncAGI event put provided AGI command arguments
in the event's environment. (closes issue ASTERISK-21304)
Patch-By: Dirk Wendland
2013-06-10 15:32 +0000 [r391269] Mark Michelson <mmichelson@digium.com>
* main/features_config.c: Temporary fix for people using sample
features.conf from previous Asterisk versions. People who use the
features.conf.sample file from Asterisk 11 and before in trunk
were given a rude awakening when features configuration changes
were made. Because it uses the config framework and the config
framework is strict about what is accepted and what isn't, people
that had parking options configured found that Asterisk no longer
started. This is because parking options are currently handled in
res_parking.conf instead of features.conf. This fix seeks to
create a temporary band-aid fix for the problem, but having
parking options from the general section be passed to a handler
that will simply print that the option is no longer supported.
This will not cause Asterisk to exit. The fix only applies to
options in the general section. There are two main reasons for
this: 1) The sample features.conf file only has parking options
in the general section. There are no configured parking lots.
Therefore it's not quite as "urgent" to get the parking lot
parsing fixed. 2) The plan is to move parking configuration back
from res_parking.conf to features.conf. When that happens, the
parking lots will also be addressed at that time.
2013-06-10 14:36 +0000 [r391245] Matthew Jordan <mjordan@digium.com>
* apps/app_queue.c, /, configs/queues.conf.sample, UPGRADE.txt: Add
announce-to-first-user option for app_queue In r386792, the
ability to play prompts to the first caller in a call queue was
added. While this is arguably a bug fix for those who expect the
first caller to continue receiving prompts while the agent is
dialed, it has the side effect of preventing the first caller
from hearing the agent immediately upon bridging. This may not be
a problem for those who really want this option, but for those
who didn't care whether or not the first caller in queue heard
their position, it was an issue. This patch disables the ability
for the first caller in the queue to hear prompts and adds a new
option, announce-to-first-user, to queues.conf. Those who the
behavior can enable it by setting this value to True. Note that
if we ever implement the ability to have the prompts be stopped
upon bridging, this option can be removed. (closes issue
ASTERISK-21782) Reported by: Remi Quezada ........ Merged
revisions 391215 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 391241 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-06-10 13:07 +0000 [r391199] Kinsey Moore <kmoore@digium.com>
* res/res_stasis_bridge_add.exports.in (added),
include/asterisk/stasis_app.h,
res/stasis_http/resource_bridges.c, res/stasis/app.h,
res/res_stasis_json_events.c, include/asterisk/stasis_bridging.h,
rest-api/api-docs/bridges.json,
res/stasis_http/resource_bridges.h, res/res_stasis_bridge_add.c
(added), main/stasis_bridging.c,
res/stasis_json/resource_events.h, res/res_stasis.c,
res/res_stasis_json_events.exports.in,
rest-api/api-docs/events.json, res/stasis/control.c,
res/stasis/app.c: Stasis-HTTP: Flesh out bridge-related
capabilities This adds support for Stasis applications to receive
bridge-related messages when the application shows interest in a
given bridge. To supplement this work and test it, this also adds
support for the following bridge-related Stasis-HTTP
functionality: * GET stasis/bridges * GET
stasis/bridges/{bridgeId} * POST stasis/bridges * DELETE
stasis/bridges/{bridgeId} * POST
stasis/bridges/{bridgeId}/addChannel * POST
stasis/bridges/{bridgeId}/removeChannel Review:
https://reviewboard.asterisk.org/r/2572/ (closes issue
ASTERISK-21711) (closes issue ASTERISK-21621) (closes issue
ASTERISK-21622) (closes issue ASTERISK-21623) (closes issue
ASTERISK-21624) (closes issue ASTERISK-21625) (closes issue
ASTERISK-21626)
2013-06-10 09:33 +0000 [r391064-391154] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
unlock bridgecallno ........ Merged revisions 391143 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 391148 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_iax2.c, /: fix bad edit after conflict resolution
........ Merged revisions 391107 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 391111 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_iax2.c, /: IAX2: refactor nativebridge transfer
remove triple checking of iaxs[fr->callno]->transferring reduce
indentation. Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2602/ ........ Merged
revisions 391065 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 391084 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_iax2.c, /: IAX2: fix race condition with
nativebridge transfers. 1). When touching the bridgecallno, we
need to lock it. 2). stop_stuff() which calls
iax2_destroy_helper() Assumes the lock on the pvt is already
held, when iax2_destroy_helper() is called. Thus we need to lock
the bridgecallno pvt before we call
stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating
the state of 'callno->transferring' of the current leg, we can't
change it to READY unless the bridgecallno is locked. Why, if we
are interrupted by the other call leg before 'transferring =
TRANSFER_RELEASED', the interrupt will find that it is READY and
that the bridgecallno is also READY so Releases the legs. (closes
issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2594/ ........ Merged
revisions 391062 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 391063 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-06-09 21:11 +0000 [r391012-391040] Matthew Jordan <mjordan@digium.com>
* main/app.c: Clean up MWI topic pool before message type
destruction Topics need to be disposed of prior to the message
types that are published on them. This includes topic pools. This
prevents an assertion from being raised on shutdown.
* main/manager.c: Only initialize manager_bridging during startup
This moves the initialization call behind the protection against
reloads. We don't want to re-add message router routes during
reloads.
* main/backtrace.c (added), main/logger.c, include/asterisk/lock.h,
main/astmm.c, utils/extconf.c, main/astobj2.c,
include/asterisk/backtrace.h (added), include/asterisk/logger.h:
Add backtrace generation to MALLOC_DEBUG memory corruption
reports This patch allows astmm to access the backtrace
generation code in Asterisk. When memory is allocated, a
backtrace is created and stored with the memory region that
tracks the allocation. If a memory corruption is detected, the
backtrace is printed to the astmm log. The backtrace will make
use of the BETTER_BACKTRACES build option if available. As a
result, this patch moves the backtrace generation code into its
own file and uses the non-wrapped versions of the C library
memory allocation routines. This allows the memory allocation
code to safely use the backtrace generation routines without
infinitely recursing. Review:
https://reviewboard.asterisk.org/r/2567
2013-06-08 06:31 +0000 [r390940-390991] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/bridging_technology.h, main/bridging.c: Add more
support for native bridging. * Added a start technology callback
that technologies can use to start bridging operations. It is
expected that native bridges will find this useful. * Factored
out bridge_channel_complete_join().
* main/bridging.c, include/asterisk/bridging_technology.h,
bridges/bridge_softmix.c: Fix a crash when a bridge switches from
the softmix bridge technology to another. A three party bridge
uses the softmix bridging technology. This technology has a
dedicated thread used to perform the analog mixing. When one of
these parties leaves the bridge, the bridge technology is changed
from the softmix technology to a two-party mixing technology.
Changing technologies is done by removing channels from the old
technology and adding them to the new technology. Since the
remaining channels do not leave the bridge, the softmix mixing
thread could continue to process all channels in the bridge. If
the bridge code is not able to start destruction of the softmix
technology before the softmix mixing thread wakes up, a crash
happens. * Added a stop technology callback that technologies can
use to request any helper threads to stop in preparation for
being destroyed. (closes issue AST-1156) Reported by: John
Bigelow
* include/asterisk/bridging_technology.h: Update some doxygen
comments.
* bridges/bridge_softmix.c: The bridge uniqueid is available for
softmix destructor.
* bridges/bridge_softmix.c: Add some bridge identifiers to some
softmix messages.
2013-06-07 20:51 +0000 [r390920] Jonathan Rose <jrose@digium.com>
* res/parking/parking_devicestate.c (added): res_parking: Add
parking_devicestate.c left out from previous commit (issue
ASTERISK-21645) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2545/
2013-06-07 19:51 +0000 [r390885-390901] Jason Parker <jparker@digium.com>
* configs/queues.conf.sample, CHANGES, apps/app_queue.c,
main/manager.c: Make app_queue AMI events more consistent. Give
Join/Leave more useful names. This also removes the
eventwhencalled and eventmemberstatus configuration options.
These events can just be filtered via manager.conf blacklists.
(closes issue ASTERISK-21469) Review:
https://reviewboard.asterisk.org/r/2586/
* res/res_stasis_http_channels.c,
res/stasis_http/resource_channels.h,
rest-api/api-docs/channels.json,
res/stasis_json/resource_channels.h,
res/stasis_http/resource_channels.c: Implement ARI POST to
/channels, to originate a call. (closes issue ASTERISK-21617)
Review: https://reviewboard.asterisk.org/r/2597/
2013-06-07 16:22 +0000 [r390864] Kinsey Moore <kmoore@digium.com>
* tests/test_devicestate.c: Ensure that all unit tests compile with
the cache clear rework in place
2013-06-07 16:07 +0000 [r390848-390849] Jonathan Rose <jrose@digium.com>
* include/asterisk/pbx.h, CHANGES,
res/parking/parking_bridge_features.c,
res/parking/parking_bridge.c, main/pbx.c,
res/parking/res_parking.h, res/res_parking.c, main/features.c,
res/parking/parking_controller.c: res_parking: Automatically
generate extensions, hints, etc. (closes issue ASTERISK-21645)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2545/
* main/manager.c, apps/app_meetme.c,
apps/confbridge/confbridge_manager.c, include/asterisk/manager.h:
app_meetme: Refactor manager events to use stasis (closes issue
ASTERISK-21467) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2564/
2013-06-07 12:56 +0000 [r390830] Kinsey Moore <kmoore@digium.com>
* main/channel.c, main/stasis_cache.c, include/asterisk/stasis.h,
main/stasis_channels.c, main/endpoints.c, tests/test_stasis.c,
main/bridging.c: Rework stasis cache clear events Stasis cache
clear message payloads now consist of a stasis_message
representative of the message to be cleared from the cache. This
allows multiple parallel caches to coexist and be cleared
properly by the same cache clear message even when keyed on
different fields. This change fixes a bug where multiple cache
clears could be posted for channels. The cache clear is now
produced in the destructor instead of ast_hangup. Additionally,
dummy channels are no longer capable of producing channel
snapshots. Review: https://reviewboard.asterisk.org/r/2596
2013-06-07 01:06 +0000 [r390803-390804] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, main/channel.c,
channels/chan_dahdi.c, channels/chan_misdn.c,
channels/sig_analog.c: Refactor chan_dahdi/sig_analog/sig_pri and
chan_misdn to use the common transfer functions. (closes issue
ASTERISK-21523) Reported by: Matt Jordan (closes issue
ASTERISK-21524) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2600/
* main/features_config.c: Tweak applicationmap and featuregroup
config containers. * Change applicationmap and featuregroup to
replace duplicate config items rather than reject them. * Remove
some unneeded warning messages when getting the applicationmap
allows duplicates from DYNAMIC_FEATURES.
2013-06-06 23:32 +0000 [r390787] Mark Michelson <mmichelson@digium.com>
* main/features_config.c: Conditionally reject duplicate entries in
applicationmap containers. When reading from a config file, it's
important to reject duplicates. Otherwise, featuregroups will
have ambiguity when pointing to applicationmap items. However,
when constructing the channel's current applicationmap, we don't
care about duplicate names since it's the DTMF that identifies a
feature, not the name.
2013-06-06 22:46 +0000 [r390771] Richard Mudgett <rmudgett@digium.com>
* configs/iax.conf.sample, configs/chan_dahdi.conf.sample,
bridges/bridge_builtin_features.c,
include/asterisk/bridging_features.h,
include/asterisk/bridging.h, main/features.c, UPGRADE.txt,
configs/sip.conf.sample, configs/skinny.conf.sample, CHANGES,
main/bridging.c: Reimplement bridging and DTMF features related
channel variables in the bridging core. * The channel variable
ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel driver
specific. If the channel variable is set on the transferrer
channel, the sound will be played to the target of an attended
transfer. * The channel variable BRIDGEPEER becomes a comma
separated list of peers in a multi-party bridge. The BRIDGEPEER
value can have a maximum of 10 peers listed. Any more peers in
the bridge will not be included in the list. BRIDGEPEER is not
valid in holding bridges like parking since those channels do not
talk to each other even though they are in a bridge. * The
channel variable BRIDGEPVTCALLID is only valid for two party
bridges and will contain a value if the BRIDGEPEER's channel
driver supports it. * The channel variable DYNAMIC_PEERNAME is
redundant with BRIDGEPEER and is removed. The more useful
DYNAMIC_WHO_ACTIVATED gives the channel name that activated the
dynamic feature. * The channel variables DYNAMIC_FEATURENAME and
DYNAMIC_WHO_ACTIVATED are set only on the channel executing the
dynamic feature. Executing a dynamic feature on the bridge peer
in a multi-party bridge will execute it on all peers of the
activating channel. (closes issue ASTERISK-21555) Reported by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/2582/
2013-06-06 21:40 +0000 [r390751] Mark Michelson <mmichelson@digium.com>
* channels/sip/include/sip.h, main/bridging.c,
channels/chan_mgcp.c, apps/app_dial.c, channels/chan_unistim.c,
channels/chan_sip.c, include/asterisk/features_config.h (added),
include/asterisk/channel.h, main/features_config.c (added),
include/asterisk/features.h, channels/chan_dahdi.c,
channels/chan_misdn.c, channels/sig_analog.c, main/manager.c,
bridges/bridge_builtin_features.c, main/features.c: Refactor the
features configuration scheme. Features configuration is handled
in its own API in features_config.h and features_config.c. This
way, features configuration is accessible to anything that needs
it. In addition, features configuration has been altered to be
more channel-oriented. Most callers of features API code will be
supplying a channel so that the individual channel's settings
will be acquired rather than the global setting. Missing from
this commit is XML documentation for the features configuration.
That will be handled in a separate commit. Review:
https://reviewboard.asterisk.org/r/2578/ (issue ASTERISK-21542)
2013-06-06 20:50 +0000 [r390733-390734] Richard Mudgett <rmudgett@digium.com>
* main/stasis_message_router.c: Fix compiler warning.
* main/bridging.c, main/features.c, apps/app_bridgewait.c: * Fix a
couple missed hook installs that need
AST_BRIDGE_HOOK_REMOVE_ON_PULL. * Rename some hook flag
parameters to remove_flags.
2013-06-06 20:37 +0000 [r390730] Kinsey Moore <kmoore@digium.com>
* res/res_agi.c: Fix documentation generation Regression from
r390701
2013-06-06 20:32 +0000 [r390729] Jason Parker <jparker@digium.com>
* /: Remove props that people will yell at me for. I'm sorry I
broke automerge. :(
2013-06-06 20:30 +0000 [r390728] Kinsey Moore <kmoore@digium.com>
* res/parking/parking_manager.c: Fix documentation that was in
review during the great suffix/prefix swap
2013-06-06 19:51 +0000 [r390698-390701] Jason Parker <jparker@digium.com>
* CHANGES, /, res/res_agi.c: Split AGI manager events, to remove
SubEvent field. This moves them to stasis, in the process.
(closes issue ASTERISK-21470) Review:
https://reviewboard.asterisk.org/r/2587/
* main/stasis_message_router.c,
include/asterisk/stasis_message_router.h: Convert message_router
routes to ao2. Add support for removal. Review:
https://reviewboard.asterisk.org/r/2591/
2013-06-06 18:21 +0000 [r390669] Jonathan Rose <jrose@digium.com>
* main/bridging.c: Parking: Enable code responsible for
intercepting park exten transfers
2013-06-06 01:52 +0000 [r390612-390639] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Add a BUGBUG note.
* main/bridging.c: Misc core external attended transfer fixes. *
Fix external attended transfer bridge move/swap method. One of
the transferrer channels was not kicked out of the bridge. * Fix
several off-nominal extended attended transfer paths. Mainly the
channels involved needed to be hung up or kicked out of the
bridge.
* main/core_local.c: Make local channels use ast_channel_move()
instead of the inlined version.
2013-06-05 21:14 +0000 [r390584-390585] David M. Lee <dlee@digium.com>
* include/asterisk/stasis.h: Corrected comment on stasis_cache_get
* main/manager_channels.c: Fixed refcounting problems with chanspy
AMI support. The ast_multi_channel_blob_get_channel function does
not bump the refcount on the channel snapshot that it returns.
This is typical for Stasis message payloads, since being
immutable means that the object won't get unreffed out from
underneath you. The manager code for chanspy was unreffing the
snapshots it got out of the multi-channel blob, which was one
unref too many.
2013-06-05 19:19 +0000 [r390510-390550] Mark Michelson <mmichelson@digium.com>
* include/asterisk/bridging_features.h, main/features.c,
bridges/bridge_builtin_interval_features.c, main/bridging.c,
res/parking/parking_bridge_features.c, main/bridging_basic.c:
Remove remaining traces of remove_on_pull from hooks and hook
APIs.
* include/asterisk/bridging_features.h: Give the
AST_BRIDGE_HOOK_REMOVE_ON_PULL a legitimate value.
* include/asterisk/bridging_features.h, main/bridging.c: Change the
remove_on_pull flag on ast_bridge_hook to be a set of flags. This
change is used to make bridge hook removal more generic. This
way, depending on the circumstance, the appropriate bridge hooks
may be removed.
2013-06-05 14:50 +0000 [r390473] Joshua Colp <jcolp@digium.com>
* main/channel.c: Publish the channel state snapshot *before*
calling device state so a device state producer can use an up to
date snapshot.
2013-06-05 14:47 +0000 [r390472] David M. Lee <dlee@digium.com>
* main/channel_internal_api.c: Fixed a consistency problem with
channel snapshot and endpoint state. When channels are added to
an endpoint, the code originally posted a channel snapshot to the
endoint's topic directly. Turns out, this is a bad idea. This
causes the endpoint to see an inconsistent view of the channel,
since it will later receive in-flight messages with old channel
snapshots. This patch instead just publishes channel state
immediately after setting up the forward to the endpoint's topic.
This gives the endpoints a consistent view of the channel's
state.
2013-06-04 22:55 +0000 [r390439-390440] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_native_rtp.c: Add BUGBUG comment.
* bridges/bridge_native_rtp.c: Simple lock, assignment, unlock
sandwich optimization.
2013-06-04 15:55 +0000 [r390352-390398] David M. Lee <dlee@digium.com>
* include/asterisk/manager.h: Corrected the docs on
ast_manager_event_blob_create
* configure.ac, makeopts.in, configure,
include/asterisk/autoconfig.h.in, main/Makefile: Correct autoconf
script for finding UUID support. The library that provides UUID
support varies greatly from system to system. On most Linux
distros, it's in libuuid. On OpenBSD, it's in libe2fs-uuid. On OS
X, it is in libsystem. This patch plays hide-and-seek with UUID
support, looking for it in the three places we know about. It
also corrects the Makefile so that it uses the configured library
name and include path. (closes issue ASTERISK-21816) Reported by:
Brad Latus (snuffy) Tested by: Brad Latus (snuffy)
2013-05-31 19:00 +0000 [r390317] Kinsey Moore <kmoore@digium.com>
* main/pbx.c, apps/app_userevent.c, main/stasis_channels.c:
Refactor code and fix a reference leak Refactor some channel blob
publishing code to use ast_channel_publish_blob now that it is
available and fix a JSON reference leak that was occurring during
varset publishing.
2013-05-31 16:15 +0000 [r390289-390291] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, main/manager.c, main/channel_internal_api.c,
include/asterisk/channel.h: Remove ast_channel_bridge() and
associated code called only by it. * Added some more BUGBUG
notes.
* include/asterisk/stasis_channels.h,
bridges/bridge_builtin_features.c, include/asterisk/bridging.h,
main/stasis_channels.c, main/bridging.c, main/channel.c: Fixup
hold/unhold with attended and blind transfers. * DTMF attended
and blind transfers have hold/unhold behavior restored. *
External attended and blind transfers unhold the transfered party
when the transfer is initiated. * Made prohibit blind
transferring a bridge marked as masquerade only. (ConfBridge
bridges) * Made running an application or playing a file inside a
bridge post the hold/unhold messages if MOH is requested. Review:
https://reviewboard.asterisk.org/r/2574/
2013-05-31 14:36 +0000 [r390268] Jason Parker <jparker@digium.com>
* main/manager.c, include/asterisk/manager.h, main/asterisk.c:
Replace ast_manager_publish_message() with a more useful version.
It's much easier to just create a blob of the message. Convert
some AMI events to use it. Review:
https://reviewboard.asterisk.org/r/2577/
2013-05-31 12:41 +0000 [r390249-390250] Kinsey Moore <kmoore@digium.com>
* apps/confbridge/include/confbridge.h, main/stasis_bridging.c,
apps/confbridge/confbridge_manager.c, apps/app_confbridge.c,
include/asterisk/stasis_bridging.h: Remove remnant of snapshot
blob JSON types Remove usage of the once-mandatory snapshot blob
type field, refactor confbridge stasis messages accordingly, and
remove ast_bridge_blob_json_type(). Review:
https://reviewboard.asterisk.org/r/2575/
* main/stasis_channels.c, include/asterisk/stasis_channels.h: Add
snapshot cache that indexes by channel name This adds a new
channel snapshot cache in parallel to the existing cache; the
difference being that it indexes the channel snapshots by channel
name instead of channel uniqueid. Review:
https://reviewboard.asterisk.org/r/2576
2013-05-31 10:42 +0000 [r390230] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, /: Multiple revisions 390228-390229
........ r390228 | may | 2013-05-31 14:19:52 +0400 (Fri, 31 May
2013) | 14 lines reject call attempts when gatekeeper is
configured but not registered (closes issue ASTERISK-21800)
Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
Tested by: Dmitry Melekhov ........ Merged revisions 390181 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 390223 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ r390229
| may | 2013-05-31 14:34:20 +0400 (Fri, 31 May 2013) | 4 lines
remove unnecessary declarations (issue ASTERISK-21800) ........
Merged revisions 390228-390229 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-31 07:57 +0000 [r390180] Walter Doekes <walter+asterisk@wjd.nu>
* Makefile: Let find do its own globbing. Previously a stray .c
file would cause xmldocs to not get built.
2013-05-30 19:23 +0000 [r390122-390154] David M. Lee <dlee@digium.com>
* main/app.c: Missed a line from a bad merge in r390122
* main/stasis_cache.c, include/asterisk.h, main/security_events.c,
include/asterisk/stasis.h, main/devicestate.c, main/named_acl.c,
include/asterisk/stasis_bridging.h, main/presencestate.c,
main/stasis.c, main/channel.c,
include/asterisk/stasis_channels.h, main/stasis_bridging.c,
main/test.c, main/app.c, main/stasis_channels.c,
include/asterisk/security_events.h, main/asterisk.c,
main/bridging.c: Avoid unnecessary cleanups during immediate
shutdown This patch addresses issues during immediate shutdowns,
where modules are not unloaded, but Asterisk atexit handlers are
run. In the typical case, this usually isn't a big deal. But the
introduction of the Stasis message bus makes it much more likely
for asynchronous activity to be happening off in some thread
during shutdown. During an immediate shutdown, Asterisk skips
unloading modules. But while it is processing the atexit
handlers, there is a window of time where some of the core
message types have been cleaned up, but the message bus is still
running. Specifically, it's still running module subscriptions
that might be using the core message types. If a message is
received by that subscription in that window, it will attempt to
use a message type that has been cleaned up. To solve this
problem, this patch introduces ast_register_cleanup(). This
function operates identically to ast_register_atexit(), except
that cleanup calls are not invoked on an immediate shutdown. All
of the core message type and topic cleanup was moved from atexit
handlers to cleanup handlers. This ensures that core type and
topic cleanup only happens if the modules that used them are
first unloaded. This patch also changes the ast_assert() when
accessing a cleaned up or uninitialized message type to an error
log message. Message type functions are actually NULL safe across
the board, so the assert was a bit heavy handed. Especially for
anyone with DO_CRASH enabled. Review:
https://reviewboard.asterisk.org/r/2562/
2013-05-29 20:24 +0000 [r390068] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Fix segfault when dealing with chan_agent
channels. Check the returned bridged pointer for NULL to avoid a
crash. It looks like chan_agent is returning a NULL pointer when
it probably should be returning a pointer to the channel the
Agent channel is pretending to be. (closes issue ASTERISK-21793)
Reported by: Rodrigo P. Telles Patches:
jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Rodrigo P. Telles ........ Merged revisions
390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 390047 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-29 19:54 +0000 [r390042] Jason Parker <jparker@digium.com>
* main/channel.c: Remove unused RAII vars.
2013-05-29 03:22 +0000 [r389990] Matthew Jordan <mjordan@digium.com>
* res/res_fax.c: Pack the right number of items into the status and
receive fax blobs The code was still attempting to pack an
additional item into the blobs that didn't exist. Crashes ensued.
This patch modifies the publishing of these messages so that the
correct number of items are packed in the JSON.
2013-05-29 02:26 +0000 [r389974] Kinsey Moore <kmoore@digium.com>
* res/res_musiconhold.c, res/res_monitor.c,
include/asterisk/stasis_channels.h, res/res_fax.c,
apps/app_fax.c, main/stasis_channels.c: Resolve a merge conflict
When ast_channel_cached_blob_create was merged,
ast_channel_blob_create_from_cache was partially removed in an
unresolved merge conflict. This restores
ast_channel_blob_create_from_cache and refactors usage of
ast_channel_cached_blob_create (requires an ast_channel) to use
ast_channel_blob_create_from_cache (requires a channel uniqueid)
instead.
2013-05-28 17:47 +0000 [r389897] Jonathan Rose <jrose@digium.com>
* /, main/slinfactory.c: Fix a memory copying bug in slinfactory
which was causing mixmonitor issues. Reported by: Michael Walton
Tested by: Jonathan Rose Patches:
slinfactory.c.ASTERISK-21799.patch uploaded by Michael Walton
(license 6502) (closes issue ASTERISK-21799) ........ Merged
revisions 389895 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 389896 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-28 15:54 +0000 [r389848-389870] Mark Michelson <mmichelson@digium.com>
* main/bridging.c: Add missing NULL check to acquire_bridge()
function.
* channels/chan_sip.c, channels/sip/include/sip.h: Add attended
transfer support for chan_sip.c This now uses the core API for
performing attended transfers. Review
https://reviewboard.asterisk.org/r/2513 (Closes issue
ASTERISK-21520) reported by Matt Jordan
* main/channel.c, main/pbx.c, bridges/bridge_builtin_features.c,
channels/chan_sip.c, apps/confbridge/confbridge_manager.c,
include/asterisk/bridging.h, main/features.c,
include/asterisk/channel.h, CHANGES, main/bridging.c,
channels/chan_mgcp.c: Adds support for a core attended transfer
function plus adds some hiding of masquerades. The attended
transfer API call can complete the attended transfer in a number
of ways depending on the current bridged states of the channels
involved. The hiding of masquerades is done in some
bridging-related functions, such as the manager Bridge action and
the Bridge dialplan application. In addition, call pickup was
edited to "move" a channel rather than masquerade it. Review:
https://reviewboard.asterisk.org/r/2511 (closes issue
ASTERISK-21334) Reported by Matt Jordan (closes issue
Asterisk-21336) Reported by Matt Jordan
2013-05-27 01:33 +0000 [r389770-389827] Matthew Jordan <mjordan@digium.com>
* res/res_fax.c, res/res_fax_spandsp.c: Fix some more fax test
errors due to needing the peer in a bridge In r389799, a number
of fax errors in gateway mode were fixed by using the appropriate
function to get a channel's peer while in a bridge. This patch
does two things: (1) It uses the same function in res_fax_spandsp
while starting the fax gateway. Without this, the fax gateway
will not actually start up, as res_fax_spandsp also must inspect
the channel's peer in a two-party bridge (2) It refactors some
ao2 objects in sendfax_exec to use RAII_VAR. This was reverted in
r389799 as some off nominal paths were getting hit without the
fix in (1) that indicated an ao2 object issue; this turned out to
be a red herring (which is an odd phrase)
* main/stasis_endpoints.c: Initialize the message type before the
topic Caching topics will during initialization attempt to
reference their message type. The message type therefore has to
be initialized prior to the topic to prevent the dreaded
assertion.
* res/res_fax.c: Fix a few fax gateway failures Fax gateway
requires knowledge of a channel's peer in a bridge. This patch
now uses the supported mechanisms to get this information. This
is acceptable for a few reasons: * Fax gateway can only ever work
in a 2-party bridge * Fax gateway cannot work when not in a
bridge * Fax gateway cannot work without knowledge of the
capabilities of both channels in the fax operation (it is, after
all, a gateway)
* main/asterisk.c, res/res_fax.c, main/devicestate.c: Fix a variety
of memory corruption/assertion errors * Initialize a Stasis-Core
message type prior to initializing a caching topic. The caching
topic will attempt to use the message type. * Don't attempt to
publish Stasis-Core messages from remote console connections.
They aren't the main process; they shouldn't attempt to behave as
it (they also don't have the infrastructure to do so) * Don't
treat a JSON object as an ao2 object (whoops) * In asterisk.c,
ref bump the JSON even package that is distributed with the event
meta data. The callers assume that they own the reference, and
the packing routine steals references.
* main/asterisk.c: Restore initialization of security topics During
a merge the security topic initialization got blown away. This
patch restores it.
2013-05-24 21:23 +0000 [r389746-389748] Jason Parker <jparker@digium.com>
* /: grr, props.
* channels/chan_h323.c, main/stasis_channels.c,
main/manager_channels.c, channels/chan_mgcp.c,
channels/chan_unistim.c, /, channels/chan_sip.c,
include/asterisk/channel.h, channels/sig_pri.c,
channels/chan_iax2.c, CHANGES, res/res_sip_sdp_rtp.c,
main/channel.c, channels/chan_dahdi.c,
include/asterisk/stasis_channels.h, channels/sig_analog.c,
channels/chan_misdn.c, channels/chan_skinny.c,
channels/chan_motif.c: Split Hold event into Hold/Unhold, and
move it into core. (closes issue ASTERISK-21487) Review:
https://reviewboard.asterisk.org/r/2565/
2013-05-24 21:01 +0000 [r389738] Kinsey Moore <kmoore@digium.com>
* res/res_stasis.c: Remove a junk define BLOB_HANDLER_BUCKETS is a
remnant of using "type" fields in JSON/snapshot blobs and is no
longer used.
2013-05-24 20:44 +0000 [r389680-389733] Matthew Jordan <mjordan@digium.com>
* include/asterisk/_private.h, include/asterisk/manager.h,
channels/sig_pri.c, CHANGES, res/res_monitor.c,
include/asterisk/app.h, main/json.c,
include/asterisk/stasis_channels.h, apps/app_chanspy.c,
res/parking/parking_manager.c, main/asterisk.c,
main/manager_mwi.c (added), apps/app_voicemail.c,
channels/chan_unistim.c, include/asterisk/json.h,
res/res_musiconhold.c, res/res_xmpp.c, channels/chan_iax2.c,
res/res_jabber.c, main/enum.c, main/loader.c, main/cli.c,
main/cdr.c, channels/chan_dahdi.c, main/manager.c,
channels/chan_skinny.c, apps/app_minivm.c, main/app.c,
main/stasis_channels.c, main/manager_channels.c,
res/res_sip_mwi.c, channels/chan_mgcp.c, main/pbx.c,
main/dnsmgr.c, channels/chan_sip.c, res/res_fax.c,
apps/app_fax.c: Migrate a large number of AMI events over to
Stasis-Core This patch moves a number of AMI events over to the
Stasis-Core message bus. This includes: * ChanSpyStart/Stop *
MonitorStart/Stop * MusicOnHoldStart/Stop * FullyBooted/Reload *
All Voicemail/MWI related events In addition, it adds some
Stasis-Core and AMI support for generic AMI messages, refactors
the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI
message types and topics to be more name compliant. Review:
https://reviewboard.asterisk.org/r/2532 (closes issue
ASTERISK-21462)
* /, main/logger.c: Print all logger messages on shutdown When
Asterisk shuts down and shuts down the loggin gsubsystem, any
messages currently in flight will not get logged. This patch
prevents the loop writing messages from breaking out prematurely,
such that all of the messages are logged. (closes issue
ASTERISK-21716) Reported by: Corey Farrell patches:
logger-process-all-messages.patch uploaded by Corey Farrell
(license 5909) ........ Merged revisions 389676 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 389677 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-24 10:23 +0000 [r389663] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: Fix several problems caused by
multiple line usage with i2004 phones. Reported by: Daniel
Bohling, MihaiMircea (closes issue ASTERISK-21061) (closes issue
ASTERISK-21120) ........ Merged revisions 389661 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-23 21:46 +0000 [r389639] David M. Lee <dlee@digium.com>
* res/res_stasis_playback.c, res/stasis_http/resource_channels.c,
include/asterisk/stasis_http.h, res/res_stasis_http.c:
stasis-http: Provide a response body for 201 created responses
2013-05-23 21:11 +0000 [r389618-389623] Jonathan Rose <jrose@digium.com>
* res/parking/parking_bridge.c: res_parking: Add a verbose message
when a channel is parked
* res/parking/parking_bridge.c: res_parking: Fix some simple bugs
Both of them are covered in the dynamic parking review on
https://reviewboard.asterisk.org/r/2550 - Remove unref against
parking lot that the bridge did on dissolve since the reference
wasn't taken in the first place. On a swap, reapply bridge roles
in order to get music on hold and such playing on the channel
that swaps into the bridge.
2013-05-23 20:25 +0000 [r389609] Joshua Colp <jcolp@digium.com>
* res/res_sip_session.c: Fix a crash due to the INVITE session
being destroyed before the session. This change ensures that the
INVITE session remains valid for the lifetime of the session
object itself by increasing the session count on the dialog that
the INVITE session is allocated from. Once this reaches zero
(normally as a result of decrementing it within the session
destructor) the dialog, and INVITE session, are destroyed.
2013-05-23 20:21 +0000 [r389587-389603] David M. Lee <dlee@digium.com>
* include/asterisk/stasis_app_playback.h,
res/stasis_http/resource_playback.c, include/asterisk/app.h,
res/res_stasis_playback.c, res/stasis/control.c,
res/stasis_http/resource_channels.c,
rest-api/api-docs/playback.json, res/res_stasis_http_channels.c,
include/asterisk/stasis_app.h, main/app.c,
include/asterisk/channel.h, res/stasis_http/resource_channels.h,
rest-api/api-docs/channels.json: This patch adds support for
controlling a playback operation from the Asterisk REST
interface. This adds the /playback/{playbackId}/control resource,
which may be POSTed to to pause, unpause, reverse, forward or
restart the media playback. Attempts to control a playback that
is not currently playing will either return a 404 Not Found
(because the playback object no longer exists) or a 409 Conflict
(because the playback object is still in the queue to be played).
This patch also adds skipms and offsetms parameters to the
/channels/{channelId}/play resource. (closes issue
ASTERISK-21587) Review: https://reviewboard.asterisk.org/r/2559
* res/res_stasis_json_events.exports.in, res/res_stasis_playback.c
(added), rest-api/api-docs/events.json, res/stasis/control.c,
main/channel_internal_api.c, include/asterisk/stasis_http.h,
res/res_stasis_http_channels.c, res/res_stasis_json_events.c,
include/asterisk/stasis_app_playback.h (added),
res/stasis_http/resource_playback.c, include/asterisk/app.h,
include/asterisk/stasis_channels.h,
res/stasis_json/resource_channels.h,
res/stasis_http/resource_channels.c,
res/stasis_http/resource_channels.h, main/stasis_channels.c,
rest-api/api-docs/channels.json,
res/res_stasis_playback.exports.in (added),
res/res_stasis_http.c, res/stasis_json/resource_events.h: This
patch implements the REST API's for POST
/channels/{channelId}/play and GET /playback/{playbackId}. This
allows an external application to initiate playback of a sound on
a channel while the channel is in the Stasis application. /play
commands are issued asynchronously, and return immediately with
the URL of the associated /playback resource. Playback commands
queue up, playing in succession. The /playback resource shows the
state of a playback operation as enqueued, playing or complete.
(Although the operation will only be in the 'complete' state for
a very short time, since it is almost immediately freed up).
(closes issue ASTERISK-21283) (closes issue ASTERISK-21586)
Review: https://reviewboard.asterisk.org/r/2531/
2013-05-23 18:40 +0000 [r389569] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix inverted test preventing DTMF disconnect
from working.
2013-05-23 18:39 +0000 [r389551-389568] Joshua Colp <jcolp@digium.com>
* res/res_sip_sdp_rtp.c: Fix a bug where the DTMF mode was not set
on newly created RTP instances in the res_sip_sdp_rtp module.
* res/res_sip_sdp_rtp.c: Fix a bug with applying the end result of
the codec negotiation to the Asterisk channel.
* res/res_sip_session.c: Fix a bug where the codec order as
configured was not being obeyed.
2013-05-22 19:15 +0000 [r389519] David M. Lee <dlee@digium.com>
* main/app.c: Fixed startup race condition which caused occasional
stasis_mwi_state_type assertions. The caching topic (which refers
to the message type) was created before the message type. If the
initial subscription message gets processed before the type can
be initialized, the assertion about using an uninitialized type
fires.
2013-05-22 18:20 +0000 [r389492-389505] Jason Parker <jparker@digium.com>
* /: Remove bad props, before anybody notices.
* /, include/asterisk/dial.h, apps/app_followme.c,
apps/app_queue.c, apps/app_dial.c, main/dial.c: Add dial events
to app_queue and app_followme. Also fixes an issue in app_dial,
where the channels were swapped on dial events. (closes issue
ASTERISK-21551) (closes issue ASTERISK-21550) Review:
https://reviewboard.asterisk.org/r/2549/
2013-05-21 22:49 +0000 [r389454] David M. Lee <dlee@digium.com>
* main/stasis_bridging.c: Fix destruction order assert for
stasis_bridging
2013-05-21 21:08 +0000 [r389426] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Conditional out more app_queue logging that
needs to be reworked. Fixes crash because app_queue was
unconditionally freeing a datastore that was still on a channel.
2013-05-21 18:45 +0000 [r389402] Matthew Jordan <mjordan@digium.com>
* apps/app_confbridge.c, apps/confbridge/confbridge_manager.c:
Raise the ConfBridgeMute/Unmute events when a CLI or AMI action
triggers the change New in 12 are the ConfBridgeMute/Unmute
events, which are triggered when a user changes their mute/unmute
state. This was typically triggered when a user hit a DTMF key
that triggered the mute/unmute menu handler. Forgotten in this is
when an AMI action or CLI command triggers the mute/unmute. This
patch now raises the events in those situations as well. (closes
issue ASTERISK-21802) Reported by: Birger "WIMPy" Harzenetter
2013-05-21 18:00 +0000 [r389378] Richard Mudgett <rmudgett@digium.com>
* rest-api-templates/res_stasis_json_resource.c.mustache,
include/asterisk/frame.h, apps/app_mixmonitor.c,
include/asterisk/parking.h (added), channels/chan_mgcp.c,
main/bridging_roles.c (added), main/pbx.c, main/strings.c,
rest-api/api-docs/events.json, include/asterisk/core_local.h
(added), configs/res_parking.conf.sample (added),
channels/chan_bridge.c (removed),
res/parking/parking_controller.c,
res/parking/parking_applications.c, include/asterisk/channel.h,
include/asterisk/manager.h, apps/app_queue.c,
include/asterisk/stasis_bridging.h (added),
include/asterisk/framehook.h, include/asterisk/config_options.h,
bridges/bridge_builtin_features.c,
apps/confbridge/confbridge_manager.c (added), main/features.c,
apps/app_dumpchan.c, channels/chan_motif.c, channels/chan_h323.c,
apps/app_confbridge.c, include/asterisk/rtp_engine.h,
apps/app_chanspy.c, include/asterisk/ccss.h,
main/manager_channels.c, main/bridging.c,
apps/confbridge/conf_chan_announce.c (added),
main/bridging_basic.c (added), include/asterisk/core_unreal.h
(added), apps/app_dial.c, res/res_stasis_json_events.exports.in,
addons/chan_ooh323.c, main/frame.c, main/parking.c (added),
bridges/bridge_holding.c (added), bridges/bridge_simple.c,
bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
res/Makefile, res/res_stasis_json_events.c, main/core_local.c
(added), CHANGES, channels/chan_iax2.c,
bridges/bridge_multiplexed.c (removed),
res/parking/parking_bridge_features.c,
include/asterisk/abstract_jb.h, channels/chan_gulp.c,
apps/confbridge/conf_config_parser.c, main/channel.c,
res/res_parking.c (added), main/manager.c, main/stasis_bridging.c
(added), res/parking (added),
bridges/bridge_builtin_interval_features.c (added),
rest-api-templates/stasis_json_resource.h.mustache,
main/config_options.c, res/stasis_json/resource_events.h,
main/asterisk.c, res/parking/parking_manager.c,
apps/app_parkandannounce.c (removed), channels/chan_unistim.c,
res/parking/parking_ui.c, channels/chan_local.c (removed),
main/rtp_engine.c, apps/confbridge/conf_chan_record.c (added),
main/core_unreal.c (added), apps/app_bridgewait.c (added),
apps/app_followme.c, configs/features.conf.sample,
channels/chan_jingle.c, channels/chan_dahdi.c,
apps/app_channelredirect.c, funcs/func_channel.c,
main/abstract_jb.c, main/manager_bridging.c (added),
include/asterisk/bridging_roles.h (added), channels/chan_vpb.cc,
channels/chan_sip.c, main/channel_internal_api.c,
channels/chan_agent.c, UPGRADE.txt, include/asterisk/_private.h,
res/parking/parking_bridge.c, main/cli.c,
res/parking/res_parking.h,
include/asterisk/bridging_technology.h, channels/chan_misdn.c,
apps/confbridge/include/confbridge.h, channels/chan_skinny.c,
include/asterisk/bridging_features.h, funcs/func_frame_trace.c,
include/asterisk/bridging.h, include/asterisk/bridging_basic.h
(added), bridges/bridge_native_rtp.c (added): Merge in the
bridge_construction branch to make the system use the Bridging
API. Breaks many things until they can be reworked. A partial
list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native
bridging app_queue COLP updates DTMF attended transfers Protocol
attended transfers
2013-05-21 14:17 +0000 [r389343] David M. Lee <dlee@digium.com>
* apps/app_userevent.c, main/stasis_channels.c: Fixed some extra
field assertion when the event WebSocket is connected
2013-05-20 19:24 +0000 [r389306] Matthew Jordan <mjordan@digium.com>
* main/pbx.c: Set the AST_CDR_FLAG_ORIGINATED flag on originated
channel's CDRs This may alleviate some of the CDR woes with
originated channels, as CDRs do like to know when a channel was
originated. Eventually this will get converted to be a channel
flag, so its location is still good to know post the great CDR
shakeup of 2013.
2013-05-20 18:03 +0000 [r389247-389251] Richard Mudgett <rmudgett@digium.com>
* res/stasis_http/resource_recordings.c, cel/cel_sqlite3_custom.c,
main/event.c, funcs/func_iconv.c,
res/stasis_http/resource_recordings.h,
res/stasis_http/resource_events.c,
res/res_stasis_http_asterisk.c, main/udptl.c,
res/res_stasis_websocket.c, res/stasis_http/resource_events.h,
tests/test_gosub.c, main/threadstorage.c, cel/cel_tds.c,
tests/test_dlinklists.c, res/res_stasis_http_endpoints.c,
res/stasis_http/resource_asterisk.c, res/ael/pval.c, main/json.c,
res/stasis_http/resource_asterisk.h, res/ael/ael_lex.c,
res/res_stasis_http_bridges.c, tests/test_stasis_http.c,
tests/test_stasis.c, res/res_clioriginate.c, cel/cel_pgsql.c,
tests/test_res_stasis.c, res/res_stasis_http_channels.c,
res/res_srtp.c, main/stasis.c, main/stasis_message.c,
main/stasis_message_router.c, main/hashtab.c, res/ael/ael.tab.c,
cel/cel_manager.c, funcs/func_odbc.c,
res/stasis_http/resource_channels.c, funcs/func_channel.c,
res/ael/ael.tab.h, res/stasis_http/resource_channels.h,
utils/ael_main.c, formats/format_h264.c, codecs/codec_dahdi.c,
contrib/utils/eagi_proxy.c, res/res_stasis.c,
main/manager_channels.c, tests/test_json.c, cel/cel_radius.c,
main/stasis_cache.c, tests/test_astobj2_thrash.c,
funcs/func_dialgroup.c, tests/test_xml_escape.c, pbx/pbx_lua.c,
res/res_ael_share.c, res/res_pktccops.c, funcs/func_realtime.c,
cel/cel_odbc.c, res/res_smdi.c, cel/cel_custom.c, res/res_curl.c,
res/res_stasis_http.c, res/stasis_http/resource_endpoints.c,
utils/refcounter.c, res/stasis_http/resource_endpoints.h,
funcs/func_rand.c, funcs/func_version.c, main/sha1.c,
tests/test_hashtab_thrash.c, res/stasis_http/resource_bridges.c,
res/res_stasis_http_recordings.c, main/cel.c,
res/stasis_http/resource_bridges.h, res/res_stasis_http_events.c,
tests/test_time.c: Fixup svn:keywords in all *.c and *.h files.
* channels/sip/include/globals.h, apps/app_celgenuserevent.c,
channels/sip/dialplan_functions.c, include/asterisk/pktccops.h,
channels/sip/include/sdp_crypto.h,
include/asterisk/ael_structs.h, include/asterisk/udptl.h,
channels/sip/include/srtp.h, include/asterisk/frame_defs.h,
apps/app_stasis.c, include/asterisk/sha1.h,
include/asterisk/smdi.h, include/asterisk/stringfields.h,
channels/sip/sdp_crypto.c, channels/sip/include/dialog.h,
include/asterisk/res_srtp.h, channels/sip/srtp.c,
include/asterisk/cel.h, include/asterisk/stasis_http.h,
include/asterisk/stasis_app.h, include/asterisk/stasis.h,
apps/app_morsecode.c, apps/app_waituntil.c,
include/asterisk/json.h,
include/asterisk/stasis_message_router.h,
include/asterisk/hashtab.h,
channels/sip/include/dialplan_functions.h,
include/asterisk/paths.h, include/asterisk/event.h,
apps/app_setcallerid.c, include/asterisk/event_defs.h: Fixup
svn:keywords in all *.c and *.h files.
2013-05-20 17:44 +0000 [r389246] Jason Parker <jparker@digium.com>
* /: Add doxygen.log to svn:ignore property. ........ Merged
revisions 389244 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 389245 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-20 14:21 +0000 [r389217] Kinsey Moore <kmoore@digium.com>
* res/res_stasis_answer.exports.in (added): Add missing exports
file This exposes stasis_app_control_answer and allows
res_stasis_http_channels to load properly.
2013-05-20 14:02 +0000 [r389204] Joshua Colp <jcolp@digium.com>
* main/sorcery.c: In Sorcery pass the name of the object being
allocated to the allocator.
2013-05-20 13:45 +0000 [r389202] Kinsey Moore <kmoore@digium.com>
* apps/confbridge/conf_config_parser.c: Add documentation for
record_file_append When this option was added, it was noted in
CHANGES, but was missing the XML documentation that this patch
adds. (closes issue ASTERISK-21780) Patch-by: Brad Latus (snuffy)
2013-05-19 20:52 +0000 [r389180] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, addons/chan_ooh323.h: add
ast_publish_channel_state according new event framework
2013-05-19 19:45 +0000 [r389164] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Add transfer softkey to ringout state to
enable blond transfers. (closes issue ASTERISK-21327) Reported
by: wedhorn Tested by: myself Patches: skinny-blindxfer01.diff
uploaded by wedhorn (license 5019)
2013-05-19 17:45 +0000 [r389148] Kinsey Moore <kmoore@digium.com>
* res/res_sip_acl.c, res/res_sip.c,
res/res_sip_outbound_registration.c,
res/res_sip_endpoint_identifier_ip.c: Add base XML documentation
for res_sip Thanks to Brad Latus, this patch adds a significant
amount much-needed documentation to res_sip. It should cover all
existing configuration options currently in Asterisk trunk.
Patch-by: Brad Latus (snuffy) Review:
https://reviewboard.asterisk.org/r/2471/
2013-05-19 02:21 +0000 [r389116-389132] Joshua Colp <jcolp@digium.com>
* main/pbx.c: Don't hold the outgoing lock for a prolonged period
of time as it may block the originator.
* main/pbx.c: If the caller of the originate API calls wants the
channel ensure it has been requested and dialed.
2013-05-18 23:20 +0000 [r389097] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c, configs/skinny.conf.sample: Add call
forward no answer to skinny and cleanup general callfwd handling.
CallforwardNoAnswer uses a sched to determine when to forward the
call. Defaults to 20secs but configurable in skinny.conf. Adds
dialType to each subchannel structure to be used to differentiate
between normal dials that result in a call being placed (default)
and other uses for the skinny_dialer (such as cfwd digit
collection). Restructured all cfwd handling to use this new
arrangement. (closes issue ASTERISK-21292) Reported by: wedhorn
Tested by: myself Patches: skinny-callfwdnoans03.diff uploaded by
wedhorn (license 5019)
2013-05-18 22:49 +0000 [r389053-389085] Joshua Colp <jcolp@digium.com>
* main/pbx.c: Fix a bug where synchronous origination (oddly enough
triggered by doing an async manager Originate) would not work
properly.
* include/asterisk/dial.h, main/manager_channels.c, main/dial.c,
main/pbx.c: Move origination to use the dialing API and send
Stasis messages on dial begin and end. (closes issue
ASTERISK-21549) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2512/
2013-05-17 21:10 +0000 [r389011] David M. Lee <dlee@digium.com>
* res/res_jabber.c, apps/app_queue.c, channels/chan_iax2.c,
main/endpoints.c, include/asterisk/stasis_message_router.h,
res/res_chan_stats.c, main/stasis.c, main/manager.c,
funcs/func_presencestate.c, main/stasis_message_router.c,
main/app.c, main/stasis_channels.c, res/res_stasis.c,
main/manager_channels.c, apps/app_voicemail.c,
main/stasis_cache.c, main/pbx.c, main/stasis_endpoints.c,
channels/chan_sip.c, include/asterisk/stasis.h,
main/devicestate.c: Fix shutdown assertions in stasis-core In
r388005, macros were introduced to consistently define message
types. This added an assert if a message type was used either
before it was initialized or after it had been cleaned up. It
turns out that this assertion fires during shutdown. This
actually exposed a hidden shutdown ordering problem. Since
unsubscribing is asynchronous, it's possible that the message
types used by the subscription could be freed before the final
message of the subscription was processed. This patch adds
stasis_subscription_join(), which blocks until the last message
has been processed by the subscription. Since joining was most
commonly done right after an unsubscribe, a
stasis_unsubscribe_and_join() convenience function was also
added. Similar functions were also added to the
stasis_caching_topic and stasis_message_router, since they wrap
subscriptions and have similar problems. Other code in trunk was
refactored to join() where appropriate, or at least verify that
the subscription was complete before being destroyed. Review:
https://reviewboard.asterisk.org/r/2540
2013-05-17 20:24 +0000 [r389009] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_iax2.c: Remove Character Limit On "inkeys" For IAX2
Currently, the buffer for processing "inkeys" is limited to 256
characters. If the user has many keys and the names of those key
files are long, the 256 character limit is not enough. * Change
inkeys buffer to be dynamic (closes issue ASTERISK-21398)
Reported by: Pavel Kopchyk Tested by: Pavel Kopchyk, Michael L.
Young Patches: asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff
by Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2501/
2013-05-17 17:43 +0000 [r388976] Matthew Jordan <mjordan@digium.com>
* apps/app_dial.c, main/channel.c, main/dial.c,
include/asterisk/stasis_channels.h, main/stasis_channels.c:
Publish the outbound channel's application/data when dialing This
patch does two things: * It fixes a bug where the outbound
channel's application/data set by the dialing API/app_dial is not
communicated until the channel is hung up. If that happens, AMI
would incorrectly send a NewExten event immediately after a
Hangup. This isn't really AMI's fault, as the dialing APIs never
communicated the 'helpful' app/data on the outbound channel until
it was hungup. * It makes public sending a stasis message about a
change in channel state. This is useful enough that - for now at
least - it should be public. If operations on a channel go to
being more coarse-grained, this function could be made private
again. Review: https://reviewboard.asterisk.org/r/2548 Note that
this problem was found and reported by Matt DiMeo.
2013-05-17 17:36 +0000 [r388975] Jonathan Rose <jrose@digium.com>
* include/asterisk/json.h, main/named_acl.c, CHANGES,
channels/chan_iax2.c, tests/test_security_events.c,
res/res_sip.c, main/json.c, main/manager.c,
channels/sip/include/config_parser.h, res/res_sip_nat.c,
channels/sip/dialplan_functions.c, include/asterisk/netsock2.h,
res/res_sip_outbound_registration.c,
channels/sip/config_parser.c, include/asterisk/security_events.h,
channels/sip/include/sip.h,
include/asterisk/security_events_defs.h, main/asterisk.c,
res/res_security_log.c, include/asterisk/acl.h,
res/res_sip/config_transport.c, channels/chan_sip.c,
main/security_events.c, channels/sip/security_events.c,
include/asterisk/res_sip.h: Stasis: Update security events to use
Stasis Also moves ACL messages to the security topic and gets rid
of the ACL topic (closes issue ASTERISK-21103) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/2496/
2013-05-15 21:13 +0000 [r388896] David M. Lee <dlee@digium.com>
* res/stasis/app.c, res/stasis/app.h: Fixed inverted logic in
app_add_channel(). Also added some missing doc comments for
stasis/app.h.
2013-05-15 15:58 +0000 [r388840] Kevin Harwell <kharwell@digium.com>
* main/lock.c, /: Fix for segfault in __ast_rwlock_destroy with
DEBUG_THREADS If DEBUG_THREADS is enabled __ast_rwlock_destroy
causes a segfault while trying to access a possible NULL t->track
object. A NULL check has been added before trying to access the
memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell
Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch
uploaded by Corey Farrell (license 5909) ........ Merged
revisions 388838 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 388839 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-15 15:03 +0000 [r388818] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c, /: Fix VM snapshot handling for combined
INBOX. The snapshot API contains an option that allow for
combining of new and old messages within a single snapshot. New
messages, however, include options beyond just 'INBOX' - it also
includes the Urgent folder. A previous patch that combined INBOX
and Urgent accidentally impacted snapshots that attempted to gain
messages from just the Old folder. This patch fixes the snapshot
gathering such that the API returns the appropriate messages for
the folder selected, with and without the combine option. This
should make it more clear about what's happening. Review:
https://reviewboard.asterisk.org/r/2539/ ........ Merged
revisions 388816 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-15 12:42 +0000 [r388770] Kinsey Moore <kmoore@digium.com>
* res/res_srtp.c, /, configure, include/asterisk/autoconfig.h.in,
configure.ac: Use srtp_shutdown when available This allows the
SRTP library to be shut down properly when the functionality is
offered by libsrtp. Review:
https://reviewboard.asterisk.org/r/2538/ (closes issue
ASTERISK-21719) ........ Merged revisions 388768 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 388769 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-15 02:37 +0000 [r388729-388751] David M. Lee <dlee@digium.com>
* main/named_acl.c, res/res_stasis_test.c, main/asterisk.c,
main/presencestate.c, main/stasis.c, main/stasis_cache.c,
main/stasis_endpoints.c, include/asterisk/stasis.h, main/test.c,
main/app.c, main/devicestate.c: Refactored the rest of the
message types to use the STASIS_MESSAGE_TYPE_* macros.
* res/res_stasis_answer.c (added), res/res_stasis.c,
apps/app_stasis.c, res/stasis (added), include/asterisk/module.h,
include/asterisk/stasis_app.h, include/asterisk/stasis_app_impl.h
(added), res/Makefile: Break res_stasis into smaller files. When
implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way. This patch breaks the
major components of res_stasis.c into individual files. *
res/stasis/app.c - Stasis application tracking *
res/stasis/control.c - Channel control objects *
res/stasis/command.c - Channel command object This refactoring
also allows res_stasis applications to be loaded as independent
modules, such as the new res_stasis_answer module. The bulk of
this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary. Review:
https://reviewboard.asterisk.org/r/2530/
2013-05-14 19:03 +0000 [r388701] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c, /, include/asterisk/astobj2.h: Make ao2 global
objects not always use the debug version of the ao2_ref() calls.
The debug versions of ao2_ref() should only be used if REF_DEBUG
is enabled so nothing is written to /tmp/refs unexpectedly.
(closes issue ASTERISK-21785) Reported by: abelbeck Patches:
jira_asterisk_21785_v11.patch (license #5621) patch uploaded by
rmudgett Tested by: abelbeck ........ Merged revisions 388700
from http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-14 12:47 +0000 [r388668] Kinsey Moore <kmoore@digium.com>
* res/stasis_http/resource_recordings.h,
rest-api-templates/stasis_http_resource.h.mustache,
res/res_stasis_json_endpoints.exports.in (added),
res/res_stasis_json_events.exports.in (added),
res/res_stasis_json_channels.c (added),
rest-api-templates/res_stasis_http_resource.c.mustache,
res/stasis_http/resource_events.h,
res/res_stasis_json_recordings.c (added),
res/stasis_json/resource_bridges.h (added),
res/stasis_http/resource_sounds.h, res/res_stasis_json_events.c
(added), res/res_stasis_json_bridges.exports.in (added),
res/stasis_json/resource_playback.h (added),
res/res_stasis_json_sounds.c (added),
res/stasis_http/resource_asterisk.h,
res/stasis_json/resource_channels.h (added),
rest-api-templates/stasis_json_resource.h.mustache (added),
res/res_stasis_json_channels.exports.in (added),
res/stasis_json/resource_recordings.h (added),
res/res_stasis_json_asterisk.c (added),
rest-api-templates/res_stasis_json_resource.c.mustache (added),
res/res_stasis_json_recordings.exports.in (added),
res/stasis_json/resource_events.h (added),
res/stasis_http/resource_endpoints.h,
res/stasis_json/resource_sounds.h (added),
tests/test_res_stasis.c, res/res_stasis_json_sounds.exports.in
(added), res/res_stasis_json_endpoints.c (added),
rest-api-templates/res_stasis_json_resource.exports.mustache
(added), res/stasis_http/resource_bridges.h,
res/stasis_json/resource_asterisk.h (added),
res/res_stasis_http_events.c,
res/res_stasis_json_asterisk.exports.in (added),
res/res_stasis_json_playback.exports.in (added),
res/stasis_http/resource_playback.h,
res/res_stasis_json_bridges.c (added),
res/stasis_http/resource_channels.h, res/stasis_json (added),
res/stasis_json/resource_endpoints.h (added),
res/res_stasis_json_playback.c (added), res/res_stasis.c,
rest-api-templates/make_stasis_http_stubs.py: Move JSON event
generators into separate modules This moves the JSON event
generators out of the Stasis-HTTP modules and into standalone
JSON-related counterparts so that Stasis-HTTP and res_stasis can
depend on them without creating dependency cycles. This also
provides a future location for Swagger Model validator functions
once the generators for that code are written. Review:
https://reviewboard.asterisk.org/r/2534/
2013-05-13 21:21 +0000 [r388602-388617] Michael L. Young <elgueromexicano@gmail.com>
* /, main/logger.c: Fix Missing CALL-ID When Logging Through Syslog
The CALL-ID (ie [C-00000074]) is missing when logging to syslog.
This was just an oversight when this feature was added. * Add
CALL-IDs when using syslog (closes issue ASTERISK-21430) Reported
by: Nikola Ciprich Tested by: Nikola Ciprich, Michael L. Young
Patches: asterisk-21430-syslog-callid_trunk.diff by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2526/ ........ Merged
revisions 388605 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: Fix Crash Caused By One-way Audio With
auto_* NAT Settings Fix The prior code committed, r385473, failed
to take into consideration that not all outgoing calls will be to
a peer. My fault. This patch does the following: * Check if there
is a related peer involved. If there is, check and set NAT
settings according to the peer's settings. * Fix a problem with
realtime peers. If the global setting has auto_force_rport set
and we issued a "sip reload" while a peer is still registered,
the peer's flags for NAT are reset to off. When this happens, we
were always setting the contact address of the peer to that of
the full contact info that we had. (closes issue ASTERISK-21374)
Reported by: jmls Tested by: Michael L. Young Patches:
asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2524/
........ Merged revisions 388601 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-13 20:37 +0000 [r388598] Kinsey Moore <kmoore@digium.com>
* res/res_srtp.c, /: Revert r388529 for now Adding the cleanup
function needs some deeper thought since it apparently doesn't
exist for all variants of libsrtp. ........ Merged revisions
388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 388597 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-13 19:29 +0000 [r388579] Jonathan Rose <jrose@digium.com>
* main/pbx.c, /: pbx: Fix lack of cleanup on macrolock and
context_table (closes issue ASTERISK-21723) Reported by: Corey
Farrell Patches: core-pbx-cleanup.patch uploaded by Correy
Farrell (license 5909) ........ Merged revisions 388532 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 388578 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-13 18:10 +0000 [r388531] Kinsey Moore <kmoore@digium.com>
* res/res_srtp.c, /: Close libsrtp properly Ensure that libsrtp is
shutdown properly when res_srtp is unloaded. (closes issue
ASTERISK-21719) Reported by: Corey Farrell Patches:
res_srtp-library-shutdown.patch uploaded by Corey Farrell
........ Merged revisions 388529 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 388530 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-13 17:20 +0000 [r388526] Jonathan Rose <jrose@digium.com>
* channels/chan_gulp.c: chan_gulp: Minor readability Improvements
to chan_gulp (closes issue ASTERISK-21670) Reported by: Snuffy
Review: https://reviewboard.asterisk.org/r/2473/ Patches:
gulp-coding-guide.diff uploaded by snuffy (license 5024)
2013-05-13 14:28 +0000 [r388479] Richard Mudgett <rmudgett@digium.com>
* main/manager.c, /: Fix SendText AMI action to never return
non-zero. AMI actions must never return non-zero unless they
intend to close the AMI connection. (Which is almost never.)
(closes issue ASTERISK-21779) Reported by: Paul Goldbaum ........
Merged revisions 388477 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 388478 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-10 22:12 +0000 [r388427] Richard Mudgett <rmudgett@digium.com>
* /, channels/misdn/isdn_msg_parser.c: Allow mISDN to send PROGRESS
messsage. * Made isdn_msg_parser.c build a progress message with
the mandatory progress indicator IE. (The mISDNuser NT state
machine rejected sending the incomplete message.) Note: The
associated mISDN and mISDNuser patches respectively are viewable
here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes
issue AST-1153) Reported by: Guenther Kelleter Patches:
progress-chan_misdn.diff (license #6372) patch uploaded by
Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch
uploaded by Guenther Kelleter progress-misdnuser.diff (license
#6372) mISDNuser patch uploaded by Guenther Kelleter ........
Merged revisions 388425 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 388426 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-10 20:50 +0000 [r388380] Mark Michelson <mmichelson@digium.com>
* /, pbx/pbx_dundi.c: Fix memory leak in pbx_dundi pbx_dundi added
an io context without removing it. This caused a memory leak when
the module was unloaded. (closes ASTERISK-21718) Reported by
Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by
Corey Farrell (License #5909) ........ Merged revisions 388376
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 388378 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-10 20:28 +0000 [r388375] Michael L. Young <elgueromexicano@gmail.com>
* res/res_config_odbc.c: Fix Finding Extensions With Patterns Using
ODBC Realtime After the merge of support for the realtime sorcery
module, extensions that contained a pattern were not being found
through odbc realtime. It was tracked down to this one line that
was advancing to the next variable list before it should have
been. The removal of this one line fixes this. Tested this fix on
my machine. Received confirmation that this is the right fix from
file on IRC.
2013-05-10 17:12 +0000 [r388318-388350] David M. Lee <dlee@digium.com>
* res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
res/res_stasis_http_recordings.c,
res/res_stasis_http_endpoints.c, main/loader.c,
res/res_stasis_http_events.c, res/res_stasis_http_sounds.c,
res/res_stasis_http_bridges.c, res/res_stasis_http.c,
res/res_stasis.c, apps/app_stasis.c,
res/res_stasis_http_asterisk.c,
rest-api-templates/res_stasis_http_resource.c.mustache,
res/res_stasis_http_playback.c, res/res_stasis_websocket.c,
tests/test_res_stasis.c: Address unload order issues for
res_stasis* modules I've noticed when doing a graceful shutdown
that the res_stasis_http.so module gets unloaded before the
modules that use it, which causes some asserts during their
unload. While r386928 was a quick hack to get it to not assert
and die, this patch increases the use counts on res_stasis.so and
res_stasis_http.so properly. It's a bigger change than I
expected, hence the review instead of just committing it. Review:
https://reviewboard.asterisk.org/r/2489/
* include/asterisk/stasis.h: Avoided __ast names for the private
variables created by the STASIS_MESSAGE_TYPE_*() macros.
2013-05-10 13:13 +0000 [r388275] Kinsey Moore <kmoore@digium.com>
* rest-api-templates/event_function_decl.mustache (added),
res/stasis_http/resource_sounds.h, CHANGES,
res/res_stasis_http_events.c, include/asterisk/stasis_channels.h,
main/stasis_channels.c, rest-api-templates/swagger_model.py,
res/res_stasis.c, main/manager_channels.c,
rest-api-templates/stasis_http_resource.h.mustache,
res/stasis_http/resource_recordings.h,
rest-api-templates/asterisk_processor.py,
rest-api-templates/res_stasis_http_resource.c.mustache,
res/stasis_http/resource_endpoints.h,
rest-api/api-docs/events.json, res/stasis_http/resource_events.h,
res/res_stasis_websocket.c, apps/app_userevent.c: Add channel
events for res_stasis apps This change adds a framework in
res_stasis for handling events from channel topics. JSON event
generation and validation code is created from event
documentation in rest-api/api-docs/events.json to assist in JSON
event generation, ensure consistency, and ensure that accurate
documentation is available for ALL events that are received by
res_stasis applications. The userevent application has been
refactored along with the code that handles userevent channel
blob events to pass the headers as key/value pairs in the JSON
blob. As a side-effect, app_userevent now handles duplicate keys
by overwriting the previous value. Review:
https://reviewboard.asterisk.org/r/2428/ (closes issue
ASTERISK-21180) Patch-By: Kinsey Moore <kmoore@digium.com>
2013-05-10 11:47 +0000 [r388254] Sean Bright <sean@malleable.com>
* /, channels/chan_sip.c: Fix copy/paste error in
one-touch-recording implementation. ........ Merged revisions
388253 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-09 14:41 +0000 [r388175] Matthew Jordan <mjordan@digium.com>
* apps/app_userevent.c: Don't expect to pack three tuples when you
only have two
2013-05-09 04:11 +0000 [r388110-388113] Michael L. Young <elgueromexicano@gmail.com>
* res/res_rtp_asterisk.c, /: Fix The Payload Being Set On CN
Packets And Do Not Set Marker Bit When we send out a CN packet
(for instance, in the case of using rtpkeepalives), we are not
setting the payload code properly. Also, we are setting the
marker bit when we shouldn't be according to RFC 3389, section 4.
AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we
should be using ast_rtp_codecs_payload_code() rather than
ast_rtp_codecs_payload_lookup(). 11 and trunk already use the
appropriate function. * In 1.8, use ast_rtp_codecs_payload_code()
* Remove the setting of the marker bit * Fix the debug message by
incrementing the seqno after the debug message is set in order to
display the correct seqno that was sent out (closes issue
ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter
Katzmann, Michael L. Young Patches:
asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2500/ ........ Merged
revisions 388111 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 388112 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_queue.c: Fix Segfault In app_queue When
"persistentmembers" Is Enabled And Using Realtime When the
"ignorebusy" setting was deprecated, we added some code to allow
us to be compatible with older setups that are still using the
"ignorebusy" setting instead of "ringinuse". We set a char
*variable with the column name to use, which helps the realtime
functions to use the correct column in their SQL queries. When
"persistentmembers" is enabled, we are not setting this variable
before the realtime functions were called to load members. This
results in the variable being NULL and therefore causing a
segfault when loading members during the module's process of
loading. The solution was to move the code that sets that
variable to be before these realtime functions are called during
the loading of the module. (closes issue ASTERISK-21738) Reported
by: JoshE Tested by: JoshE Patches:
asterisk-21738-rt-ringinuse-field-not-set.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2499/ ........ Merged
revisions 388108 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-08 22:00 +0000 [r388014-388075] David M. Lee <dlee@digium.com>
* res/res_stasis_websocket.c: Fixed MODFLAG for
res_stasis_websocket
* build_tools/cflags.xml, include/asterisk/inline_api.h: Add
development flag to disable the inline API. A GCC bug[1] can, in
some cases, pop up an unsuppressible pedwarn when using a static
inline standard library function from a non-static inline
function. This normally doesn't show up, but can occur if you're
running an upgrade version of GCC (such as GCC 4.8 on OS X, which
normally runs GCC 4.2). [1]:
http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
* main/srv.c, main/enum.c: Removed #if checks for crazy old
versions of OS X. The <arpa/nameser_compat.h> was introduced way
back in OS X Panther, which itself was end-of-lifed back in 2007.
We can assume that any OS X machine we build on will need that
header file :-) Why bother removing it? The flag we're checking
(__APPLE_CC__) is actually Apple's build number. Self-compiled
versions of GCC (such as installing the latest version of GCC
from homebrew) sets the value to 0, making it useless for this
sort of compile flaggery.
* tests/test_stasis_endpoints.c: Fixed set-but-not-used warning
caught by newer GCC
2013-05-08 18:36 +0000 [r388008] Matthew Jordan <mjordan@digium.com>
* apps/app_directory.c: Don't perform a realtime lookup with a NULL
keyword Previously, a call to ast_load_realtime_multientry could
get away with passing a NULL parameter to the function, even
though it really isn't supposed to do that. After the change over
to using ast_variable instead of variadic arguments, the realtime
engine gets unhappy if you do this. This was always an unintended
function call in app_directory anyway - now, we just don't call
into the realtime function calls if we don't have anything to
query on.
2013-05-08 18:34 +0000 [r388005] David M. Lee <dlee@digium.com>
* main/stasis_channels.c, res/res_stasis.c,
main/manager_channels.c, main/channel.c,
include/asterisk/stasis_channels.h, tests/test_stasis_channels.c,
apps/app_userevent.c, include/asterisk/stasis.h: Remove required
type field from channel blobs When we first introduced the
channel blob types, the JSON blobs were self identifying by a
required "type" field in the JSON object itself. This, as it
turns out, was a bad idea. When we introduced the message router,
it was useless for routing based on the JSON type. And messages
had two type fields to check: the stasis_message_type() of the
message itself, plus the type field in the JSON blob (but only if
it was a blob message). This patch corrects that mistake by
removing the required type field from JSON blobs, and introducing
first class stasis_message_type objects for the actual message
type. Since we now will have a proliferation of message types, I
introduced a few macros to help reduce the amount of boilerplate
necessary to set them up. Review:
https://reviewboard.asterisk.org/r/2509
2013-05-08 16:58 +0000 [r387974] Richard Mudgett <rmudgett@digium.com>
* utils: Add version.c to list of ignored files in the utils
directory.
2013-05-08 13:39 +0000 [r387932] David M. Lee <dlee@digium.com>
* tests/test_endpoints.c (added),
include/asterisk/stasis_endpoints.h (added),
res/res_stasis_test.c (added),
res/stasis_http/resource_endpoints.c, channels/sip/include/sip.h,
main/asterisk.c, rest-api/api-docs/endpoints.json,
res/stasis_http/resource_endpoints.h, main/stasis_cache.c,
main/stasis_endpoints.c (added), channels/chan_sip.c,
include/asterisk/endpoints.h (added), include/asterisk/astobj2.h,
main/channel_internal_api.c, include/asterisk/stasis_test.h
(added), include/asterisk/stasis.h, main/endpoints.c (added),
main/astobj2.c, res/res_stasis_http_endpoints.c,
tests/test_stasis_endpoints.c (added),
res/res_stasis_test.exports.in (added): Initial support for
endpoints. An endpoint is an external device/system that may
offer/accept channels to/from Asterisk. While this is a very
useful concept for end users, it is surprisingly not a core
concept within Asterisk itself. This patch defines ast_endpoint
as a separate object, which channel drivers may use to expose
their concept of an endpoint. As the channel driver creates
channels, it can use ast_endpoint_add_channel() to associate
channels to the endpoint. This updated the endpoint
appropriately, and forwards all of the channel's events to the
endpoint's topic. In order to avoid excessive locking on the
endpoint object itself, the mutable state is not accessible via
getters. Instead, you can create a snapshot using
ast_endpoint_snapshot_create() to get a consistent snapshot of
the internal state. This patch also includes a set of topics and
messages associated with endpoints, and implementations of the
endpoint-related RESTful API. chan_sip was updated to create
endpoints with SIP peers, but the state of the endpoints is not
updated with the state of the peer. Along for the ride in this
patch is a Stasis test API. This is a stasis_message_sink object,
which can be subscribed to a Stasis topic. It has functions for
blocking while waiting for conditions in the message sink to be
fulfilled. (closes issue ASTERISK-21421) Review:
https://reviewboard.asterisk.org/r/2492/
2013-05-08 07:21 +0000 [r387885] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_sip.c: chan_sip: NOTIFYs for BLF start queuing
up and fail to be sent out after retries fail RFC6665 4.2.2: ...
after a failed State NOTIFY transaction remove the subscription
The problem is that the State Notify requests rely on the 200OK
reponse for pacing control and to not confuse the notify
susbsystem. The issue is, the pendinginvite isn't cleared if a
response isn't received, thus further notify's are never sent.
The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the
subscription after failure. (closes issue ASTERISK-21677)
Reported by: Dan Martens Tested by: alecdavis alecdavis (license
585) Review https://reviewboard.asterisk.org/r/2475/ ........
Merged revisions 387875 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387880 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-07 18:32 +0000 [r387803-387825] David M. Lee <dlee@digium.com>
* include/asterisk/lock.h: Fixed up \example marker in lock.h
Doxygen comment. The \example tags marks an entire file as an
example, not a code snippet.
* res/res_config_pgsql.c, main/manager.c, /: Minor fixups to
Doxygen comments. The \example tags marks an entire file as an
example, not a code snippet. ........ Merged revisions 387823
from http://svn.asterisk.org/svn/asterisk/branches/11
* include/asterisk/json.h: Better explained the depths of reference
stealing.
2013-05-07 17:53 +0000 [r387802] Jason Parker <jparker@digium.com>
* include/asterisk.h: Fix build breakage, from LOW_MEMORY fix.
2013-05-06 17:15 +0000 [r387740-387741] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h: Update ao2_destructor_fn doxygen.
* channels/chan_dahdi.c: Make a log NOTICE more explicit that the
event comes from DAHDI and not PRI.
2013-05-06 17:01 +0000 [r387738] Jason Parker <jparker@digium.com>
* main/asterisk.c: Fix building with LOW_MEMORY defined.
2013-05-06 15:58 +0000 [r387690] Russell Bryant <russell@russellbryant.com>
* /, apps/app_meetme.c: Make SLA reload more paranoid. Reload
support was originally not included for SLA. It was added later,
but in a fairly non-traditional way. It basically sets a flag
indicating that a reload is pending, and then waits for a time
where it thinks everything SLA related is idle and unused, and
*then* executes the reload. It does this because the reload
process is destructive. It starts by throwing everything away and
starting over. There are a number of problems with this approach.
One of them is that the check to see if anything in use was
incomplete. This patch makes it more complete and thus less
likely for a crash to occur during reload processing. However,
this approach still has problems so some much more significant
reworking of this code will need to come in as a next step. Patch
credit and testing by CoreDial, LLC. ........ Merged revisions
387688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 387689 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-06 13:04 +0000 [r387662] Joshua Colp <jcolp@digium.com>
* include/asterisk/sorcery.h, res/res_sorcery_astdb.c,
tests/test_sorcery.c, main/sorcery.c: Add support for observers
and JSON objectset creation to sorcery. This change adds the
ability for modules to add themselves as observers to sorcery
object types. Observers can be notified when objects are created,
updated, or deleted as well as when the object type is loaded or
reloaded. Observer notifications are done using a thread pool in
a serialized fashion so the caller of the sorcery API calls is
minimally impacted. This also adds the ability to create JSON
changesets of a sorcery object. Tests are also present to confirm
all of the above functionality. Review:
https://reviewboard.asterisk.org/r/2477/
2013-05-04 16:00 +0000 [r387630-387633] Matthew Jordan <mjordan@digium.com>
* main/asterisk.c, include/asterisk.h: Clean up documentation;
prevent ref leak on exit This patch: * Cleans up some doxygen *
Prevents leaking the system level Stasis topics and messages on
exit (users of valgrind will be happier)
* funcs/func_global.c: Migrate SHARED's use of the VarSet AMI event
to Stasis-Core This patch removes the direct call to AMI from the
SHARED function and instead call Stasis-Core. Stasis-Core
delivers the notification that a shared variable has changed on a
channel to all interested consumers. (issue ASTERISK-21462)
2013-05-03 18:03 +0000 [r387594] Jonathan Rose <jrose@digium.com>
* main/asterisk.c, include/asterisk.h, channels/chan_sip.c,
res/res_stun_monitor.c, main/event.c, channels/chan_iax2.c:
Stasis: Convert network change events into network change stasis
messages (issue ASTERISK-21103) Review:
https://reviewboard.asterisk.org/r/2490/
2013-05-03 11:35 +0000 [r387545] Joshua Colp <jcolp@digium.com>
* res/res_sip_sdp_rtp.c, channels/chan_gulp.c: Use the configured
formats for Gulp sessions if there are no joint formats between
requested formats and configured formats. (closes issue
ASTERISK-21756)
2013-05-02 20:59 +0000 [r387519] Matthew Jordan <mjordan@digium.com>
* build_tools/post_process_documentation.py, apps/app_stack.c:
Migrate AMI VarSet events raised by GoSub local variables This
patch moves VarSet events for local variables raised by GoSub
over to Stasis-Core. It also tweaks up the post-processing
documentation scripts to not combine parameters if both
parameters are already documented. (issue ASTERISK-21462)
2013-05-02 19:06 +0000 [r387482] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Remove the ABI compatability ast_channel_alloc().
It is no longer needed.
2013-05-02 17:15 +0000 [r387423] Matthew Jordan <mjordan@digium.com>
* utils/Makefile, /: Update utils Makefile to handle r387294 Alec's
patch that added the Asterisk version to 'core show locks'
angered the items in utils, as they exist somewhat outside of the
Asterisk build system. Some day, this Makefile should get nuked
from high orbit, but for now, include version.c in its list of
stuff to pile in. ........ Merged revisions 387421 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387422 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-02 16:39 +0000 [r387420] Jonathan Rose <jrose@digium.com>
* include/asterisk/event_defs.h, main/event.c: Putting all event
defs and names back for now due to res_corosync dependency
2013-05-02 08:24 +0000 [r387296-387369] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip:
Session-Expires: Set timer to correctly expire at (~2/3) of the
interval when not the refresher RFC 4028 Section 10 if the side
not performing refreshes does not receive a session refresh
request before the session expiration, it SHOULD send a BYE to
terminate the session, slightly before the session expiration.
The minimum of 32 seconds and one third of the session interval
is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the
Session-Expires interval, or if the remote device was the
refresher, asterisk would timeout at interval end. Now, when not
refresher, timeout as per RFC noted above. (closes issue
ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2488/ ........ Merged
revisions 387344 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387345 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: chan_sip: Honor Session-Expires in 200OK
response when it's a RE-INVITE when asterisk is the refresher.
RFC 4028 Section 7.2 "UACs MUST be prepared to receive a
Session-Expires header field in a response, even if none were
present in the request." What changed After ASTERISK-20787,
inbound calls to asterisk with no Session-Expires in the INVITE
are now are offered a Session-Expires (1800 asterisk default) in
the response, with asterisk as the refresher. Symptom: After 900
seconds (asterisk default refresher period 1800), asterisk
RE-INVITEs the device, the device may respond with a much lower
Session-Expires (180 in our case) value that it is now using.
Asterisk ignores this response, as it's deemed both an INBOUND
CALL, and a RE-INVITE. After 180 seconds the device times out and
sends BYE (hangs up), asterisk is still working with the
refresher period of 1800 as it ignored the 'Session Expires: 180'
in the previous 200OK response. Fix: handle_response_invite()
when 200OK, remove check for outbound and reinvite. (closes issue
ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2463/ ........ Merged
revisions 387312 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387319 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_dahdi.c, /: chan_dahdi: fix lower bound check with
-ve integer conversion from a float Lower bound of a 16bit signed
int is -32768 not -32767 (closes issue ASTERISK-21744) Reported
by: alecdavis Tested by: alecdavis alecdavis (license 585)
........ Merged revisions 387297 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387298 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/utils.c: Add Asterisk Version to core show locks Assist
with reporting 'core show locks' when submitting bug reports.
Example below: =========================== == SVN-branch-1.8-...
== Currently Held Locks =========================== (closes issue
ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) ........ Merged revisions 387294 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387295 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-01 21:55 +0000 [r387260-387261] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: Simplify
chan_local.c:manager_optimize_away() using ao2_find().
* channels/chan_local.c: Cleanup chan_local.c:local_new(). * Remove
t and ama local variables. There is no way they could be anything
other than default because p->owner can only be NULL at this
point. * Rename tmp and tmp2 to owner and chan respectively. *
Remove redundant initialization of channel context, exten,
priority.
2013-05-01 21:18 +0000 [r387220] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c, /: Clear the DTMF sending digit tracking
on off nominal paths In certain situations, when the RTP engine
goes to send a DTMF end digit it may be in a situation where the
remote address is no longer available, or the digit that was
supposed to be sent is invalid. In such cases, we need to clear
the RTP counters appropriately. Otherwise, when the RTP source is
set again, we'll continue to think that we're in the middle of
sending a DTMF digit, which can confuse the remote party
(signficantly). (closes issue ASTERISK-21522) Reported by: Corey
Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey
Farrell (License 5909) ........ Merged revisions 387213 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387216 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-01 21:09 +0000 [r387181-387212] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: Trivial changes. Comments, parentheses,
spelling, wording.
* channels/chan_local.c: Make chan_local locals container an
explicit list container. Pretending that chan_local locals
container can have more than one bucket is silly. The container
has no key to help search.
* channels/chan_local.c: Whitespace changes.
* main/loader.c: Make mod_load_cmp() not as klunky. There is a
reason the heap comparison functions like qsort(), and other
comparison functions specify <0, >0, and =0 for the return
values.
* channels/chan_unistim.c: Remove some unnecessary calls to
ast_bridged_channel() in chan_unistim.c
* channels/chan_mgcp.c: Remove some unnecessary calls to
ast_bridged_channel() in chan_mgcp.c
* channels/chan_skinny.c: Remove some unnecessary calls to
ast_bridged_channel() in chan_skinny.c
* channels/chan_iax2.c: Remove some unnecessary calls to
ast_bridged_channel() in chan_iax2.c
* channels/chan_dahdi.c, channels/sig_analog.c: Remove some
unnecessary calls to ast_bridged_channel() in
chan_dahdi.c/sig_analog.c
2013-05-01 18:38 +0000 [r387135] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Prevent crash in 'sip show peers' when
the number of peers on a system is large When you have lots of
SIP peers (according to the issue reporter, around 3500), the
'sip show peers' CLI command or AMI action can crash due to a
poorly placed string duplication that occurs on the stack. This
patch refactors the command to not allocate the string on the
stack, and handles the formatting of a single peer in a separate
function call. (closes issue ASTERISK-21466) Reported by:
Guillaume Knispel patches:
fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch
uploaded by gknispel (License 6492) ........ Merged revisions
387134 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-05-01 17:15 +0000 [r387108] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Move some annoying chan_dahdi debug
messages to level 5.
2013-04-30 22:50 +0000 [r387039] Matthew Jordan <mjordan@digium.com>
* main/features.c, /: Fix CDR not being created during an
externally initiated blind transfer Way back when in the dark
days of Asterisk 1.8.9, blind transferring a call in a context
that included the 'h' extension would inadvertently execute the
hangup code logic on the transferred channel. This was a "bad
thing". The fix was to properly check for the softhangup flags on
the channel and only execute the 'h' extension logic (and, in
later versions, hangup handler logic) if the channel was well and
truly dead (Jim). Unfortunately, CDRs are fickle. Setting the
softhangup flag when we detected that the channel was leaving the
bridge (but not to die) caused some crucial snippet of CDR code,
lying in ambush in the middle of the bridging code, to not get
executed. This had the effect of blowing away one of the CDRs
that is typically created during a blind transfer. While we live
and die by the adage "don't touch CDRs in release branches", this
was our bad. The attached patch restores the CDR behavior, and
still manages to not run the 'h' extension during a blind
transfer (at least not when it's supposed to). Thanks to Steve
Davies for diagnosing this and providing a fix. Review:
https://reviewboard.asterisk.org/r/2476 (closes issue
ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq
Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by
one47 (License 5012) ........ Merged revisions 387036 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 387038 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-30 22:37 +0000 [r387035-387037] Jonathan Rose <jrose@digium.com>
* main/event.c, include/asterisk/json.h, channels/chan_iax2.c,
main/named_acl.c, include/asterisk/acl.h, main/json.c,
main/manager.c, channels/chan_sip.c,
include/asterisk/event_defs.h: Stasis Core: Refactor ACL Change
events to go out over the stasis core msg bus (issue
ASTERISK-21103) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2481/
* /, main/event.c: Add forgotten event types to event_names array
........ Merged revisions 387030 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-30 18:12 +0000 [r386990] Jason Parker <jparker@digium.com>
* channels/chan_gulp.c: Fix a log message.
2013-04-30 13:48 +0000 [r386931] Sean Bright <sean@malleable.com>
* include/asterisk/utils.h, /: Use the proper lower bound when
doing saturation arithmetic. 16 bit signed integers have a range
of [-32768, 32768). The existing code was using the interval
(-32768, 32768) instead. This patch fixes that. Review:
https://reviewboard.asterisk.org/r/2479/ ........ Merged
revisions 386929 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 386930 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-30 13:37 +0000 [r386928] David M. Lee <dlee@digium.com>
* tests/test_stasis_http.c, res/res_stasis_http.c: Just a couple of
Stasis-HTTP nitpick fixes. * Fixed crash when res_stasis_http is
unloaded before the implementation modules. * Cleaned up test
initialization for test_stasis_http.so.
2013-04-29 23:36 +0000 [r386879] Rusty Newton <rnewton@digium.com>
* sounds/Makefile, /: Modifying sounds/Makefile to pull down 1.4.24
core sounds 1.4.24 core sounds includes a full set of Italian
prompts for core sounds and a fix for the missing voicemail
prompts in the Russian language. (closes issue ASTERISK-19431)
(closes issue ASTERISK-19721) ........ Merged revisions 386877
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 386878 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-29 13:38 +0000 [r386793-386841] Olle Johansson <oej@edvina.net>
* /, CHANGES, apps/app_queue.c: Play periodic prompts for first
call in a call queue Review:
https://reviewboard.asterisk.org/r/2263/ ........ Merged
revisions 386792 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 386794 from
http://svn.asterisk.org/svn/asterisk/branches/11
* include/asterisk/doxygen/commits.h: Change pointer to existing
wiki page instead of non-existing page
2013-04-28 03:32 +0000 [r386774] Kinsey Moore <kmoore@digium.com>
* rest-api-templates/swagger_model.py: Fix spelling error in python
doc
2013-04-27 19:03 +0000 [r386731-386760] Joshua Colp <jcolp@digium.com>
* res/res_sip.c: Tweak res_sip priority so it gets loaded first
before all other SIP stuff.
* res/res_config_sqlite.c: Update res_config_sqlite to use the
ast_variable lists.
* CHANGES, res/res_config_ldap.c, main/config.c,
tests/test_sorcery_realtime.c (added), main/sorcery.c,
res/res_sorcery_realtime.c (added), addons/res_config_mysql.c,
res/res_config_sqlite3.c, res/res_config_curl.c,
res/res_config_pgsql.c, res/res_config_odbc.c,
include/asterisk/config.h: Add support for a realtime sorcery
module. This change does the following: 1. Adds the sorcery
realtime module 2. Adds unit tests for the sorcery realtime
module 3. Changes the realtime core to use an ast_variable list
instead of variadic arguments 4. Changes all realtime drivers to
accept an ast_variable list Review:
https://reviewboard.asterisk.org/r/2424/
2013-04-26 21:52 +0000 [r386685-386686] Matthew Jordan <mjordan@digium.com>
* res/res_sip_nat.c, res/res_sip_registrar.c,
res/res_sip_dtmf_info.c,
res/res_sip_outbound_authenticator_digest.c,
res/res_sip_rfc3326.c, res/res_sip_outbound_registration.c,
res/res_sip_endpoint_identifier_ip.c,
res/res_sip_endpoint_identifier_constant.c, res/res_sip_mwi.c,
res/res_sip_acl.c, res/res_sip_logger.c,
res/res_sip_endpoint_identifier_user.c, res/res_sip_pubsub.c: Add
missing module dependencies to various res_sip* modules This
patch updates the various res_sip modules with their proper
menuselect options and proper dependencies, such that Asterisk
still has a snowball's chance in hell of compiling without
pjproject. Much thanks to snuffy(-home|-work) for making
everyone's life easier with this patch. Review:
https://reviewboard.asterisk.org/r/2472/ (closes issue
ASTERISK-21669) Reported by: snuffy patches: xml-depends.diff
uploaded by snuffy (license 5024)
* /, main/config.c: Clean up memory leak in config file on off
nominal paths when glob is allowed If a system allows for its
usage, Asterisk will use glob to help parse Asterisk .conf files.
The config file loading routine was leaking the memory allocated
by the glob() routine when the config file was in an unmodified
or invalid state. This patch properly calls globfree in those off
nominal paths. (closes issue ASTERISK-21412) Reported by: Corey
Farrell patches: config_glob_leak.patch uploaded by Corey Farrell
(license 5909) ........ Merged revisions 386672 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 386677 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-26 21:31 +0000 [r386684] David M. Lee <dlee@digium.com>
* main/loader.c: By popular demand, putting the
about-to-load-module printf back. But now it only prints during
the initial startup, and prints at verbose 1 level.
2013-04-26 21:27 +0000 [r386676] Matthew Jordan <mjordan@digium.com>
* /, main/features.c: Clean up resources in features on exit This
patch cleans up two things features: * It properly unregisters
the CLI commands that features registered * It cancels and
performs a pthread_join on the created parking thread. This not
only properly joins a non-detached thread, but also prevents
disposing of the parking lots prior to the parking thread
completely exiting. (closes issue ASTERISK-21407) Reported by:
Corey Farrell patches: features_shutdown-r2.patch uploaded by
Corey Farrell (License 5909) ........ Merged revisions 386641
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 386642 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-26 21:00 +0000 [r386640] David M. Lee <dlee@digium.com>
* main/loader.c: Removing stray printf from r386540
2013-04-26 20:32 +0000 [r386638] Mark Michelson <mmichelson@digium.com>
* main/uuid.c: Add an \extref doxygen pointer for libuuid. Thanks
to Olle Johansson for suggesting this.
2013-04-26 20:05 +0000 [r386623-386624] David M. Lee <dlee@digium.com>
* res/res_chan_stats.c (added), res/res_statsd.exports.in (added),
configs/statsd.conf.sample (added), include/asterisk/utils.h,
include/asterisk/statsd.h (added), res/res_statsd.c (added):
Example of how to use the Stasis message bus In order to get
people familiar with the Stasis message bus, it would be useful
to have something of a tutorial. Since I'm not clever enough to
think of some cool integration we could do with Twitter, I
settled for something that might actually be useful. This patch
adds a res_statsd.so module, which implements a basic statsd[1]
client. Statsd is a very simple statistics gathering server,
which can publish its results to a backend graphing engine, like
Graphite[2]. There are several different Statsd server
implementations[3], so you can pick what works best for your
environment. The actual example of how to use the Stasis message
bus is in res_chan_stats.so. This module demonstrates how to use
subscriptions and the message router by monitoring messages and
posting channels stats to the statsd server. A wiki page walking
through res_chan_stats.so is forthcoming. [1]:
https://github.com/etsy/statsd/ [2]:
http://graphite.readthedocs.org/en/latest/ [3]:
http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/
Review: https://reviewboard.asterisk.org/r/2460/
* res/res_sip: Ignore *.[oi] files in res/res_sip
2013-04-25 21:32 +0000 [r386577] Joshua Colp <jcolp@digium.com>
* configs/res_sip.conf.sample: Don't bind to anything in the sample
configuration so we don't clash with chan_sip on a "make samples"
right now.
2013-04-25 18:28 +0000 [r386540-386541] Mark Michelson <mmichelson@digium.com>
* /: REmove automerge properties.
* res/res_sip/sip_options.c, res/res_sip_pubsub.exports.in (added),
res/res_sip_rfc3326.c (added), res/res_sip_mwi.c (added),
main/sorcery.c, res/res_sip (added),
include/asterisk/threadpool.h, res/res_sip_registrar.c (added),
res/res_sip/sip_distributor.c, res/res_sip/config_auth.c,
include/asterisk/res_sip_session.h (added),
res/res_sip_endpoint_identifier_ip.c (added), channels/Makefile,
tests/test_sorcery.c, res/res_sip/config_domain_aliases.c,
res/res_sip_endpoint_identifier_user.c (added), res/res_sip.c
(added), include/asterisk/res_sip_pubsub.h (added),
include/asterisk/sorcery.h,
res/res_sip_outbound_authenticator_digest.c (added),
res/res_sip/location.c, res/res_sip_outbound_registration.c
(added), res/res_sip_endpoint_identifier_constant.c (added),
res/res_sip_acl.c (added), res/res_sip_pubsub.c (added),
res/res_sorcery_config.c, res/res_sip/config_transport.c,
configs/res_sip.conf.sample (added),
res/res_sip/sip_configuration.c, /,
include/asterisk/autoconfig.h.in, include/asterisk/res_sip.h
(added), res/res_sip_dtmf_info.c (added),
res/res_sip/include/res_sip_private.h, res/res_sip.exports.in
(added), main/threadpool.c, res/Makefile,
res/res_sip_authenticator_digest.c (added), main/taskprocessor.c,
res/res_sip_session.exports.in (added), main/astobj2.c,
res/res_sip_sdp_rtp.c (added), res/res_sip/sip_outbound_auth.c,
main/loader.c, channels/chan_gulp.c (added),
res/res_sip_caller_id.c (added), res/res_sip_logger.c (added),
res/res_sip/include, res/res_sip_nat.c (added), configure,
res/res_sip_session.c (added): Merge the pimp_my_sip branch into
trunk. The pimp_my_sip branch is being merged at this point
because it offers basic functionality, and from an API
standpoint, things are complete. SIP work is *not*
feature-complete; however, with the completion of the
SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been
created, and thus it is possible for developers to attempt to
create new SIP work. API documentation can be found in the
doxygen in the code, but usability documentation is still
lacking.
2013-04-25 03:04 +0000 [r386485-386487] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix Displaying Symmetric RTP Global
Setting * Use comedia_string() to display correctly the symmetric
rtp setting when running "sip show settings" ........ Merged
revisions 386486 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: Change Case On Forcerport For Consistency
* Change "ForcerPort" to "Forcerport" to match everywhere else it
is displayed ........ Merged revisions 386483 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 386484 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-24 21:47 +0000 [r386461-386462] David M. Lee <dlee@digium.com>
* res/stasis_http/resource_bridges.h,
res/stasis_http/resource_recordings.h,
rest-api-templates/stasis_http_resource.h.mustache,
res/stasis_http/resource_endpoints.h,
res/stasis_http/resource_events.h,
res/stasis_http/resource_asterisk.h,
res/stasis_http/resource_playback.h,
res/stasis_http/resource_channels.h,
res/stasis_http/resource_sounds.h: Document JSON models in
resource_*.h
* rest-api-templates/swagger_model.py: Oops. Mustache doesn't like
dictionaries
2013-04-23 20:18 +0000 [r386375] Richard Mudgett <rmudgett@digium.com>
* apps/confbridge/conf_config_parser.c, apps/app_confbridge.c:
confbridge: Make search the conference bridges container using
OBJ_KEY. * Make confbridge config parsing user profile, bridge
profile, and menu container hash/cmp functions correctly check
the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags. * Made
confbridge load_module()/unload_module() free all resources on
failure conditions.
2013-04-23 18:57 +0000 [r386352] Kinsey Moore <kmoore@digium.com>
* res/res_stasis.c: Fix some bad whitespace This crept in with the
RESTful HTTP interface merge.
2013-04-22 16:44 +0000 [r386289] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Fix crash when AMI redirect action redirects
two channels out of a bridge. The two party bridging loops were
changing the bridge peer pointers without the channel locks held.
Thus when ast_channel_massquerade() tested and used the pointer
there is a small window of opportunity for the pointers to become
NULL even though the masquerade code has the channels locked.
(closes issue ASTERISK-21356) Reported by: William luke Patches:
jira_asterisk_21356_v11.patch (license #5621) patch uploaded by
rmudgett Tested by: William luke ........ Merged revisions 386256
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 386286 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-22 16:22 +0000 [r386266] Andrew Latham <lathama@gmail.com>
* include/asterisk/srv.h: Doxygen - Markup Guidelines Expand on a
commit by OEJ to use the Coding-Guidelines (issue ASTERISK-20259)
2013-04-22 14:58 +0000 [r386232] David M. Lee <dlee@digium.com>
* res/stasis_http/resource_channels.c, res/res_stasis_http_sounds.c
(added), rest-api (added), main/http.c,
res/res_stasis_http_bridges.c (added), tests/test_stasis_http.c
(added), include/asterisk/strings.h, res/res_stasis_http.c
(added), tests/test_stasis.c, res/res_stasis.c,
res/res_stasis_http_asterisk.c (added),
res/res_stasis_http_playback.c (added), res/stasis_http (added),
configs/stasis_http.conf.sample (added),
include/asterisk/stasis_http.h (added),
res/res_stasis_http_channels.c (added),
include/asterisk/stasis_app.h, res/Makefile,
include/asterisk/json.h, res/res_stasis_http_recordings.c
(added), res/stasis_http.make (added), tests/test_strings.c,
res/res_stasis_http_endpoints.c (added),
res/res_stasis_http_events.c (added), include/asterisk/http.h,
Makefile, main/json.c, res/res_stasis_http.exports.in (added),
rest-api-templates (added): This patch adds a RESTful HTTP
interface to Asterisk. The API itself is documented using
Swagger, a lightweight mechanism for documenting RESTful API's
using JSON. This allows us to use swagger-ui to provide
executable documentation for the API, generate client bindings in
different languages, and generate a lot of the boilerplate code
for implementing the RESTful bindings. The API docs live in the
rest-api/ directory. The RESTful bindings are generated from the
Swagger API docs using a set of Mustache templates. The code
generator is written in Python, and uses Pystache. Pystache has
no dependencies, and be installed easily using pip. Code
generation code lives in rest-api-templates/. The generated code
reduces a lot of boilerplate when it comes to handling HTTP
requests. It also helps us have greater consistency in the REST
API. (closes issue ASTERISK-20891) Review:
https://reviewboard.asterisk.org/r/2376/
2013-04-22 12:45 +0000 [r386211] Olle Johansson <oej@edvina.net>
* include/asterisk/srv.h: Fix mistake in Doxygen. Doxygen is only
*ONE* comment that applies to the NEXT piece of code.
2013-04-22 01:05 +0000 [r386190] Russell Bryant <russell@russellbryant.com>
* apps/app_meetme.c: sla: remove redundant locking. sla.lock was
already locked in the only place that sla_check_reload() was
called. Remove the redundant locking of sla.lock done in this
function. Less recursive locking is A Good Thing.
2013-04-19 22:27 +0000 [r386160] Matthew Jordan <mjordan@digium.com>
* /, res/res_timing_pthread.c: Prevent res_timing_pthread from
blocking callers There were several reports of deadlock when
using res_timing_pthread. Backtraces indicated that one thread
was blocked waiting for the write to the pipe to complete and
this thread held the container lock for the timers. Therefore any
thread that wanted to create a new timer or read an existing
timer would block waiting for either the timer lock or the
container lock and deadlock ensued. This patch changes the way
the pipe is used to eliminate this source of deadlocks: 1) The
pipe is placed in non-blocking mode so that it would never block
even if the following changes someone fail... 2) Instead of
writing bytes into the pipe for each "tick" that's fired the pipe
now has two states--signaled and unsignaled. If signaled, the
pipe is hot and any pollers of the read side filedescriptor will
be woken up. If unsigned the pipe is idle. This eliminates even
the chance of filling up the pipe and reduces the potential
overhead of calling unnecessary writes. 3) Since we're tracking
the signaled / unsignaled state, we can eliminate the exta poll
system call for every firing because we know that there is data
to be read. (closes issue ASTERISK-21389) Reported by: Matt
Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches:
0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch
uploaded by sruffell (License 5417) (closes issue ASTERISK-19754)
Reported by: Nikola Ciprich (closes issue ASTERISK-20577)
Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported
by: Henry Fernandes (closes issue ASTERISK-17467) Reported by:
isrl (closes issue ASTERISK-17458) Reported by: isrl Review:
https://reviewboard.asterisk.org/r/2441/ ........ Merged
revisions 386109 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 386159 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-19 05:20 +0000 [r386019-386054] David M. Lee <dlee@digium.com>
* main/cli.c, /: cli.c: Properly initialize debug_modules and
verbose_modules. This avoids some lock errors on the core set
{debug,verbose} commands. ........ Merged revisions 386049 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 386051 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_http_websocket.c, include/asterisk/http_websocket.h:
Allow WebSocket connections on more URL's This patch adds the
concept of ast_websocket_server to res_http_websocket, allowing
WebSocket connections on URL's more more than /ws. The existing
funcitons for managing the WebSocket subprotocols on /ws still
work, so this patch should be completely backward compatible.
(closes issue ASTERISK-21279) Review:
https://reviewboard.asterisk.org/r/2453/
* main/message.c, /: Fix lock errors on startup. In messages.c,
there are several places in the code where we create a
tmp_tech_holder and pass that into an ao2_find call.
Unfortunately, we weren't initializing the rwlock on the
tmp_tech_holder, which the hash function was locking. It's
apparently harmless, but still not the best code. This patch
extracts all that copy/pasted code into two functions,
msg_find_by_tech and msg_find_by_tech_name, which properly
initialize and destroy the rwlock on the tmp_tech_holder. Review:
https://reviewboard.asterisk.org/r/2454/ ........ Merged
revisions 386006 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-16 23:44 +0000 [r385939] Alec L Davis <sivad.a@paradise.net.nz>
* res/res_xmpp.c, res/res_jabber.c, /: res_xmpp and res_jabber need
to search 'cachable' in the attrib section of the received IE,
not data. (issue ASTERISK-20175) (closes issue ASTERISK-21429)
(closes issue ASTERISK-21069) (closes issue ASTERISK-21164)
Reported by: alecdavis Tested by: alecdavis alecdavis (license
585) Review https://reviewboard.asterisk.org/r/2452/
2013-04-16 17:50 +0000 [r385860-385886] Kinsey Moore <kmoore@digium.com>
* res/res_corosync.c: Allow res_corosync to build
ast_enable_distributed_devstate is no longer applicable to how
the distributed device state system works and is no longer
necessary.
* main/pbx.c, funcs/func_presencestate.c,
include/asterisk/presencestate.h, main/presencestate.c: Move
presence state distribution to Stasis-core Convert presence state
events to Stasis-core messages and remove redundant serializers
where possible. Review: https://reviewboard.asterisk.org/r/2410/
(closes issue ASTERISK-21102) Patch-by: Kinsey Moore
<kmoore@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, main/ccss.c,
include/asterisk/xmpp.h, tests/test_devicestate.c,
main/devicestate.c, res/res_xmpp.c, apps/app_queue.c,
res/res_jabber.c, main/asterisk.c: Move device state distribution
to Stasis-core In the move from Asterisk's event system to
Stasis, this makes distributed device state aggregation
always-on, removes unnecessary task processors where possible,
and collapses aggregate and non-aggregate states into a single
cache for ease of retrieval. This also removes an intermediary
step in device state aggregation. Review:
https://reviewboard.asterisk.org/r/2389/ (closes issue
ASTERISK-21101) Patch-by: Kinsey Moore <kmoore@digium.com>
2013-04-16 14:09 +0000 [r385835] David M. Lee <dlee@digium.com>
* include/asterisk/stasis_channels.h: Fixed a typo
2013-04-15 17:26 +0000 [r385782] Jason Parker <jparker@digium.com>
* Makefile, /: Don't unnecessarily rebuild things on every run of
'make'. Review: https://reviewboard.asterisk.org/r/2449/ ........
Merged revisions 385745 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385768 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-15 16:47 +0000 [r385718-385743] David M. Lee <dlee@digium.com>
* res/res_stasis_websocket.c: Avoid unused variable warning when
not in devmode
* main/json.c, include/asterisk/stasis_channels.h,
res/res_stasis.exports.in (added), apps/Makefile,
apps/app_stasis.exports.in (removed), apps/stasis_json.c
(removed), main/stasis_channels.c, tests/test_app_stasis.c
(removed), res/res_stasis.c (added), main/manager_channels.c,
apps/app_stasis.c, tests/test_json.c, res/res_stasis_websocket.c,
tests/test_res_stasis.c (added), tests/test_stasis_channels.c,
include/asterisk/app_stasis.h (removed),
include/asterisk/stasis_app.h (added), include/asterisk/json.h:
Moved core logic from app_stasis to res_stasis After some
discussion on asterisk-dev, it was decided that the bulk of the
logic in app_stasis actually belongs in a resource module instead
of the application module. This patch does that, leaves the app
specific stuff in app_stasis, and fixes up everything else to be
consistent with that change. * Renamed test_app_stasis to
test_res_stasis * Renamed app_stasis.h to stasis_app.h * This is
still stasis application support, even though it's no longer in
an app_ module. The name should never have been tied to the type
of module, anyways. * Now that json isn't a resource module
anymore, moved the ast_channel_snapshot_to_json function to
main/stasis_channels.c, where it makes more sense. Review:
https://reviewboard.asterisk.org/r/2430/
* apps/app_stasis.c, main/manager_channels.c, main/channel.c,
include/asterisk/cli.h, include/asterisk/strings.h: DTMF events
are now published on a channel's stasis_topic. AMI was refactored
to use these events rather than producing the events directly in
channel.c. Finally, the code was added to app_stasis to produce
DTMF events on the WebSocket. The AMI events are completely
backward compatible, including sending events on transmitted
DTMF, and sending DTMF start events. The Stasis-HTTP events are
somewhat simplified. Since DTMF start and DTMF send events are
generally less useful, Stasis-HTTP will only send events on
received DTMF end. (closes issue ASTERISK-21282) (closes issue
ASTERISK-21359) Review: https://reviewboard.asterisk.org/r/2439
* apps/app_saycounted.c, channels/sip/security_events.c,
contrib/realtime/mysql/voicemail_messages.sql, BSDmakefile,
contrib/realtime/mysql/voicemail_data.sql,
build_tools/sha1sum-sh, res/res_mutestream.c,
configs/res_curl.conf.sample, tests/test_func_file.c,
res/res_rtp_multicast.c, include/asterisk/select.h,
include/asterisk/bridging_technology.h,
include/asterisk/bridging_features.h, tests/test_locale.c,
doc/Makefile, tests/test_poll.c,
contrib/realtime/mysql/musiconhold.sql, res/res_timing_kqueue.c,
contrib/realtime/mysql/queue_log.sql,
channels/sip/include/security_events.h, channels/sig_ss7.c,
channels/chan_multicast_rtp.c, channels/sig_ss7.h, /,
tests/test_expr.c: Fix the svn:keywords property on several
files. Normally I think keyword expansion is silly, but the one
time it would have been good, it didn't work because the property
had quotes in it. This patch fixes obviously busted svn:keywords
properties. ........ Merged revisions 385683 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385689 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-14 03:01 +0000 [r385635-385638] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_multicast.c, /: Calculate the timestamp for outbound
RTP if we don't have timing information This patch calculates the
timestamp for outbound RTP when we don't have timing information.
This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches. (closes issue
ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro
Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded
by tzafrir (License 5035) rtp-timestamp.patch uploaded by
pbertera (License 5943) ........ Merged revisions 385636 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385637 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_alsa.c: Don't attempt to create a voice frame on
a read error Prior to this patch, a read error in snd_pcm_readi
would still be treated as a nominal result when constructing a
voice frame from the expected data. Since the value returned is
negative, as opposed to the number of samples read, this could
result in a crash. With this patch, we now return a null frame
when a read error is detected. Note that the patch on
ASTERISK-21329 was modified slightly for this commit, in that we
bail immediately on detecting the read error, rather than
bypassing the construction of the voice frame. (closes issue
ASTERISK-21329) Reported by: Keiichiro Kawasaki patches:
chan_alsa.diff uploaded by kawasaki (License 6489) ........
Merged revisions 385633 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385634 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-12 22:38 +0000 [r385595] Michael L. Young <elgueromexicano@gmail.com>
* /, apps/app_queue.c: Fix Manager Segfault When app_queue Is
Unloaded When app_queue is unloaded, some manager commands are
not being unregistered which result in a segfault. This patch
corrects this. (closes issue ASTERISK-21397) Reported by: Peter
Katzmann, Corey Farrell Tested by: Corey Farrell Patches:
asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L.
Young (license 5026)
asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2444/
........ Merged revisions 385593 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385594 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-12 22:26 +0000 [r385585] Kinsey Moore <kmoore@digium.com>
* /, codecs/codec_resample.c: Allow codec_resample to be unloaded
Ensure that trans_size is correct to prevent uninitialized
entries from preventing reload. (closes issue ASTERISK-21401)
Reported by: Corey Farrell Tested by: Corey Farrell Patches:
codec_resample-unload.patch uploaded by Corey Farrell ........
Merged revisions 385582 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-12 22:22 +0000 [r385573] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_voicemail.c, /: Fix app_voicemail Segfault And A Few
Memory Leaks The original report was that app_voicemail would
crash. This was caused by ast_config_load() returning
CONFIG_STATUS_FILEINVALID but no checks being performed for that
return status. After adding the initial patch to fix this issue,
Jaco Kroon (jkroon) added some fixes to memory leaks he had
discovered. During review, Walter Doekes (wdoekes) suggested
adding a helper function in order to determine if we had a valid
configuration or not. This patch does the following: * Creates a
helper function to check if the configuration is valid * Adds
calls to the new helper function where appropiate * Fixes memory
leaks where the code returned without running
ast_config_destroy() on the configuration that was loaded (closes
issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco
Kroon, Michael L. Young Patches:
asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon
(license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2443/ ........ Merged
revisions 385551 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385557 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-12 21:48 +0000 [r385548] Jason Parker <jparker@digium.com>
* include/asterisk/sorcery.h: Fix documentation.
2013-04-12 21:11 +0000 [r385522] Kinsey Moore <kmoore@digium.com>
* include/asterisk/manager.h, main/manager_channels.c: Expose
channel snapshot manager blob generation These functions are
already used in one branch (jrose's parking branch) and will soon
be used in other branches as well.
2013-04-12 15:06 +0000 [r385474] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix One-Way Audio With auto_* NAT
Settings When SIP Calls Initiated By PBX When we reload Asterisk
or chan_sip, the flags force_rport and comedia that are turned on
and off when using the auto_force_rport and auto_comedia nat
settings go back to the default setting off. These flags are
turned on when needed or off when not needed at the time that a
peer registers, re-registers or initiates a call. This would
apply even when only the default global setting
"nat=auto_force_rport" is being used, which in this case would
only affect the force_rport flag. Everything is good except for
the following: The nat setting is set to auto_force_rport and
auto_comedia. We reload Asterisk and the peer's registration has
not expired. We load in the settings for the peer which turns
force_rport and comedia back to off. Since the peer has not
re-registered or placed a call yet, those flags remain off. We
then initiate a call to the peer from the PBX. The force_rport
and comedia flags stay off. If NAT is involved, we end up with
one-way audio since we never checked to see if the peer is behind
NAT or not. This patch does the following: * Moves the checking
of whether a peer is behind NAT into its own function * Create a
function to set the peer's NAT flags if they are using the auto_*
NAT settings * Adds calls in sip_request_call() to these new
functions in order to setup the dialog according to the peer's
settings (closes issue ASTERISK-21374) Reported by: Michael L.
Young Tested by: Michael L. Young Patches:
asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2421/
........ Merged revisions 385473 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-12 08:52 +0000 [r385406-385431] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_iax2.c, /: IAX2 defer_full_frames fail to get sent
Ensure iax2_process_thread is signalled when a deferred frame is
queued to it. (closes issue ASTERISK-18827) Reported by:
alecdavis Tested by: alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2426/ ........ Merged
revisions 385429 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385430 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_iax2.c: IAX2, prevent network thread starting
before all helper threads are ready On startup, it's possible for
a frame to arrive before the processing threads were ready. In
iax2_process_thread() the first pass through falls into
ast_cond_wait, should a frame arrive before we are at
ast_cond_wait, the signal will be ignored. The result
iax2_process_thread stays at ast_cond_wait forever, with deferred
frames being queued. Fix: When creating initial idle
iax2_process_threads, wait for init_cond to be signalled after
each thread is started. (issue ASTERISK-18827) Reported by:
alecdavis Tested by: alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2427/ ........ Merged
revisions 385402 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385403 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-11 16:53 +0000 [r385277-385314] Richard Mudgett <rmudgett@digium.com>
* /, configs/cli_aliases.conf.sample: Fix 'pri intense debug span'
alias. ........ Merged revisions 385313 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/features.c: Eliminated dial_features_destroy() since it is
equivalent to ast_free_ptr()
* main/manager.c, main/features.c: * Fix unlocked accesses to
feature_list. The feature_list is now also protected by the
features_lock. * Made all calls to ast_find_call_feature() have
the features_lock held. * Fixed set_config_flags() to actually
use find_group() to look for feature groups in DYNAMIC_FEATURES.
The code originally assumed all feature groups were listed in
DYNAMIC_FEATURES. * Make everyone use ast_rdlock_call_features(),
ast_unlock_call_features(), and new ast_wrlock_call_features()
instead of directly calling the rwlock API on features_lock.
2013-04-10 15:34 +0000 [r385236] David M. Lee <dlee@digium.com>
* main/stasis_channels.c: Fixed manager channelvars support. For
the events that have been ported to Stasis, this was broken in
r384910, when a couple of lines of code was lost in a merge.
2013-04-10 14:26 +0000 [r385174-385202] Matthew Jordan <mjordan@digium.com>
* /, res/res_config_ldap.c: Use LDAP memory management functions
instead of Asterisk's When MALLOC_DEBUG is enabled with
res_config_ldap, issues (munmap_chunk: invalid pointer errors)
can occur as the memory is being allocated with Asterisk's
wrappers around malloc/calloc/free/strdup, as opposed to the LDAP
library's wrappers. This patch uses the LDAP library's wrappers
where appropriate, so that compiling with MALLOC_DEBUG doesn't
cause more problems than it solves. Note that the patch listed
below was modified slightly for this commit to account for some
additional memory allocation/deallocations. (closes issue
ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham
patches: issue18789-1.8-r316873.patch uploaded by seanbright
(License 5060) ........ Merged revisions 385190 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385199 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: Fix crash in chan_sip when a core
initiated op occurs at the same time as a BYE When a BYE request
is processed in chan_sip, the current SIP dialog is detached from
its associated Asterisk channel structure. The tech_pvt pointer
in the channel object is set to NULL, and the dialog persists for
an RFC mandated period of time to handle re-transmits. While this
process occurs, the channel is locked (which is good).
Unfortunately, operations that are initiated externally have no
way of knowing that the channel they've just obtained (which is
still valid) and that they are attempting to lock is about to
have its tech_pvt pointer removed. By the time they obtain the
channel lock and call the channel technology callback, the
tech_pvt is NULL. This patch adds a few checks to some channel
callbacks that make sure the tech_pvt isn't NULL before using it.
Prime offenders were the DTMF digit callbacks, which would crash
if AMI initiated a DTMF on the channel at the same time as a BYE
was received from the UA. This patch also adds checks on
sip_transfer (as AMI can also cause a callback into this
function), as well as sip_indicate (as lots of things can queue
an indication onto a channel). Review:
https://reviewboard.asterisk.org/r/2434/ (closes issue
ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions
385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 385173 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-09 19:58 +0000 [r385142] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Rename struct feature_ds to struct
feature_datastore. Because "struct feature_ds *feature_ds" is not
a good thing.
2013-04-09 18:22 +0000 [r385116] David M. Lee <dlee@digium.com>
* apps/app_stasis.c: Backported app_stasis fix from stasis-http
branch. The hash and compare functions for the control container
was reusing the wrong ones, causing some problems. I fixed it,
but in the wrong branch. Oh well, it happens.
2013-04-09 06:16 +0000 [r385088] Russell Bryant <russell@russellbryant.com>
* main/features.c, CHANGES: Add inheritance support to
FEATURE()/FEATUREMAP(). The settings saved on the channel for
FEATURE()/FEATUREMAP() were only for that channel. This patch
adds the ability to have these settings inherited to child
channels if you set FEATURE(inherit)=yes. Closes issue
ASTERISK-21306. Review: https://reviewboard.asterisk.org/r/2415/
2013-04-08 23:38 +0000 [r385049] Rusty Newton <rnewton@digium.com>
* /, configs/extconfig.conf.sample: Modified the list of keys for
the driver backends for sake of sample clarity Added a line
showing the mapping of "mysql" to res_config_mysql available in
add-ons. We used "mysql" as an example driver key in the sample,
but didn't show what module it mapped too. Also added a subtitle
above the list of keys for driver backends. ........ Merged
revisions 385047 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 385048 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-08 18:24 +0000 [r384989] Walter Doekes <walter+asterisk@wjd.nu>
* build_tools/make_buildopts_h,
build_tools/make_linker_version_script, Makefile,
build_tools/mkpkgconfig, build_tools/make_version: Clean up
Makefile "warning" clutter when makeopts doesn't exist. Review:
https://reviewboard.asterisk.org/r/2304
2013-04-08 15:38 +0000 [r384910-384942] Matthew Jordan <mjordan@digium.com>
* res/res_http_websocket.c, res/res_stasis_websocket.c: Don't
attempt a websocket protocol removal if res_http_websocket isn't
there This patch sets the protocols container provided by
res_http_websocket to NULL when the module gets unloaded and adds
the necessary checks when adding/ removing a websocket protocol.
This prevents some FRACKing on an invalid pointer to the disposed
container if a module that uses res_http_websocket is unloaded
after it.
* apps/app_stasis.c, main/manager_channels.c, apps/app_dial.c,
main/pbx.c, main/channel_internal_api.c,
tests/test_stasis_channels.c (added),
include/asterisk/app_stasis.h, apps/app_userevent.c,
include/asterisk/channel.h, CHANGES, main/channel.c, main/dial.c,
include/asterisk/stasis_channels.h (added), main/features.c,
apps/stasis_json.c, pbx/pbx_realtime.c, main/stasis_channels.c
(added): Add multi-channel Stasis messages; refactor Dial AMI
events to Stasis This patch does the following: * A new Stasis
payload has been defined for multi-channel messages. This payload
can store multiple ast_channel_snapshot objects along with a
single JSON blob. The payload object itself is opaque; the
snapshots are stored in a container keyed by roles. APIs have
been provided to query for and retrieve the snapshots from the
payload object. * The Dial AMI events have been refactored onto
Stasis. This includes dial messages in app_dial, as well as the
core dialing framework. The AMI events have been modified to send
out a DialBegin/DialEnd events, as opposed to the subevent type
that was previously used. * Stasis messages, types, and other
objects related to channels have been placed in their own file,
stasis_channels. Unit tests for some of these objects/messages
have also been written.
2013-04-08 13:27 +0000 [r384879] David M. Lee <dlee@digium.com>
* main/json.c, res/res_stasis_websocket.c (added), main/frame.c,
apps/Makefile, tests/test_abstract_jb.c,
apps/app_stasis.exports.in (added), apps/stasis_json.c (added),
include/asterisk/app_stasis.h (added), include/asterisk/json.h,
include/asterisk/localtime.h, tests/test_app_stasis.c (added),
include/asterisk/frame.h, apps/app_stasis.c (added),
tests/test_json.c: Stasis application WebSocket support This is
the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support. This module registers a dialplan function
named Stasis, which is used to put a channel into the named
Stasis app. As a channel enters and leaves the Stasis diaplan
application, the Stasis app receives a 'stasis-start' and
'stasis-end' events. Stasis apps register themselves using the
stasis_app_register and stasis_app_unregister functions. Messages
are sent to an application using stasis_app_send. Finally, Stasis
apps control channels through the use of the stasis_app_control
object, and the family of stasis_app_control_* functions. Other
changes along for the ride are: * An ast_frame_dtor function
that's RAII_VAR safe * Some common JSON encoders for name/number,
timeval, and context/extension/priority Review:
https://reviewboard.asterisk.org/r/2361/
2013-04-06 16:00 +0000 [r384857] Joshua Colp <jcolp@digium.com>
* tests/test_sorcery_astdb.c (added), res/res_sorcery_astdb.c
(added): Add a res_sorcery_astdb module which uses the astdb to
persist objects. Review: https://reviewboard.asterisk.org/r/2420/
2013-04-05 20:41 +0000 [r384828] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c, UPGRADE-11.txt: Fix For Not Overriding
The Default Settings In chan_sip The initial report was that the
"nat" setting in the [general] section was not having any effect
in overriding the default setting. Upon confirming that this was
happening and looking into what was causing this, it was
discovered that other default settings would not be overriden as
well. This patch works similar to what occurs in build_peer(). We
create a temporary ast_flags structure and using a mask, we
override the default settings with whatever is set in the
[general] section. In the bug report, the reporter who helped to
test this patch noted that the directmedia settings were being
overriden properly as well as the nat settings. This issue is
also present in Asterisk 1.8 and a separate patch will be applied
to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young Patches:
asterisk-21225-handle-options-default-prob_v4.diff Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2385/ ........ Merged
revisions 384827 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-04 18:15 +0000 [r384696-384760] Richard Mudgett <rmudgett@digium.com>
* main/event.c: Separate some event struct definitions from
instantiation.
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
UPGRADE.txt: chan_dahdi: Change inband_on_proceeding option
default to no/disabled. (issue ASTERISK-21151)
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, /, channels/sig_pri.c:
chan_dahdi: Add inband_on_proceeding compatibility option. The
new inband_on_proceeding option causes Asterisk to assume inband
audio may be present when a PROCEEDING message is received. Q.931
Section 5.1.2 says the network cannot assume that the CPE side
has attached to the B channel at this time without explicitly
sending the progress indicator ie informing the CPE side to
attach to the B channel for audio. However, some non-compliant
ISDN switches send a PROCEEDING without the progress indicator ie
indicating inband audio is available and assume that the CPE
device has connected the media path for listening to ringback and
other messages. ASTERISK-17834 which causes this issue was
dealing with a non-compliant network switch. (closes issue
ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett
........ Merged revisions 384685 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 384689 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-03 17:17 +0000 [r384642] Matthew Jordan <mjordan@digium.com>
* funcs/func_channel.c, /: Update documentation for CHANNEL
function Document that you can read/write the 'accountcode' and
'amaflags' on a channel. ........ Merged revisions 384640 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 384641 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-03 16:01 +0000 [r384616] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c: astobj2: Fix rbtree duplicate handling.
OBJ_PARTIAL_KEY searching a rbtree did not find all possible
matches if the container did not accept duplicates. Added
matching node bias to indicate which matching node is being
searched for: first, last, any.
2013-04-02 17:35 +0000 [r384546] David M. Lee <dlee@digium.com>
* Makefile, /: Fixed spurious rebuilds of func_version.
func_version.so was being rebuilt every time, because build.h was
changing every build, because of the cleantest dependency that
was added in r384410 to fix parallel make bugs. Now build.h will
only be created if it does not exist, which was the original
behavior of the Makefile. ........ Merged revisions 384544 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 384545 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-02 12:18 +0000 [r384518] Joshua Colp <jcolp@digium.com>
* main/sorcery.c: Pass the object type name to the configuration
framework.
2013-04-02 11:40 +0000 [r384514] Matthew Jordan <mjordan@digium.com>
* main/xmldoc.c, include/asterisk/app.h: Make things work again
Sorry folks. ',' are still greater than '|'. Thanks for playing
along :-)
2013-04-01 20:10 +0000 [r384488] David M. Lee <dlee@digium.com>
* contrib/scripts/install_prereq: install_prereq: Build jansson
from source, when necessary When r383579 was committed, it made
Jansson a required dependency. While libjansson-dev and
jansson-devel are available on recent distros, some older (but
still supported) distros don't have it. There's a pull request[1]
to get it into repoforge, but that still doesn't help everyone.
(And helps no one until the pull request is merged and packages
are built). This patch adds Jansson install from source to the
install_unpackaged() function. There are a few gotcha's, which
makes this change not completely trivial. * Since Jansson may be
installed by a package, don't install from source if a package
installation can be found * libresample may also be installed via
package, so I added a similar check to that. * Since Jansson
installs into /usr/local, this patch also adds /usr/local/lib to
/etc/ld.so.conf.d so that the library can be found. * The
alternative was to install into /usr, but then it gets
complicated having to deal with EL's /usr/lib{32,64} shenanigans.
[1]: https://github.com/repoforge/rpms/pull/250 Review:
https://reviewboard.asterisk.org/r/2414/
2013-04-01 14:44 +0000 [r384452] Matthew Jordan <mjordan@digium.com>
* main/xmldoc.c, include/asterisk/app.h: Make appropriate items
parse using '|' instead of ',' This patch fixes a bug introduced
in r76703, wherein Asterisk could only parse arguments in the
so-called 'recommended' way, e.g., NoOp(foo,bar). The proper
syntax of NoOp,foo|bar is now parsed correctly.
2013-04-01 14:10 +0000 [r384416] Joshua Colp <jcolp@digium.com>
* /, apps/app_voicemail.c: Remove silly use of strncmp. ........
Merged revisions 384414 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-04-01 13:37 +0000 [r384412-384413] David M. Lee <dlee@digium.com>
* main/stasis.c, tests/test_stasis.c: stasis: Fixed message
ordering issues when forwarding This patch fixes an issue of
message ordering that occurs when multiple topics are forwarded
to an aggregator topic (such as ast_channel_topic_all()). It is
(very reasonably) expected that the rules governing message
dispatch order still apply, so long as the messages start from
the same thread, and are received by the same subscription.
Because the existing code had an additional layer of dispatching
via the Stasis thread pool for forwards, those promises couldn't
be kept. Forwarding subscriptions no longer have their own
mailbox, and now dispatch directly from the forwarding topic's
stasis_publish() call. This means that the topic's lock is held
for the duration of not only a message's dispatch, but the
dispatch of all the forwards. This shouldn't be a problem right
now, but if an aggregator topic had many subscribers, it could
become a problem. But I figure we can write more clever code when
the time comes, if necessary. Review:
https://reviewboard.asterisk.org/r/2419/
* /, Makefile: Fix parallel make problems. Occasionally, make -j
would fail due to missing includes, or other unusual errors. This
was due to the 'cleantest' target, which was designed to force a
make clean when some change in the code would cause the typical
depedency checking to fail. Several targets in the main Makefile
did not depend upon cleantest, hence would run in parallel to it.
By adding the dependency, make -j runs happily now. Review:
https://reviewboard.asterisk.org/r/2418/ ........ Merged
revisions 384410 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 384411 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-30 05:15 +0000 [r384389-384390] Matthew Jordan <mjordan@digium.com>
* main/manager.c: Properly format an intmax_t value
* include/asterisk/test.h, main/manager.c, main/test.c,
apps/app_voicemail.c: Convert TestEvent AMI events over to Stasis
Core This patch migrates the TestEvent AMI events to first be
dispatched over the Stasis-Core message bus. This helps to
preserve the ordering of the events with other events in the AMI
system, such as the various channel related events.
2013-03-29 16:37 +0000 [r384327] Jonathan Rose <jrose@digium.com>
* apps/app_voicemail.c: app_voicemail: Add blank argument to
externnotify if no context argument At least one call to
run_externnotify provides a NULL context parameter and because
the snprintf statement doesn't account for a NULL context
parameter, it simply writes '(null)' to the arguments string
instead. This patch makes it write two quotes back to back for
that argument instead in the event of a NULL context. (closes
issue ASTERISK-18207) Reported by: Barry L. Kline Patches:
modified from patch-20130306 uploaded by Karsten Wemheuer
(License 5930) ........ Merged revisions 384325 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 384326 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-28 23:59 +0000 [r384302] Richard Mudgett <rmudgett@digium.com>
* main/sorcery.c, main/stasis.c, main/uuid.c,
res/res_calendar_exchange.c, res/res_sorcery_config.c,
include/asterisk/uuid.h, tests/test_uuid.c: Add uuid wrapper API
call ast_uuid_generate_str(). * Updated test_uuid.c to test the
new API call. * Made system use the new API call to eliminate
"10's of lines" where used. * Fixed untested ast_strdup() return
in stasis_subscribe() by eliminating the need for it. struct
stasis_subscription now contains the uniqueid[] string. * Fixed
some issues in exchangecal_write_event(): Create uid with enough
space for a UUID string to avoid a realloc. Fix off by one error
if the calendar event provided a UUID string. There is no need to
check for NULL before calling ast_free().
2013-03-28 15:45 +0000 [r384219-384261] Kinsey Moore <kmoore@digium.com>
* include/asterisk/stasis.h, main/app.c, pbx/pbx_realtime.c,
include/asterisk/channel.h, tests/test_stasis.c,
main/manager_channels.c, main/stasis.c, apps/app_voicemail.c,
main/channel.c, main/pbx.c, main/stasis_cache.c: Break the world.
Stasis message type accessors should now all be named correctly.
* main/app.c, res/res_xmpp.c, channels/chan_iax2.c,
channels/sig_pri.c, res/res_jabber.c, channels/chan_mgcp.c,
channels/chan_unistim.c, channels/chan_dahdi.c,
include/asterisk/app.h, channels/chan_sip.c,
channels/chan_skinny.c: Convert MWI state message type to the new
stasis naming convention
2013-03-27 21:52 +0000 [r384201] David M. Lee <dlee@digium.com>
* include/asterisk/app.h, include/asterisk/stasis.h,
include/asterisk/channel.h: Added a doxygen group for Stasis
messages and topics
2013-03-27 19:52 +0000 [r384164] Kinsey Moore <kmoore@digium.com>
* main/format_pref.c, /, channels/chan_sip.c: Address uninitialized
conditional that valgrind found ........ Merged revisions 384162
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 384163 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-27 18:52 +0000 [r384120] Matthew Jordan <mjordan@digium.com>
* /, main/http.c: Fix a file descriptor leak in off nominal path
While looking at the security vulnerability in ASTERISK-20967,
Walter noticed a file descriptor leak and some other issues in
off nominal code paths. This patch corrects them. Note that this
patch is not related to the vulnerability in ASTERISK-20967, but
the patch was placed on that issue. (closes issue ASTERISK-20967)
Reported by: wdoekes patches:
issueA20967_file_leak_and_unused_wkspace.patch uploaded by
wdoekes (License 5674) ........ Merged revisions 384118 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 384119 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-27 17:07 +0000 [r384050] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_asterisk.c, /: Fix white noise on SRTP decryption
When res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both
endpoints depending on the call legs involved). The test now
properly checks the version field in the RTP header to ensure
that RTP and RTCP are decrypted while other types of packets are
not. (closes issue ASTERISK-21323) Reported by: andrea Tested by:
Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff
uploaded by Kinsey Moore ........ Merged revisions 384048 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 384049 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-27 15:27 +0000 [r383975-384019] Matthew Jordan <mjordan@digium.com>
* channels/sip/include/sip.h, /, channels/chan_sip.c,
channels/sip/security_events.c: AST-2013-003: Prevent username
disclosure in SIP channel driver When authenticating a SIP
request with alwaysauthreject enabled, allowguest disabled, and
autocreatepeer disabled, Asterisk discloses whether a user exists
for INVITE, SUBSCRIBE, and REGISTER transactions in multiple
ways. The information is disclosed when: * A "407 Proxy
Authentication Required" response is sent instead of a "401
Unauthorized" response * The presence or absence of additional
tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)")
* A "401 Unauthorized" response is sent instead of "403
Forbidden" response after a retransmission * Retransmission are
sent when a matching peer did not exist, but not when a matching
peer did exist. This patch resolves these various vectors by
ensuring that the responses sent in all scenarios is the same,
regardless of the presence of a matching peer. This issue was
reported by Walter Doekes, OSSO B.V. A substantial portion of the
testing and the solution to this problem was done by Walter as
well - a huge thanks to his tireless efforts in finding all the
ways in which this setting didn't work, providing automated
tests, and working with Kinsey on getting this fixed. (closes
issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes,
kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes
(License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes
(License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes
(License 6273, 5674) ........ Merged revisions 384003 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/http.c: AST-2013-002: Prevent denial of service in HTTP
server AST-2012-014, fixed in January of this year, contained a
fix for Asterisk's HTTP server for a remotely-triggered crash.
While the fix put in place fixed the possibility for the crash to
be triggered, a denial of service vector still exists with that
solution if an attacker sends one or more HTTP POST requests with
very large Content-Length values. This patch resolves this by
capping the Content-Length at 1024 bytes. Any attempt to send an
HTTP POST with Content-Length greater than this cap will not
result in any memory allocation. The POST will be responded to
with an HTTP 413 "Request Entity Too Large" response. This issue
was reported by Christoph Hebeisen of TELUS Security Labs (closes
issue ASTERISK-20967) Reported by: Christoph Hebeisen patches:
AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
AST-2013-002-10.diff uploaded by mmichelson (License 5049)
AST-2013-002-11.diff uploaded by mmichelson (License 5049)
........ Merged revisions 383978 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_format_attr_h264.c, /: AST-2013-001: Prevent buffer
overflow through H.264 format negotiation The format attribute
resource for H.264 video performs an unsafe read against a media
attribute when parsing the SDP. The value passed in with the
format attribute is not checked for its length when parsed into a
fixed length buffer. This patch resolves the vulnerability by
only reading as many characters from the SDP value as will fit
into the buffer. (closes issue ASTERISK-20901) Reported by: Ulf
Harnhammar patches: h264_overflow_security_patch.diff uploaded by
jrose (License 6182) ........ Merged revisions 383973 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-27 07:24 +0000 [r383948] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Fix skinny encall button to not blind
xfer. The softbutton endcall should not turn a transfer into a
blind transfer but hangup the exten being called and leave the
original call on hold. This does that. (closes issue
ASTERISK-21321) Reported by: wedhorn Tested by: snuffy, myself
Patches: skinny-xferendcall01.diff uploaded by wedhorn (license
5019)
2013-03-26 23:34 +0000 [r383925] Joshua Colp <jcolp@digium.com>
* main/sorcery.c: Remove the noop handler from sorcery so it does
not produce an empty value.
2013-03-26 02:30 +0000 [r383841-383879] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Resolve deadlock between SIP registration
and channel based functions In r373424, several reentrancy
problems in chan_sip were addressed. As a result, the SIP channel
driver is now properly locking the channel driver private
information in certain operations that it wasn't previously. This
exposed two latent problems either in register_verify or by
functions called by register_verify. This includes: * Holding the
private lock while calling sip_send_mwi_to_peer. This can create
a new sip_pvt via sip_alloc, which will obtain the channel
container lock. This is a locking inversion, as any channel
related lock must be obtained prior to obtaining the SIP channel
technology private lock. Note that this issue was already fixed
in Asterisk 11. * Holding the private lock while calling
sip_poke_peer. In the same vein as sip_send_mwi_to_peer,
sip_poke_peer can create a new SIP private, causing the same
locking inversion. Note that this locking inversion typically
occured when CLI commands were run while a SIP REGISTER request
was being processed, as many CLI commands (such as 'sip show
channels', 'core show channels', etc.) have to obtain the channel
container lock. (issue ASTERISK-21068) Reported by: Nicolas
Bouliane (issue ASTERISK-20550) Reported by: David Brillert
(issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue
ASTERISK-21296) Reported by: Gabriel Birke ........ Merged
revisions 383863 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383878 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/cdr.c, /: Resolve deadlock between pending CDR and batch CDR
locks r375757 attempted to resolve a race condition between
multiple submissions of CDRs while in batch mode from attempting
to destroy the scheduled batch submission by extending the batch
CDR lock. Unfortunately, this causes a deadlock between the
pending CDR lock and the batch CDR lock. This patch resolves the
intent of r375757 by simply providing a new lock that protects
the scheduling of the batches. The original batch CDR lock is
kept to protect manipulation of the batch CDR settings, but has
been placed such that it is not held when the pending lock is
held. Thanks to Chase Venters for providing lock analysis on the
issue. (issue ASTERISK-21162) Reported by: Chase Venters ........
Merged revisions 383839 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383840 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-26 01:46 +0000 [r383837-383838] Russell Bryant <russell@russellbryant.com>
* channels/chan_skinny.c: Suppress compiler warning. This code
caused a compiler warning when --enable-dev-mode was not used.
The warning was that this variable was set but not used. That was
indeed the case as the only place this is used is as an argument
to SKINNY_DEBUG which is compiled out when not in dev mode.
* /, apps/app_meetme.c: Fix multi-station answer race condition.
When an SLA trunk is ringing (inbound call on the trunk) Asterisk
will make outbound calls to the stations that have that trunk. If
more than one station answers the call at the same time, all
channels other than the first one to answer are left in a bad
state. The channel gets leaked, is not connected to anything, and
there's no way to get rid of it. We now properly clean up these
losing channels by hanging up on them. Since they lost the race,
as we process their answer, there is no ringing trunk for them to
answer. ........ Merged revisions 383835 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383836 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-25 23:25 +0000 [r383799] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Set the CALLERID(dnid-num-plan) for
incoming ISDN calls. The CALLEDTON channel variable is set for
incoming ISDN calls to the lower 7 bits of the Q.931
type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan)
should have the same value. (closes issue ASTERISK-21248)
Reported by: rmudgett ........ Merged revisions 383796 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383798 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-25 20:15 +0000 [r383753-383754] Kinsey Moore <kmoore@digium.com>
* main/manager_channels.c: Fix typo
* main/stasis.c: Fix missing ' ' around '='
2013-03-25 19:28 +0000 [r383726-383747] David M. Lee <dlee@digium.com>
* contrib/scripts/install_prereq: install_prereq: removed some
out-of-date comments
* contrib/scripts/install_prereq: install_prereq: Adding
jansson-devel to RH packages
* main/channel_internal_api.c, include/asterisk/channel.h, CHANGES,
main/manager_channels.c, main/channel.c, main/manager.c: Move
NewCallerid, HangupRequest and SoftHangupRequest to Stasis
HangupRequest and SoftHangupRequest are now ast_channel_blob
Stasis messages, with the cause code as an optional field in the
blob. NewCallerid now simply watches for changes in the callerid
information in channel snapshots, and creates the AMI event
appropriately. Since the original NewCallerid event honored the
channelvars setting in manager.conf, the channel variables
configured there had to become a part of the channel snapshot.
These are now a part of every snapshot based event, making the
configuration description "every time a channel-oriented event is
emitted" less of a lie. There a a few other changes wrapped up in
here as well. * When ast_channel_topic() is given NULL for a
channel, it returns the ast_channel_topic_all() topic instead of
NULL. This can clean up a lot of NULL checking we're doing
currently. * The fields Cause and Cause-txt were removed from the
base channel information and put only on the Hangup events, since
those fields are meaningless outside of a Hangup event. * Removed
the pipe-delimiter processing of the channelvars field, since
that's been deprecated forever. (closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/
2013-03-25 12:38 +0000 [r383669] Sean Bright <sean@malleable.com>
* res/res_config_curl.c, /: Properly delimit post data in
res_config_curl. ........ Merged revisions 383667 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383668 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-22 20:51 +0000 [r383633] David M. Lee <dlee@digium.com>
* main/json.c, main/Makefile: Fixed another issue from r383579.
Core modules don't honor <depend> flags in MODULEINFO, which
broke jansson if specified --with-jansson to configure.
2013-03-22 20:43 +0000 [r383632] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_mixmonitor.c, /: Fix StopMixMonitor Hanging Up When
Unable To Stop MixMonitor On A Channel A regression was
accidentally introduced when allowing an optional ID to be used
when calling StopMixMonitor. When we are unable to stop
MixMonitor on a channel, -1 is being returned which triggers the
hangup of the channel. This patch restores the prior behavior by
returning 0 whether we were successful or not. It also allows the
call from the manager to use the return code when the action
fails. (closes issue ASTERISK-21294) Reported by: daroz Tested
by: daroz Patches: asterisk-21294-stop_mixmonitor_hangingup.diff
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2404/ ........ Merged
revisions 383631 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-22 19:26 +0000 [r383579-383611] David M. Lee <dlee@digium.com>
* main/asterisk.c, main/json.c, include/asterisk/json.h: Corrected
some module issues introduced by r383579. When I moved res_json.c
to json.c, I left the MODULE_INFO stuff in there, which was
interesting if you ran module show. I also forgot to call what
was in module_load() from asterisk main().
* pbx/pbx_realtime.c, main/manager_channels.c (added),
tests/test_json.c, res/res_json.c (removed), main/pbx.c,
include/asterisk/autoconfig.h.in, configure.ac,
apps/app_userevent.c, include/asterisk/channel.h, CHANGES,
include/asterisk/manager.h, main/channel.c, main/json.c (added),
main/manager.c, configure, res/res_json.exports.in (removed):
Move more channel events to Stasis; move res_json.c to
main/json.c. This patch started out simply as fixing the bouncing
tests introduced in r382685, but required some other changes to
give it a decent implementation. To fix the bouncing tests, the
UserEvent and Newexten AMI events needed to be refactored to
dispatch via Stasis. Dispatching directly to AMI resulted in
those events sometimes getting ahead of the associated Newchannel
events, which would understandably confuse anyone. I found that
instead of creating a zillion different message types and
structures associated with them, it would be preferable to define
a message type that has a channel snapshot and a blob of
structured data with a small bit of additional information. The
JSON object model provides a very nice way of representing
structured data, so I went with that. * Move JSON support from
res_json.c to main/json.c * Made libjansson-dev a required
dependency * Added an ast_channel_blob message type, which has a
channel snapshot and JSON blob of data. * Changed UserEvent and
Newexten events so that they are dispatched via ast_channel_blob
messages on the channel's topic. * Got rid of the
ast_channel_varset message; used ast_channel_blob instead. *
Extracted the manager functions converting Stasis channel events
to AMI events into manager_channel.c. (issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
2013-03-22 06:32 +0000 [r383560] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Fix skinny voicemail indication issues.
Unsubscribe from MWI stasis event on channel reload. (closes
issue ASTERISK-21216) Reported by: wedhorn Tested by: snuffy,
myself Patches: skinny-mwiind02.diff uploaded by snuffy (license
5024)
2013-03-21 20:09 +0000 [r383541] David M. Lee <dlee@digium.com>
* include/asterisk/stasis.h: Corrected doc error for Stasis. I
guess the mutex isn't necessary. Thanks, rmudgett!
2013-03-21 17:41 +0000 [r383519] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h: Fix astobj2 doxygen comment.
2013-03-20 20:27 +0000 [r383458-383462] Walter Doekes <walter+asterisk@wjd.nu>
* funcs/func_curl.c, /: Have func_curl log a warning when a curl
request fails. Review: https://reviewboard.asterisk.org/r/2403/
........ Merged revisions 383460 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383461 from
http://svn.asterisk.org/svn/asterisk/branches/11
* funcs/func_curl.c, /: Minor cleanup in func_curl near hashcompat
code. Review: https://reviewboard.asterisk.org/r/2402/ ........
Merged revisions 383457 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-20 16:01 +0000 [r383422] Kinsey Moore <kmoore@digium.com>
* main/stasis.c: Resolve a race condition in Stasis Because of the
way that topics were handled when publishing, it was possible to
dispatch a message to a subscription after that subscription had
been unsubscribed such that the dispatched message arrived at the
callback after the callback had received its final message. In
callbacks that cleaned up user data, this would often cause a
segfault. This has been resolved by locking the topic during the
entirety of dispatch. To prevent long publishing and topic
locking times, forwarding subscriptions have been made to be
standard subscriptions instead of mailboxless subscriptions which
were dispatched at publishing time.
2013-03-20 14:52 +0000 [r383405] Joshua Colp <jcolp@digium.com>
* main/sorcery.c, res/res_sorcery_memory.c,
include/asterisk/sorcery.h, tests/test_sorcery.c: Pass the
sorcery instance to wizards for CUD operations as well as
retrieve.
2013-03-19 19:07 +0000 [r383377] Kinsey Moore <kmoore@digium.com>
* main/stasis_message_router.c: Fix lock destruction/unlock
inversion When using scoped locks, the unref of an AO2 object
should happen after the unlock occurs which requires usage of
scoped refs.
2013-03-19 16:00 +0000 [r383343] David M. Lee <dlee@digium.com>
* codecs/Makefile, /: Multiple revisions 383341-383342 ........
r383341 | dlee | 2013-03-19 10:57:29 -0500 (Tue, 19 Mar 2013) | 5
lines Removed codecs/g722/*.i on make clean ........ Merged
revisions 383340 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
r383342 | dlee | 2013-03-19 10:58:33 -0500 (Tue, 19 Mar 2013) | 1
line Remove codecs/speex/*.i on make clean ........ Merged
revisions 383341-383342 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-16 16:00 +0000 [r383284-383287] Kinsey Moore <kmoore@digium.com>
* res/res_jabber.c, channels/chan_mgcp.c: Make sure things
compile...
* channels/sip/include/sip.h, main/asterisk.c,
channels/chan_mgcp.c, apps/app_voicemail.c,
channels/chan_unistim.c, channels/chan_sip.c,
include/asterisk/stasis.h, res/res_xmpp.c, channels/sig_pri.c,
channels/chan_iax2.c, res/res_jabber.c, main/stasis.c,
channels/sig_pri.h, main/channel.c, include/asterisk/app.h,
channels/chan_dahdi.c, channels/chan_skinny.c,
include/asterisk/xmpp.h, apps/app_minivm.c, main/app.c:
Transition MWI to Stasis-core Remove MWI's dependency on the
event system by moving it to Stasis-core. This also introduces
forwarding topic pools in Stasis-core which aggregate many
dynamically allocated topics into a single primary topic. Review:
https://reviewboard.asterisk.org/r/2368/ (closes issue
ASTERISK-21097) Patch-by: Kinsey Moore
2013-03-16 15:40 +0000 [r383267-383283] Joshua Colp <jcolp@digium.com>
* res/res_xmpp.c, CHANGES: Add support for using XMPP buddy state
via device state. This change allows you to use XMPP buddy state
in places where device state can be used be used, such as
dialplan hints. If at least one resource is available the buddy
is considered available. Now your phone can reflect their IM
status too!
* res/res_xmpp.c, /: Fix a bug where resources were not found due
to hashing on the priority itself. ........ Merged revisions
383266 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-15 17:35 +0000 [r383225-383242] David M. Lee <dlee@digium.com>
* main/stasis_cache.c, main/stasis_message_router.c (added),
main/stasis_message.c, include/asterisk/stasis_message_router.h
(added), tests/test_stasis.c, main/stasis.c: A simplistic router
for stasis_message's. Often times, when subscribing to a topic,
one wants to handle different message types differently. While
one could cascade if/else statements through the subscription
handler, it is much cleaner to specify a different callback for
each message type. The stasis_message_router is here to help! A
stasis_message_router is constructed for a particular
stasis_topic, which is subscribes to. Call
stasis_message_router_unsubscribe() to cancel that subscription.
Once constructed, routes can be added using
stasis_message_router_add() (or
stasis_message_router_set_default() for any messages not handled
by other routes). There may be only one route per
stasis_message_type. The route's callback is invoked just as if
it were a callback for a subscription; but it only gets called
for messages of the specified type. (issue ASTERISK-20887)
Review: https://reviewboard.asterisk.org/r/2390/
* configs/stasis_core.conf.sample (added): Sample config file for
stasis-core. (issue ASTERISK-20887)
2013-03-15 13:04 +0000 [r383167-383169] Kinsey Moore <kmoore@digium.com>
* tests/test_stasis.c, main/manager.c, main/channel_internal_api.c:
Take advantage of the fact that stasis_unsubscribe now returns
NULL
* main/stasis.c, main/stasis_cache.c, include/asterisk/stasis.h:
Make stasis unsubscription functions return NULL Unsubscribing
things in Asterisk seems to very commonly follow with NULLing out
the variable that was unsubscribed. This change makes that a bit
simpler.
* main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
main/http.c: tcptls: Prevent unsupported options from being set
AMI, HTTP, and chan_sip all support TLS in some way, but none of
them support all the options that Asterisk's TLS core is capable
of interpreting. This prevents consumers of the TLS/SSL layer
from setting TLS/SSL options that they do not support. This also
gets tlsverifyclient closer to a working state by requesting the
client certificate when tlsverifyclient is set. Currently, there
is no consumer of main/tcptls.c in Asterisk that supports this
feature and so it can not be properly tested. Review:
https://reviewboard.asterisk.org/r/2370/ Reported-by: John
Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........
Merged revisions 383165 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383166 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-15 01:38 +0000 [r383122-383126] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: When a session timer expires during a
T.38 call, re-invite with correct SDP When a session timer
expires during a dialog that has re-negotiated to T.38 and
Asterisk is the refresher, Asterisk will send a re-INVITE with an
SDP containing audio media only. This causes some hilarity with
the poor fax session under weigh. This patch corrects that by
sending T.38 parameters if we are in the middle of a T.38
session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal
patches:
dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch
uploaded by nbansal (License 6418) ........ Merged revisions
383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 383125 from
http://svn.asterisk.org/svn/asterisk/branches/11
* pbx/pbx_spool.c, /: Fix processing of call files when using
KQueue on OS X In certain situations, call files are not
processed when using KQueue with pbx_spool. Asterisk was sending
an invalid timeout value when the spool directory is empty,
causing the call to kevent to error immediately. This can create
a tight loop, increasing the CPU load on the system. (closes
issue ASTERISK-21176) Reported by: Carlton O'Riley patches:
kqueue_osx.patch uploaded by coriley (License 6473) ........
Merged revisions 383120 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383121 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-14 16:57 +0000 [r383063] Jason Parker <jparker@digium.com>
* /, autoconf/ast_ext_lib.m4: Fix whitespace in AST_EXT_LIB_CHECK
macro. ........ Merged revisions 383061 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 383062 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-13 14:39 +0000 [r383008] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c: Always set the RTP instance data in the
RTP engine Not informing the RTP engine of the instance data
creates shrapnel.
2013-03-12 22:43 +0000 [r382989] Andrew Latham <lathama@gmail.com>
* res/res_config_ldap.c: Update Doxygen Push some cleanups upstream
before testing another ticket. (issue ASTERISK-20259)
2013-03-12 21:19 +0000 [r382941-382954] Michael L. Young <elgueromexicano@gmail.com>
* addons/res_config_mysql.c, /: Fix Sorting Order For Parking Lots
Stored In Static Realtime When retrieving the parking lots from a
MySQL database table, the current order is "filename, cat_metric
desc, var_metric asc, category". If there are multiple parking
lots with the same cat_metric but different categories,
everything is being sorted on cat_metric first resulting in
errors when loading the parking lots. This patch fixes the
problem by sorting on the category field first, then the
cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex
Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young
(license 5026) ........ Merged revisions 382942 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382943 from
http://svn.asterisk.org/svn/asterisk/branches/11
* contrib/realtime/mysql/sippeers.sql, /,
contrib/realtime/postgresql/realtime.sql: Update Contributed
Realtime Schema Files - IPv6 Addresses This commit updates some
fields in the contributed realtime schema files to handle IPv6
addresses. (closes issue ASTERISK-21173) Reported by: Torrey
Searle Patches: realtime_sql.patch Torrey Searle (license 5334)
asterisk-21173-update-ip-fields.diff Michael L. Young (license
5026) ........ Merged revisions 382939 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382940 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-12 20:07 +0000 [r382924] Joshua Colp <jcolp@digium.com>
* res/res_xmpp.c, /: Fix a crash when res_xmpp is configured using
a username without a domain. (closes issue ASTERISK-21156)
Reported by: amsoft2001 ........ Merged revisions 382923 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-12 19:08 +0000 [r382900] Jason Parker <jparker@digium.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac, res/Makefile,
CHANGES, makeopts.in, res/pjproject (removed),
res/res_rtp_asterisk.c: Switch to using external pjproject
libraries. ICE/STUN/TURN support in res_rtp_asterisk is also now
optional.
2013-03-12 16:30 +0000 [r382852] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Include the Username field in SIP
Registry events when Status is registered In ASTERISK-17888, the
AMI Registry event during SIP registrations was supposed to
include the Username field. Somehow, one of the events was
missed. This patch corrects that - the Username field should be
included in all AMI Registry events involving SIP registrations.
(issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by:
Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov
(license 6479) ........ Merged revisions 382847 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382848 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-12 08:55 +0000 [r382828] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: Fix core dump on CLI usage Fix issue
with 'unistim show info' CLI command when device connected not
configured ........ Merged revisions 382827 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-11 15:22 +0000 [r382787] Kevin Harwell <kharwell@digium.com>
* CHANGES, channels/sip/include/sip.h, channels/chan_sip.c: Added
an option to disallow music on hold Added an option
"discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event. This essentially stops
telling the peer to start music on hold. (issue ABE-2899)
Reported by: Denis Alberto Martinez Review:
https://reviewboard.asterisk.org/r/2336/
2013-03-09 00:21 +0000 [r382764] Richard Mudgett <rmudgett@digium.com>
* apps/confbridge/include/conf_state.h,
apps/confbridge/conf_state_multi.c, apps/app_confbridge.c,
apps/confbridge/conf_state_multi_marked.c,
apps/confbridge/conf_state_empty.c, apps/confbridge/conf_state.c,
apps/confbridge/conf_config_parser.c,
apps/confbridge/conf_state_single.c,
apps/confbridge/conf_state_inactive.c,
apps/confbridge/conf_state_single_marked.c,
apps/confbridge/include/confbridge.h: confbridge: Rename items
for clarity and consistency. struct conference_bridge_user ->
struct confbridge_user struct conference_bridge -> struct
confbridge_conference struct conference_state -> struct
confbridge_state struct conference_bridge_user
*conference_bridge_user -> struct confbridge_user *user struct
conference_bridge_user *cbu -> struct confbridge_user *user
struct conference_bridge *conference_bridge -> struct
confbridge_conference *conference The names are now generally
shorter, consistently used, and don't conflict with the struct
names. This patch handles the renaming part of the issue. (issue
ASTERISK-20776) Reported by: rmudgett
2013-03-08 20:26 +0000 [r382746] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: chan_sip: Update the via header when
relaying SMS MESSAGE Prior to this change, certain conditions for
sending the message would result in an address of '(null)' being
used in the via header of the SIP message because a NULl value of
pvt->ourip was used when initially generating the via header.
This is fixed by adding a call to build_via when the address is
set before sending the message. (closes issue ASTERISK-21148)
Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch
uploaded by Zhi Cheng (license 6475) ........ Merged revisions
382739 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-08 16:59 +0000 [r382721-382724] David M. Lee <dlee@digium.com>
* main/stasis_cache.c, include/asterisk/stasis.h: Stasis
documentation updates. (issue ASTERISK-20887) (issue
ASTERISK-20959)
* main/stasis.c, main/channel.c, main/channel_internal_api.c:
Ensure dummy channels get a stasis topic. Fixes test failure
introduced in r382685. (issue ASTERISK-20887) (issue
ASTERISK-20959)
2013-03-08 16:00 +0000 [r382705] Kinsey Moore <kmoore@digium.com>
* include/asterisk/stasis.h, tests/test_stasis.c,
main/stasis_cache.c: Add message dump capability to stasis cache
layer The cache dump mechanism allows the developer to retreive
multiple items of a given type (or of all types) from the cache
residing in a stasis caching topic in addition to the existing
single-item cache retreival mechanism. This also adds to the
caching unit tests to ensure that the new cache dump mechanism is
functioning properly. Review:
https://reviewboard.asterisk.org/r/2367/ (issue ASTERISK-21097)
2013-03-08 15:15 +0000 [r382685] David M. Lee <dlee@digium.com>
* include/asterisk/channel.h, tests/test_stasis.c (added),
main/asterisk.c, main/stasis.c (added), main/channel.c,
main/stasis_cache.c (added), main/pbx.c, main/stasis_message.c
(added), main/manager.c, main/asterisk.exports.in,
include/asterisk/channel_internal.h, main/channel_internal_api.c,
include/asterisk/stasis.h (added): This patch adds a new message
bus API to Asterisk. For the initial use of this bus, I took some
work kmoore did creating channel snapshots. So rather than create
AMI events directly in the channel code, this patch generates
Stasis events, which manager.c uses to then publish the AMI
event. This message bus provides a generic publish/subscribe
mechanism within Asterisk. This message bus is: - Loosely
coupled; new message types can be added in seperate modules. -
Easy to use; publishing and subscribing are straightforward
operations. In addition to basic publish/subscribe, the patch
also provides mechanisms for message forwarding, and for message
caching. (issue ASTERISK-20887) (closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/
2013-03-08 04:11 +0000 [r382670-382671] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Remove unused function After r382670,
get_ip_and_port_from_sdp was no longer used.
* channels/chan_sip.c: Don't reset the RTP address on a glare
re-INVITE Originally, way back in r201583, we added the alternate
RTP address so that the RTP engine would expect to receive audio
from a new source when a glare re-INVITE occurred. In r382589, we
remove the alternate RTP source, as the 'secret' probation mode
allows for switching to a new RTP source when a previous source
stops sending RTP. At the time, it seemed appropriate to set the
RTP source based on the information in the glared re-INVITE.
Unfortunately, that doesn't work so well - in a glared re-INVITE
that occurs with no SDP - such as in a connected line update that
glances - we'll set the RTP source to an invalid address. In
subsequent re-INVITE requests from this Asterisk instance, we'll
then send an invalid media address, which will result in the
remote side sending a 488. Whoops. There isn't any need to reset
the RTP source - if we're using strictrtp, we'll simply
synchronize to a new source when we stop getting packets from the
old one. If we aren't using strictrtp, then again there shouldn't
be a problem. Note that the Asterisk Test Suite's connectedline
test caught this error.
2013-03-07 21:55 +0000 [r382648] David M. Lee <dlee@digium.com>
* main/threadpool.c: Changing log level of "Not changing threadpool
size" from notice to debug.
2013-03-07 21:14 +0000 [r382636] Jason Parker <jparker@digium.com>
* res/res_sorcery_config.c, res/res_sorcery_memory.c: Load sorcery
modules earlier, so they can actually be used.
2013-03-07 19:14 +0000 [r382621] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c, /: Let vm_mailbox_snapshot combine "Urgent"
when no folder is specified r381835 fixed a bug in
vm_mailbox_snapshot where combining INBOX and Old forgot that
Urgent also "counts" as new messages. This fixed the problem when
any of the three folders was specified and the combine option was
used. It missed the case where the folder isn't specified and we
build a snapshot of all folders. This patch corrects that.
........ Merged revisions 382617 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-07 16:48 +0000 [r382600-382604] Kinsey Moore <kmoore@digium.com>
* main/xmldoc.c: Fix a memory leak in xmldoc Another instance of
attribute retrieval not being freed properly.
* main/xmldoc.c: Resolve more memory leaks in xmldoc Many places
that allocated to pull out an attribute are now freed properly.
2013-03-07 15:48 +0000 [r382589] Matthew Jordan <mjordan@digium.com>
* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
main/rtp_engine.c, /, channels/chan_sip.c: Add a 'secret'
probation strictrtp mode to handle delayed changes in RTP source
Often, Asterisk may realize that a change in the source of an RTP
stream is about to occur and ask that the RTP engine reset it's
lock on the current RTP source. In certain scenarios, it may take
awhile for the new remote system to send RTP packets, while the
old remote system may continue providing RTP during that time
period. This causes Asterisk to re-lock onto the old source,
thereby rejecting the new source when the old source stops
sending RTP and the new source begins. This patch prevents that
by having a constant secondary, 'secret' probation mode enabled
when an RTP source has been chosen. RTP packets from other
sources are always considered, but never chosen unless the
current RTP source stops sending RTP. Review:
https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124)
Reported by: John Bigelow Tested by: John Bigelow (closes issue
AST-1125) Reported by: John Bigelow Tested by: John Bigelow
........ Merged revisions 382573 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-07 15:36 +0000 [r382489-382587] Kinsey Moore <kmoore@digium.com>
* main/xmldoc.c: Fix minor memory leak in xmldoc Strings retrieved
via ast_xml_get_text() must be freed with ast_xml_free_text().
* /, main/logger.c: Ensure that logmsgs are freed properly Messages
sent while the logger thread is shutting down will now have their
associated callid freed properly. ........ Merged revisions
382574 from http://svn.asterisk.org/svn/asterisk/branches/11
* main/threadpool.c: Fix ref leak in threadpool.c If
ast_threadpool_set_size with a size equal to the current size, a
reference to a set_size_data structure would be leaked.
* main/threadpool.c: Resolve a ref leak in threadpool.c Ownership
of the listener reference is not transferred because the listener
is reffed when placed into the taskprocessor. Ensure that the
listener is dereffed properly.
2013-03-05 13:14 +0000 [r382440] Matthew Jordan <mjordan@digium.com>
* configs/res_ldap.conf.sample,
contrib/realtime/postgresql/realtime.sql,
configs/sip.conf.sample, CHANGES,
contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif, channels/sip/include/sip.h,
CREDITS, contrib/realtime/mysql/sippeers.sql,
channels/chan_sip.c: Add RFC 3327 Path header support to chan_sip
This patch adds support for RFC 3327 "Path" headers. This can be
enabled in sip.conf using the 'supportpath' setting, either on a
global basis or on a peer basis. This setting enables Asterisk to
route outgoing out-of-dialog requests via a set of proxies by
using a pre-loaded route-set defined by the Path headers in the
REGISTER request. This patch also adds Realtime support for
dynamically updating the Path information for a peer. A huge
thank-you to Klaus Darillion and Olle E Johansson for their
efforts in writing this patch. Review:
https://reviewboard.asterisk.org/r/2235/ Review:
https://reviewboard.asterisk.org/r/991/ (closes issue
ASTERISK-16884) Reported by: klaus3000 Tested by: klaus3000, oej,
mjordan patches: path-1.8.0-patch.txt uploaded by klaus3000
(License 5054) oolong-path-support-trunk in team branch by oej
(License 5267)
2013-03-05 03:53 +0000 [r382411] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* /, channels/chan_unistim.c: Fix several unreleased mutex locks
that cause problem with processing calls Reported by: Daniel
Bohling Tested by: Daniel Bohling (Closes issue ASTERISK-21119)
........ Merged revisions 382409 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382410 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-04 21:15 +0000 [r382392] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/format_cap.h, main/bridging.c: Fixup some bridge
and format capabilities comments and whitespace.
2013-03-04 21:14 +0000 [r382391] Jason Parker <jparker@digium.com>
* /, main/event.c: Fix comparison of presence state in event
subsystem. Several new IEs were not given types (or names),
causing the comparison function to improperly succeed. This adds
those. (closes issue AST-1128) ........ Merged revisions 382390
from http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-04 20:18 +0000 [r382386] Kevin Harwell <kharwell@digium.com>
* /, apps/app_confbridge.c: Confbridge CLI new record file name
check. This fix checks to make sure that if a confbridge record
start command is issued from the CLI it will always use the file
name given on the CLI even if it changes between start/stop
records for a conference. Previously it had been reusing the same
file between start/stops even if a new filename was given. (issue
AST-1088) Reported by: John Bigelow ........ Merged revisions
382385 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-03-01 18:01 +0000 [r382340] Joshua Colp <jcolp@digium.com>
* include/asterisk/sorcery.h, tests/test_sorcery.c, main/sorcery.c:
Add support for registering a sorcery handler which supports
multiple fields using a regex. Review:
https://reviewboard.asterisk.org/r/2332/
2013-03-01 04:32 +0000 [r382323] Michael L. Young <elgueromexicano@gmail.com>
* contrib/realtime/postgresql/realtime.sql, CHANGES,
contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c: Fix
/ Clean Up Some Items To Handle The New auto_* NAT Options The
original report had to do with a realtime peer behind NAT being
pruned and the peer's private address being used instead of its
external address. Upon debugging, it was discovered that this was
being caused by the addition of the auto_force_rport and
auto_comedia settings. This patch does the following: * Adds a
missing note to the CHANGES file indicating that the default
global nat setting is auto_force_rport * Constify the 'req'
parameter for check_via() * Add calls to check_via() in a couple
of places in order for the auto_* settings to do their job in
attempting to determine if NAT is involved * Set the flags
SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
settings are in use where it was needed * Moves the copying of
peer flags up in build_peer() to before they are used; this fixes
the realtime prune issue * Update the contrib/realtime schemas to
allow the nat column to handle the different nat setting
combinations we have This patch received a review and "Ship It!"
on the issue itself. (closes issue ASTERISK-20904) Reported by:
JoshE Tested by: JoshE, Michael L. Young Patches:
asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young
(license 5026) ........ Merged revisions 382322 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-28 21:59 +0000 [r382297-382299] Joshua Colp <jcolp@digium.com>
* /, res/res_rtp_asterisk.c: While the ICE negotiation is occurring
leave strictrtp in an open state, media can and will come from
different places. ........ Merged revisions 382298 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_rtp_asterisk.c, /: Fix a bug with ICE and strictrtp where
media could get dropped. If the end result of the ICE negotiation
resulted in the path for media changing it was possible for the
strictrtp code to discard the RTP packets. This change causes
strictrtp to enter learning mode once again when the ICE
negotiation has completed successfully. ........ Merged revisions
382296 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-28 21:31 +0000 [r382294-382295] Richard Mudgett <rmudgett@digium.com>
* main/threadpool.c: threadpool: Make ast_threadpool_push() return
-1 if shutting_down
* include/asterisk/threadpool.h, main/threadpool.c: threadpool:
Whitespace and comment corrections.
2013-02-28 21:21 +0000 [r382292] Jason Parker <jparker@digium.com>
* res/res_rtp_asterisk.c, include/asterisk.h: Don't undefine
bzero()/bcopy(). This was causing build failures against external
libraries that happened to use them, unless silly hacks were
added to the modules that used those headers. Review:
https://reviewboard.asterisk.org/r/2359/
2013-02-28 17:17 +0000 [r382232-382236] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_iax2.c: Prevent deadlock in chan_iax2 when
attempting to set caller ID A deadlock can occur in chan_iax2
when it attempts to set the caller ID, as it already holds the
iax2 private lock and improperly fails to obtain the channel lock
before calling ast_set_callerid. By not safely obtaining the
channel lock, a locking inversion can take place, causing a
deadlock. This patch solves this by calling the required deadlock
avoidance functions that obtain the channel lock before setting
the caller ID. Thanks to Pavel for fixing my syntax errors and
testing this patch out. (closes issue ASTERISK-21128) Reported
by: Pavel Troller Tested by: Pavel Troller patches:
ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller
(license 6302) ........ Merged revisions 382233 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382234 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_meetme.c, CHANGES: Let channels joining a MeetMe
conference opt out of the denoiser For some channel drivers,
specifically those that have a varying rate in the number of
audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the
DENOISE function in func_speex to channels joining the
conference. The denoiser function in the speex library is
initialized with the number of audio samples in each sample that
will be provided to it. If the number of audio samples changes,
the denoiser has to be thrown away and re-initialized. While this
could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the
system. This patches does the following: * Checks for the
presence of func_speex as opposed to codec_speex when determining
if the DENOISE function is present (which is where the function
is actually implemented) * Adds an option to MeetMe 'n' that
causes the denoiser to not be applied to a channel when it joins.
This keeps the current behavior the default, but let's users
disable the denoiser if it causes problems on their system.
Review: https://reviewboard.asterisk.org/r/2358 (closes issue
AST-1062) Reported by: Thomas Arimont ........ Merged revisions
382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 382230 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-27 20:31 +0000 [r382203-382204] Richard Mudgett <rmudgett@digium.com>
* channels/chan_skinny.c: More places to eliminate the cast to argv
but were not giving warnings.
* channels/chan_skinny.c: Fix compiler warning by eliminating the
need for a cast.
2013-02-27 16:19 +0000 [r382182] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Relax dialog checking in
get_sip_pvt_byid_locked so it works when the dialog is forked.
(closes issue ASTERISK-20638) Reported by: eelcob Patches:
pedantic-call-pickup-from-tag.patch uploaded by eelcob (license
6442) ........ Merged revisions 382171 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382174 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-26 20:05 +0000 [r382113] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, configure, configure.ac: Consider linux-gnuspe as linux-gnu *
The powerpcspe Linux port uses linux-gnuspe as the OS string. *
Our build system shouldn't really care for that, so just call it
linux-gnu. * Original report: Roland Stigge ,
http://bugs.debian.org/701505 Review:
https://reviewboard.asterisk.org/r/2357/ ........ Merged
revisions 382110 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382111 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-26 19:36 +0000 [r382109] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: Correct RPID parsing for unquoted
display-name. Parsing Remote-Party-ID will now succeed if
display-name is of the *(token LWS) kind and not just the
quoted-string kind. Review:
https://reviewboard.asterisk.org/r/2341/ ........ Merged
revisions 382107 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382108 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-26 19:29 +0000 [r382106] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, main/Makefile: Remove unneeded linux-gnueabi* As of r380522
the configure scripts converts the value of linux-gnueabi* of
OSARCH to "linux-gnu". So no point in testing for those values.
........ Merged revisions 382087 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382096 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-26 15:52 +0000 [r382067-382070] Matthew Jordan <mjordan@digium.com>
* apps/app_confbridge.c, /: Clean up ConfBridge commands to account
for wait_marked users When ConfBridge was refactored to better
handle the concept of marked, wait_marked, and normal users
co-existing in a conference (thereby implementing a state machine
for the conference), the wait_marked users were put into their
own list of conference participants, separate from the active
users. This list is used for wait_marked users when they are
waiting in a conference but no marked user has joined; normal
users may have joined at this point however. There are several
AMI/CLI commands that affect conference users that were not
checking the wait_marked users list: * CLI/AMI commands that
mute/unmute a participant. In this case, wait_marked users have
to remain in their particular state and should not be affected -
however, the commands would return "Channel not found" as opposed
to the appropriate error condition. * CLI/AMI commands that kick
a participant. An admin should always be able to kick a
participant out of the conference. This patch fixes both sets of
commands, and cleans up the CLI commands slightly by allowing
them to complete a participant name (this was supposed to have
been added, but the function call was commented out and wasn't
implemented). Review: https://reviewboard.asterisk.org/r/2346/
(closes issue AST-1114) Reported by: John Bigelow Tested by: John
Bigelow ........ Merged revisions 382068 from
http://svn.asterisk.org/svn/asterisk/branches/11
* configs/confbridge.conf.sample, /,
apps/confbridge/conf_config_parser.c: Ensure that the default
bridge/user profiles are always available ConfBridge and Page
require that there always be a default bridge and user profile
available. While properties of the default profiles can be
overriden in the configuration file, removing them can create
situations where neither application can function properly. This
patch ensures that if an administrator removes the profiles from
the confbridge.conf configuration file, the profiles are added
upon load. Documentation clarifying this has been added to the
confbridge.conf.sample file. Review:
https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115)
Reported by: John Bigelow Tested by: John Bigelow ........ Merged
revisions 382066 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-25 12:51 +0000 [r382023] Matthew Jordan <mjordan@digium.com>
* addons/res_config_mysql.c, /: Clean up use of va_end/va_args in
res_config_mysql There were several problems using variadic
argument macros in res_config_mysql. * Improper use of va_end.
Multiple calls to va_end were possible resulting in an unbalanced
matching of va_start/va_end. * Calls to va_arg after a possible
encounter of a SENTINEL value. This patch corrects those errors.
(closes issue ASTERISK-19451) Reported by: wdoekes patches:
ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
........ Merged revisions 382021 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 382022 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-25 07:09 +0000 [r382007-382008] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: More called details fixup for skinny.
Basically sets the callerid and callername to the first device
talked to for the purposes of putting the the calls made log on
the device. Does not affect the device displaying who the device
is currently talking to. Also some minor changes to use
sub->exten in lieu of l->lastnumberdialed. (closes issue
ASTERISK-21095) Reported by: wedhorn Tested by: snuffy, myself
Patches: skinny-calllogsoutbound03.diff uploaded by wedhorn
(license 5019)
* channels/chan_skinny.c: Add prinotify messages to skinny. Adds
both fixed and variable prinotify messages and clearprinotify
messages to skinny. Also adds cli function for pushing messages
to devices. i Initial code by snuffy, expanded by myself to
include fixed messages. (closes issue ASTERISK-21091) Reported
by: snuffy Tested by: snuffy, myself Patches:
skinny-prinotify02.diff uploaded by wedhorn (license 5019)
2013-02-24 23:01 +0000 [r381918-381977] Matthew Jordan <mjordan@digium.com>
* channels/chan_jingle.c, /: Set the sin_family on the bind address
socket during initialization Somehow, chan_jingle has managed to
operate for years without setting the sin_family on its bindaddr
socket. This patch properly sets the field during initial module
load to AF_INET. Note that the patch on the issue was modified
slightly to change the initialization of the socket from
allocation of a chan_jingle private to the module initialization,
as the bindaddr object (which is static) only needs to have the
address set once. (closes issue ASTERISK-19341) Reported by:
andre valentin patches: 0105-chan_jingle.patch uploaded by
avalentin (License 6064) ........ Merged revisions 381975 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381976 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/manager.c, /: Don't display the AMI ALL class authorization
for users if they don't have it When converting AMI class
authorizations to a string representation, the method always
appends the ALL class authorization. This is especially important
for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as
a class authorization for itself. (Events have always assumed
that the ALL class authorization is implied when they are raised)
Unfortunately, this did mean that specifying a user with
restricted class authorizations would show up in the 'manager
show user' CLI command as having the ALL class authorization.
Rather then modifying the existing string manipulation function,
this patch adds a function that will only return a string if the
field being compared explicitly matches class authorization field
it is being compared against. This prevents ALL from being
returned unless it is actually specified for the user. (closes
issue ASTERISK-20397) Reported by: Johan Wilfer ........ Merged
revisions 381939 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381943 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_parkandannounce.c: Make ParkAndAnnounce return to
priority + 1 when return context is not defined The
ParkAndAnnounce application documentation for the optional
return_context parameter states the following: return_context The
goto-style label to jump the call back into after timeout.
Default 'priority+1'. Unfortunately, the application was sending
the channel back into the dialplan at 'priority', which is the
ParkAndAnnounce application call. This causes an infinite loop of
the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those
callers you wish to drive to the end of madness, this was not the
intent of the application. (closes issue ASTERISK-20113) Reported
by: serginuez patches: app_parkandannounce.diff uploaded by
serginuez (License 6405) ........ Merged revisions 381916 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381917 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-22 19:40 +0000 [r381894] Michael L. Young <elgueromexicano@gmail.com>
* /, res/res_agi.c: Fix FastAGI To Properly Check For A Connection
When IPv6 support was added to FastAGI, the intent was to have
the ability to check all addresses resolved for a host since we
might receive an IPv4 address and an IPv6 address. The problem
with the current code, is that, since we are doing O_NONBLOCK, we
get EINPROGRESS when calling ast_connect() but are ignoring this
instead of handling it. We break out of the loop and continue on.
When we later call ast_poll(), it succeeds but we never check if
we have a connection or not on the socket level. We then attempt
to send data to the host address that we think is setup and it
fails. We then check the errno and see that we have "connection
refused" and then return with agi failed. This patch does the
following: * Handles EINPROGRESS by creating the function
handle_connection() - ast_poll() was moved into this function -
This function checks the results of the connection on the socket
level after calling ast_poll() * Continues to the next address if
the above fails to create a connection * Once all addresses
resolved are tried and we still are unable to establish a
connection, then we return that the FastAGI call failed (closes
issue ASTERISK-21065) Reported by: Jeremy Kister Tested by:
Jeremy Kister, Michael L. Young Patches:
asterisk-21065_poll_correctly_v4.diff Michael L. Young (license
5026) Review: https://reviewboard.asterisk.org/r/2330/ ........
Merged revisions 381893 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-22 15:51 +0000 [r381881] Jonathan Rose <jrose@digium.com>
* /, apps/app_dial.c: app_dial: Honor the 'c' flag when the calling
party hangs up Apparently this feature became broken in 11,
probably as a result of the Hangup Cause project. (closes issue
ASTERISK-21113) Reprted by: Heiko Wundram Patches: app_dial.patch
uploaded by Heiko Wundram (license 5822) ........ Merged
revisions 381880 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-22 01:52 +0000 [r381869] Matthew Jordan <mjordan@digium.com>
* configure, configure.ac, /: Properly detect launchd Asterisk was
a little too pro-active in claiming that it found launchd. On
systems without launchd - such as FreeBSD - this resulted in
certain items in Asterisk that conflict with launchd to not be
selectable, such as res_timing_kqueue. (closes issue
ASTERISK-20749) Reported by: Oleg Baranov ........ Merged
revisions 381847 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381848 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-19 19:47 +0000 [r381792] Kevin Harwell <kharwell@digium.com>
* main/features.c: Write the correct callid to the data1 field in
queue_log for transfer events. The incorrect callid was being
written to the "data1" field in queue_log table for transfer
events. The callid of the queue was being written instead of the
transfer target's callid. This now gets the correct "transfer to"
number and places that in the "data1" field of the queue_log
table when a transfer event is triggered. (closes issue
ASTERISK-19960) Reported by: vladimir shmagin ........ Merged
revisions 381770 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381791 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-19 17:17 +0000 [r381749] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_motif.c, include/asterisk/module.h,
res/snmp/agent.c, main/loader.c, main/cli.c: Add The Status Of A
Module To The Output Of "CLI> module show" When a module's
configuration is not loadable, we still load the module but it is
not in a running state. When trying to troubleshoot, let's say,
why chan_motif is ignoring inbound XMPP traffic, there is no way
to indicate that a loaded module is not currently running.
(closes issue ASTERISK-21108) Reported by: Rusty Newton Tested
by: Michael L. Young Patches: asterisk-21108_add_status-v2.diff
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2331/
2013-02-19 16:23 +0000 [r381729-381741] Kevin Harwell <kharwell@digium.com>
* apps/app_confbridge.c: Confbridge channels staying active when
all participants leave. If you started/stopped recording of a
conference multiple times channels would remain active even when
all participants left the conference. This was due to the fact
that a reference to the confbridge was being added every time a
start record command was issued, but when the recording was
stopped there was no matching de-reference thus keeping the
conference alive. Made sure only a single reference is added for
the record thread no matter how many times recording is
started/stopped. A de-reference is issued upon thread ending.
Note, this issue is being fixed under AST-1088 since it relates
to it and should have been corrected along with those
modifications. (issue AST-1088) Reported by: John Bigelow
........ Merged revisions 381737 from
http://svn.asterisk.org/svn/asterisk/branches/11
* CHANGES, apps/confbridge/conf_config_parser.c,
apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
Added Confbridge record_file_append option. Currently, if one
starts, stops, and then starts a recording again for a conference
the recorded data is appended to the file originally created on
the first record start. An option record_file_append has been
added that defaults to "yes", but when set to "no" will force
creation of a new file between every record start/stop. (issue
AST-1088) Reported by: John Bigelow Review:
http://reviewboard.digium.internal/r/374/
2013-02-19 06:54 +0000 [r381717-381718] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c, configs/skinny.conf.sample: Add
serviceURL stuff to skinny. Patch adds all the packet and
structure stuff to skinny to enable setting service URLs in
skinny, such as corporate directories. This stuff is only
relevant during load/unload as when activated. Also some minor
changes removing duplicated counting of addons and speedials in
handle_skinny_show_devices. Review:
https://reviewboard.asterisk.org/r/2321/
* channels/chan_skinny.c: Fixup skinny CLI completion. Auto
complete for skinny debug allows multiple options and negation,
also add debug all option. Usage example: 'skinny debug all
-packets' (each can be autocompleted including -packet). Change
show device to use device name. Remove the duplicate ast_strdup's
from place calling device complete return immediately from
complete devicename and complete linename so that multiple
options are displayed on the CLI if more than one option
available. Review: https://reviewboard.asterisk.org/r/2333/
2013-02-18 22:23 +0000 [r381703] Kevin Harwell <kharwell@digium.com>
* apps/app_confbridge.c, /: Fixed Confbridge file recording
deadlock and appending. A deadlock occurred after
starting/stopping and then restarting a confbridge recording.
Upon starting a recording a record thread is created that holds a
lock until just before exiting. Stopping the recording does not
stop/exit the thread or release the lock. The thread waits until
recording begins again. Starting a stopped recording signals the
thread to continue and start recording again. However restarting
the recording also created another record thread resulting in a
deadlock. The fix was to make sure the record thread was only
created once. Also it was noted that filenames for the recordings
were being concatenated for each start/stop. This was fixed by
creating a new file for each conference session and appending the
actual recorded data within the file (e.g. passing the 'a' option
to MixMonitor). (issue AST-1088) Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/ ........ Merged
revisions 381702 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-18 20:31 +0000 [r381670] Walter Doekes <walter+asterisk@wjd.nu>
* configs/sip.conf.sample, /: Remove "registertrying" and add
"rtp_engine" from/to sip.conf.sample The "registertrying" option
was removed in r343220. The "rtp_engine" option was added in
r186078 but erroneously named "engine" in the sample. Note that
there is no global sip setting for a different engine. ........
Merged revisions 381668 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381669 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-18 19:48 +0000 [r381656] Jonathan Rose <jrose@digium.com>
* funcs/func_presencestate.c, /: PRESENCE_STATE: Provide better
documentation for the 'e' option. Notes that the 'e' option
actually decodes data when used as a write function such as with
the SET application while it encodes data when used to read.
Review: https://reviewboard.asterisk.org/r/2335/ ........ Merged
revisions 381655 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-18 19:12 +0000 [r381644] Richard Mudgett <rmudgett@digium.com>
* apps/app_confbridge.c: confbridge: Add flags column to CLI
"confbridge list <conference>" * Added the following flags to the
CLI "confbridge list <conference>" output: A - The user is an
admin M - The user is a marked user W - The user must wait for a
marked user to join E - The user will be kicked after the last
marked user leaves the conference w - The user is waiting for a
marked user to join * Added the following header to the AMI
ConfbridgeList events: WaitMarked, EndMarked, and Waiting.
(closes issue AST-1101) Reported by: John Bigelow Patches:
confbridge-show-admin3.txt (license #5091) patch uploaded by John
Bigelow Modified
2013-02-16 20:44 +0000 [r381628] Richard Mudgett <rmudgett@digium.com>
* apps/app_confbridge.c: confbridge: Rename i iterator variables to
iter.
2013-02-16 16:28 +0000 [r381615] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Don't send presencestate information if
the state is invalid Previously, presencestate information was
sent whenever the state was not NOT_SET. When r381594 actually
returned INVALID presence state in all the places it was supposed
to, it caused chan_sip to start adding presence state information
to NOTIFY requests that it previously would not have added.
chan_sip shouldn't be adding presence state information when the
provider is in an invalid state; users can't set the state to
invalid and an invalid state always implies that the provider is
in an error condition. (issue AST-1084) ........ Merged revisions
381613 from http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-16 16:24 +0000 [r381614] Joshua Colp <jcolp@digium.com>
* res/res_sorcery_memory.c, include/asterisk/sorcery.h,
tests/test_sorcery.c, main/sorcery.c, res/res_sorcery_config.c:
Add support for retrieving multiple objects from sorcery using a
regex on their id. Review:
https://reviewboard.asterisk.org/r/2329/
2013-02-15 23:29 +0000 [r381595] Matthew Jordan <mjordan@digium.com>
* /, main/presencestate.c, funcs/func_presencestate.c,
main/manager.c: Fix crash in PresenceState AMI action when
specifying an invalid provider This patch fixes a crash in
Asterisk that could be caused by using the PresenceState AMI
action while providing an invalid provider. This patch also adds
some additional warnings when a user attempts to provide the
PresenceState action with invalid data, and removes some NOTICE
statements that were still lurking in the code from testing.
(closes issue AST-1084) Reported by: John Bigelow Tested by: John
Bigelow ........ Merged revisions 381594 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-15 18:51 +0000 [r381568] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix a crash that occurred when a BYE was
received on a replaced dialog. Reference counting for the channel
and its tech_pvt got messed up at some point between 1.8 and 11.
The result was that if a BYE for a dialog that had been replaced
(via an INVITE with Replaces) was received, Asterisk would crash
due to trying to access data on a channel that was no longer
there. The fix I introduced is to remove code that both unrefs
the sip_pvt and sets the channel's tech_pvt to NULL when an
INVITE with Replaces is handled. This way when a BYE is received,
the tech_pvt will be non-NULL and so the BYE can be processed and
not cause a crash. (closes issue ASTERISK-20929) reported by
Kristopher Lalletti patches: ASTERISK-20929.patch uploaded by
Mark Michelson (License #5049) ........ Merged revisions 381566
from http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-15 18:44 +0000 [r381567] Matthew Jordan <mjordan@digium.com>
* include/asterisk/sorcery.h, main/config_options.c,
main/sorcery.c: Disable strict XML documentation config checking;
fix crash caused by sorcery This patch does two things: 1. It
disables (temporarily) strict XML documentation checking for
module configurations. We should re-enable it before making any
release from trunk. 2. Pass the module flag AST_MODULE through
sorcery. This means several of the API calls are now macros and
will do this automatically for you. The config framework needs
the module that objects are registering to so it can properly
construct the documentation. (This was already a required field,
but sorcery was getting by without it)
2013-02-15 17:38 +0000 [r381557] Kevin Harwell <kharwell@digium.com>
* main/logger.c, include/asterisk/logger.h, main/autoservice.c:
Stopped spamming of debug messages during attended transfer.
While autoservice is running and servicing a channel the callid
is being stored and removed in the thread's local storage for
each iteration of the thread loop. If debug was set to a
sufficient level the log file would be spammed with callid thread
local storage debug messages. Added a new function that checks to
see if the callid to be stored is different than what is already
contained (if anything). If it is different then store/replace
and log, otherwise just leave as is. Also made it so all logging
of debug messages pertaining to the callid thread storage outputs
only when TEST_FRAMEWORK is defined. (issue ASTERISK-21014)
(closes issue ASTERISK-21014) Report by: Rusty Newton Review:
https://reviewboard.asterisk.org/r/2324/
2013-02-15 17:33 +0000 [r381556] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: chan_sip: Use video and text crypto
attributes to append RTP profiles to SDP Some bad copy/pasting
resulted in using the audio crypto attribute for both text and
video RTP. Also the audio crypto isn't set until after these, so
it was really just bad all around. (closes ASTERISK-20905)
Reported by: Kristopher Lalletti patches:
rtp_crypto_video_text.diff uploaded by Jonathan Rose (license
6182) ........ Merged revisions 381553 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-15 15:26 +0000 [r381527-381543] Matthew Jordan <mjordan@digium.com>
* /: Remove automerge propertrties added in r381527
* main/asterisk.c, main/xmldoc.c, main/udptl.c,
include/asterisk/xml.h, /, main/xml.c,
include/asterisk/_private.h, res/res_xmpp.c, main/named_acl.c,
configs/motif.conf.sample, apps/confbridge/conf_config_parser.c,
Makefile, include/asterisk/config_options.h,
configs/xmpp.conf.sample, apps/app_skel.c, channels/chan_motif.c,
include/asterisk/xmldoc.h, main/config_options.c,
doc/appdocsxml.dtd: Add CLI configuration documentation This
patch allows a module to define its configuration in XML in
source, such that it can be parsed by the XML documentation
engine. Documentation is generated in a two-pass approach: 1. The
documentation is first generated from the XML pulled from the
source 2. The documentation is then enhanced by the registration
of configuration options that use the configuration framework
This patch include configuration documentation for the following
modules: * chan_motif * res_xmpp * app_confbridge * app_skel *
udptl Two new CLI commands have been added: * config show help -
show configuration help by module, category, and item * xmldoc
dump - dump the in-memory representation of the XML documentation
to a new XML file. Review:
https://reviewboard.asterisk.org/r/2278 Review:
https://reviewboard.asterisk.org/r/2058 patches: on review 2058
uploaded by twilson
2013-02-14 19:58 +0000 [r381470-381471] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Remove extraneous stuff from r381470.
* channels/chan_skinny.c: Add back sending dialnumber to skinny.
Don't know why it seemed to work during testing, but it really is
needed for protocol v17 (and probably above).
2013-02-14 19:52 +0000 [r381469] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: End stuck DTMF if AST_SOFTHANGUP_ASYNCGOTO
because it isn't a real hangup. It doesn't hurt to check
AST_SOFTHANGUP_UNBRIDGE either, but it should not be set outside
of a bridge. (issue ASTERISK-20492) ........ Merged revisions
381466 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 381467 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-14 19:25 +0000 [r381465] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Respect callerid presentation in skinny.
Fix chan_skinny so that it respects callerID presentation of
inbound calls to device and a couple of other minor fixes: 145
packet (add OCTAL_FROM amd callerid), and dont send dialednumber
message if protocol >= 17. (closes issue ASTERISK-21066) Reported
by: snuffy Tested by: snuffy, myself Patches:
skinny-respect-clid-restrictions-v2.diff uploaded by snuffy
(license 5024)
2013-02-14 18:47 +0000 [r381448] Kinsey Moore <kmoore@digium.com>
* main/logger.c, include/asterisk/term.h, apps/app_queue.c,
main/asterisk.c, main/term.c, main/data.c, main/pbx.c,
main/manager.c: Revamp of terminal color codes The core module
related to coloring terminal output was old and needed some love.
The main thing here was an attempt to get rid of the obscene
number of stack-local buffers that were allocated for no other
reason than to colorize some output. Instead, this uses a simple
trick to allocate several buffers within threadlocal storage,
then automatically rotates between them, so that you can make
multiple calls to the colorization routine within one function
and not need to allocate multiple buffers. Review:
https://reviewboard.asterisk.org/r/2241/ Patches: bug.patch
uploaded by Tilghman Lesher
2013-02-14 17:06 +0000 [r381398-381427] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: Use a shuffling algorithm to find unused
IAX2 call numbers. While adding red-black tree containers to
astobj2 in r376575, Richard pointed out the way chan_iax2 finds
unused call numbers will prevent ao2_container integrity checks
at runtime. This patch removes the ao2_container and instead uses
fixed sized arrays and a modified Fisher-Yates-Durstenfeld
shuffle to maintain the call number list. While the locking
semantics are similar to the ao2_container implementation, this
implementation should be faster and more memory efficient.
Review: https://reviewboard.asterisk.org/r/2288/
* include/asterisk/doxygen/asterisk-git-howto.h: Update the name of
the update_tags utility in the git mirror how-to.
2013-02-14 03:49 +0000 [r381366] Matthew Jordan <mjordan@digium.com>
* apps/app_db.c, /: Don't throw a spurious error when using
DBdeltree The function call ast_db_deltree returns the number of
row deleted, or a negative number if it failed. DBdeltree was
treating any non-zero return as an error, causing a spurious
verbose error message to be displayed. This patch handles the
return code of ast_db_deltree correctly. (closes issue
ASTERISK-21070) Reported by: ianc patches: dbdeltree.diff
uploaded by ianc (License #5955) ........ Merged revisions 381364
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 381365 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-12 21:45 +0000 [r381326] David M. Lee <dlee@digium.com>
* tests/test_threadpool.c, tests/test_taskprocessor.c,
main/threadpool.c, main/taskprocessor.c,
include/asterisk/threadpool.h: Add a serializer interface to the
threadpool This patch adds the ability to create a serializer
from a thread pool. A serializer is a ast_taskprocessor with the
same contract as a default taskprocessor (tasks execute serially)
except instead of executing out of a dedicated thread, execution
occurs in a thread from a ast_threadpool. Think of it as a
lightweight thread. While it guarantees that each task will
complete before executing the next, there is no guarantee as to
which thread from the pool individual tasks will execute. This
normally only matters if your code relys on thread specific
information, such as thread locals. This patch also fixes a bug
in how the 'was_empty' parameter is computed for the push
callback, and gets rid of the unused 'shutting_down' field.
Review: https://reviewboard.asterisk.org/r/2323/
2013-02-12 20:57 +0000 [r381307] Mark Michelson <mmichelson@digium.com>
* main/rtp_engine.c, /: Do not allow native RTP bridging if
packetization of media streams differs. The RTP engine will no
longer allow for local and remote native RTP bridges if
packetization of streams differs. Allowing native bridging in
this scenario has been known to cause FAX failures. (closes
ASTERISK-20650) Reported by: Maciej Krajewski Patches:
ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)
Review: https://reviewboard.asterisk.org/r/2319 ........ Merged
revisions 381281 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381306 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-12 20:18 +0000 [r381285] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c, channels/sip/security_events.c,
channels/sip/include/sip.h: Fix some more REF_DEBUG-related build
errors When sip_ref_peer and sip_unref_peer were exported to be
usable in channels/sip/security_events.c, modifications to those
functions when building under REF_DEBUG were not taken into
account. This change moves the necessary defines into sip.h to
make them accessible to other parts of chan_sip that need them.
........ Merged revisions 381282 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-12 03:31 +0000 [r381256] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_confbridge.c: Adding Some More Manager Events To
ConfBridge Currently, ConfBridge does not send manager events for
ConfbridgeMute, ConfbridgeUnmute, ConfbridgeStartRecord and
ConfbridgeStopRecord. This patch adds these events to the
manager. The reporter's patch moves some other events up to the
beginning of the file. The patch being committed is based on the
patch contributed from the reporter of this issue. I have made a
lot of modifications to the patch in order for it to fit in
better with what we currently are doing in the code when it comes
to manager events. I also made a few changes to the <see-also>
elements on some of the events. (closes issue ASTERISK-20827)
Reported by: Clint Davis Tested by: Clint Davis, Michael L. Young
Patches: 20827.diff uploaded by Clint Davis (license 6453)
asterisk-20827-confbridge-events.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2309/
2013-02-11 21:17 +0000 [r381219] Kevin Harwell <kharwell@digium.com>
* /, apps/app_playback.c: Properly load say.conf upon reload of
module app_playback. If say.conf did not exists prior to
originally loading module app_playback it would not load on
subsequent reloads of the module once it had been created. This
occurred because upon reload of the app_playback module it would
only load a new configuration if an old one had previously
existed. This fix simply removed the association between checking
if an old configuration existed and the loading of the new one.
(closes issue ASTERISK-20800) Reported by: pgoergler ........
Merged revisions 381216 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 381217 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-11 21:10 +0000 [r381218] Kinsey Moore <kmoore@digium.com>
* include/asterisk/astobj2.h: Fix compilation error with REF_DEBUG
When the red/black tree work was committed, there was an extra ",
" in the REF_DEBUG definition of ao2_container_alloc_rbtree.
2013-02-11 20:39 +0000 [r381214] David M. Lee <dlee@digium.com>
* tests/test_json.c, res/res_json.c: Minor fixes to res_json and
test_json. * Made input checking more consistent with other
Asterisk code * Added validation to ast_json_dump_new_file *
Fixed tests for ownereship semantics (issue ASTERISK-20887)
2013-02-11 18:54 +0000 [r381195] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Fix some issues with skinny callid. Add
extra string to transmit_callinfo_var, Only set string2 to tonum
for outgoing calls and changes to send_callinfo and push_callinfo
to not set callid name to last number. (closes issue
ASTERISK-21063) Reported by: wedhorn Tested by: snuffy, myself
Patches: skinny-callinfoupdate03.diff uploaded by wedhorn
(license 5019)
2013-02-11 18:00 +0000 [r381177] Richard Mudgett <rmudgett@digium.com>
* main/features.c: features: Don't cache a struct ast_app pointer.
Caching a struct ast_app pointer is not a good idea because
someone could unload the application. After the applicaiton
unload the cached ast_app pointer is no longer valid. Only pbx.c
can cache the pointer because it knows when the application is
unloaded and removes the pointer. * Fixed one-touch Monitor and
MixMonitor to not cache the ast_app pointer and not use the silly
monitor_ok/mixmonitor_ok/stopmixmonitor_ok flags. * Extracted
bridge_check_monitor() from ast_bridge_call() and use propper
locking.
2013-02-11 15:11 +0000 [r381160] Matthew Jordan <mjordan@digium.com>
* /, res/res_xmpp.c: Fix crash in res_xmpp when deleting pubsub
node from CLI An error existed in res_xmpp where it would attempt
to delete attributes from a node that itself was also deleted.
Per the iksemel documentation, attributes added using iks_insert
are copied to the parent node's stack, and will be reclaimed when
that node is itself destroyed. (closes issue ASTERISK-20982)
Reported by: marcelloceschia patches: delete-node-fix.diff
uploaded by marcelloceschia (License 6036) ........ Merged
revisions 381159 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-10 14:58 +0000 [r381134] Joshua Colp <jcolp@digium.com>
* include/asterisk/sorcery.h, tests/test_sorcery.c, main/sorcery.c:
Add additional functionality to the Sorcery API. This commit adds
native implementation support for copying and diffing objects, as
well as the ability to load or reload on a per-object type level.
Review: https://reviewboard.asterisk.org/r/2320/
2013-02-09 20:58 +0000 [r381069-381118] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c: pbx: Fix regression caused by taking advantage of the
function name sort. Taking advantage of the sorted order of the
registered functions container requires that they are actually
inserted in the expected sort order. * Insert the registered
functions into the container in case sensitive position. As a
result, only the complete_functions() routine needs to search the
entire container because it does a case insensitive search for
convenience. Caught by the unit tests.
* main/pbx.c: pbx: Make function and application containers take
advantage of being sorted. * Fixed "core show function" tab
completion and token count checking. * Refactored function and
application container handling code to reduce redundancy. * Made
__ast_pbx_run() return using the defines the caller should
expect. Doesn't change the returned values. Just made use the
defines.
* include/asterisk/channel.h, main/channel.c, channels/chan_sip.c:
Make ast_do_masquerade() a void function.
* /, apps/app_confbridge.c: app_confbridge: Fix crash from
receiving an AMI action after ConfBridge unloaded. Unloading
ConfBridge caused the next AMI action received to crash Asterisk.
* Add the missing unregister of AMI action
ConfbridgeSetSingleVideoSrc when ConfBridge is unloaded. (closes
issue ASTERISK-20994) Reported by: Jeremy Kister Patches:
jira_asterisk_20994_v11.patch (license #5621) patch uploaded by
rmudgett Tested by: Rusty Newton, Jeremy Kister ........ Merged
revisions 381067 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-08 17:36 +0000 [r381068] Jonathan Rose <jrose@digium.com>
* configs/features.conf.sample, main/features.c, CHANGES: Call
Parking: Set PARKINGLOT and PARKINGSLOT variables on all parked
calls These two variables were previously not being set when
comebacktoorigin=yes and the example configs seemed to imply that
they should be. Since there is no harm in this and since calls
that are sent back to origin are capable of continuing in the
dialplan, this seemed like a no-brainer. Also it supports some
bridging tests I've been working on.
2013-02-07 17:57 +0000 [r381037] Joshua Colp <jcolp@digium.com>
* res/res_sorcery_config.c: Fix a bug where a changed configuration
file might not be available to all sorcery object types. Since
res_sorcery_config used a static name of "res_sorcery_config" to
inform the configuration file API that it asked for the
configuration file it was possible during a reload for some
sorcery object types not to receive the new configuration file.
This change introduces a UUID on a per-sorcery config instance
basis so that the unchanged state is kept on an instance basis
and not for the res_sorcery_config module as a whole.
2013-02-07 15:16 +0000 [r381017] Kinsey Moore <kmoore@digium.com>
* include/asterisk/stringfields.h, tests/test_stringfields.c: Add
aggregate operations for stuctures with string fields Add
struct-level comparison and copying of string fields to reduce
the complexity of whole-struct comparison and copying when using
string fields. The new macros do not take into account
non-stringfield data. Review:
https://reviewboard.asterisk.org/r/2308/
2013-02-06 20:18 +0000 [r380977] David M. Lee <dlee@digium.com>
* /, channels/chan_sip.c: Fixed failing test from r380696. When I
added my extensive suite of session timer unit tests, apparently
one of them was failing and I never noticed. If neither Min-SE
nor Session-Expires is set in the header, it was responding with
a Session-Expires of the global maxmimum instead of the
configured max for the endpoint. (issue ASTERISK-20787) ........
Merged revisions 380973 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380974 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-06 08:44 +0000 [r380925-380943] Damien Wedhorn <voip@facts.com.au>
* /, channels/chan_skinny.c: Fix reload skinny with active devices.
Patch ensures that d->activeline and l->activesub are moved over
to the new device and line so that on callend the appropriate
subs can be found to complete hangup before device resets.
(closes issue ASTERISK-16610) Reported by: wedhorn Tested by:
snuffy, myself Patches: skinny-reloadactive01.diff uploaded by
wedhorn (license 5019) ........ Merged revisions 380942 from
http://svn.asterisk.org/svn/asterisk/branches/11
* configs/skinny.conf.sample, channels/chan_skinny.c: Reset skinny
vmexten and immeddial char on reload. Make skinny reset vmexten
and immeddial to '\0' on reload to ensure that it is set to '\0'
if the appropriate item is removed/commented in skinny.conf. Also
small fix re immeddial char in skinny.conf and add immedial
setting to skinny show settings. (closes issue ASTERISK-21037)
Reported by: snuffy Tested by: snuffy, myself Patches:
immed_dial_fix.diff uploaded by snuffy (license 5024)
2013-02-05 19:11 +0000 [r380855-380896] Richard Mudgett <rmudgett@digium.com>
* apps/app_confbridge.c, /, apps/app_page.c: app_page and
app_confbridge: Fix custom announcement on entering conference.
The Page and ConfBridge custom announcement did not play when
users entered the conference. * Fix the
CONFBRIDGE(user,announcement) file not getting played. The code
to do this got removed accidentally when the ConfBridge code was
restructured to be more state machine like. * Fixed
play_prompt_to_user() doxygen comments. * Fixed the Page A(x) and
n options for the caller. The caller never played the
announcement file and totally ignored the n option. The code to
do this was lost when the application was converted to use
ConfBridge. * Factored out setup_profile_bridge(),
setup_profile_paged(), and setup_profile_caller() routines to
setup ConfBridge profiles. Made each profile setup routine use
the default template if one has not already been setup by
dialplan. (closes issue ASTERISK-20990) Reported by: Jeremy
Kister Tested by: rmudgett ........ Merged revisions 380894 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
error messages on exiting conference. A marked user ending a
conference with only end_marked users generates error messages:
ERROR[0000][C-00000000]: confbridge/conf_state.c:47
conf_invalid_event_fn: Invalid event for confbridge user '' * The
MULTI_MARKED state was doing too much when it was kicking out the
end_marked users from the conference. The kicked out users will
clean up after themselves when they exit the conference. (closes
issue ASTERISK-20991) Reported by: Jeremy Kister Tested by:
rmudgett ........ Merged revisions 380892 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_page.c: app_page: Fixup application XML documentation
typos and inaccuracies. ........ Merged revisions 380869 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/confbridge/conf_config_parser.c, /: Because the compiler can
check types with a struct copy and memcpy() cannot. ........
Merged revisions 380856 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/dial.c, /: Separate option_types[] from the struct
definition. Updated the option_types[] doxygen comment. ........
Merged revisions 380853 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380854 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-04 19:52 +0000 [r380817] Jason Parker <jparker@digium.com>
* /, res/Makefile, res/pjproject/build/common.mak,
res/pjproject/aconfigure, res/pjproject/build/os-auto.mak.in,
Makefile, res/pjproject/aconfigure.ac: Fix how we build
pjproject. Allow parallel builds, better tolerate failures, build
faster. This also stops running dependencies before top-level
configure has been run. (closes issue ASTERISK-20815) Review:
https://reviewboard.asterisk.org/r/2292/ ........ Merged
revisions 380816 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-02-02 01:52 +0000 [r380792] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Add variable length displayprompt packet
to skinny and use octals. Add new variable length displayprompt
packet (0x0145) to skinny. Uses the new packet if the device is
reporting protocol versions >= 17. Add the use of octal codes for
sending prompts to both the new and old displayprompt messages
(also cleaned up soft_key_template_default to use the defined
octal codes). Review: https://reviewboard.asterisk.org/r/2294/
2013-02-01 19:35 +0000 [r380774] Richard Mudgett <rmudgett@digium.com>
* channels/iax2/firmware.c: chan_iax2: Fix compile error if
MALLOC_DEBUG enabled. NEVER INCLUDE astmm.h DIRECTLY!!
2013-02-01 06:37 +0000 [r380755] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Adds variable length callinfo packets to
skinny. Add packet 0x014A (variable length call info messages) to
skinny for newer firmware. Plenty of unknown information but
includes the equivalent functionality as the fixed size callinfo
packet already included. Only send this packet if protocol
reported is >= 17. Review:
https://reviewboard.asterisk.org/r/2290/
2013-01-31 22:03 +0000 [r380738] Jason Parker <jparker@digium.com>
* res/pjproject/pjlib/src/pj/ssl_sock_ossl.c,
res/pjproject/pjlib/src/pj/log.c,
res/pjproject/pjlib/src/pj/pool_buf.c, /,
res/pjproject/pjsip-apps/src/samples/icedemo.c,
res/pjproject/pjlib/include/pj/config_site.h,
res/pjproject/pjmedia/src/test/test.c: Multiple revisions
380735-380736 ........ r380735 | qwell | 2013-01-31 15:40:09
-0600 (Thu, 31 Jan 2013) | 1 line Fix a few compiler warnings.
........ r380736 | qwell | 2013-01-31 15:42:34 -0600 (Thu, 31 Jan
2013) | 1 line Ignore warnings caused by PJ_TODO()s in pjproject.
........ Merged revisions 380735-380736 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-31 20:17 +0000 [r380699] David M. Lee <dlee@digium.com>
* /, channels/chan_sip.c: Process session timers, even if
Session-Expires header is missing Previously, Asterisk only
processed session timer information if both the 'Supported:
timer' and 'Session-Expires' headers were present. However, the
Session-Expires header is optional. If we were to receive a
request with a Min-SE greater than our configured
session-expires, we would respond with a 'Session-Expires' header
that was too small. This patch cleans the situation up a bit,
always processing timer information if the 'Supported: timer'
header is present. (closes issue ASTERISK-20787) Reported by:
Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/
........ Merged revisions 380696 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380698 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-31 19:52 +0000 [r380695] Sean Bright <sean@malleable.com>
* channels/iax2/include/firmware.h (added),
channels/iax2/include/parser.h, channels/chan_iax2.c,
channels/iax2/firmware.c (added): Move IAX firmware related
functionality into separate files. This patch is mostly a
reorganization of existing code with a few exceptions: * Added
doxygen comments to all of the extracted functions. * Split
reload_firmware(int unload) into iax_firmware_reload() and
iax_firmware_unload() for readability. * Create
iax_firmware_traverse() to support the 'iax2 show firmware' CLI
command. * Renamed iax_check_version() to
iax_firmware_get_version() and change its arguments and return
value so that it returns a success/failure value and sets the
selected version into an out parameter to avoid confusion with
failure and version 0.
2013-01-31 19:04 +0000 [r380674] Jason Parker <jparker@digium.com>
* res/pjproject/build/rules.mak,
res/pjproject/pjnath/build/Makefile,
res/pjproject/pjsip/build/Makefile, res/pjproject/aconfigure,
res/pjproject/pjsip-apps/build/Makefile,
res/pjproject/aconfigure.ac,
res/pjproject/pjmedia/build/Makefile,
res/pjproject/build/cc-auto.mak.in, /,
res/pjproject/pjlib-util/build/Makefile,
res/pjproject/pjlib/build/Makefile: Multiple revisions
380671-380673 ........ r380671 | qwell | 2013-01-31 12:59:28
-0600 (Thu, 31 Jan 2013) | 4 lines Remove a cross-compile
workaround. ar and ranlib can be easily detected with autoconf.
........ r380672 | qwell | 2013-01-31 13:00:38 -0600 (Thu, 31 Jan
2013) | 2 lines Always check for libm, regardless of configure
options. ........ r380673 | qwell | 2013-01-31 13:03:03 -0600
(Thu, 31 Jan 2013) | 7 lines Add support for parallel builds of
pjproject. Also adds proper dependency checking, and direct .a
file targets. We don't take advantage of this currently, but we
will soon. (issue ASTERISK-20815) ........ Merged revisions
380671-380673 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-31 18:22 +0000 [r380576-380666] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_multiplexed.c: bridge_multiplexed: Keep the
multiplexed thread until no more bridges use it. * Fixed the
potential of losing the multiplexed bridge thread when the last
channel leaves and another joins while the multiplexed thread is
being shut down. * Refactored and improved the management of the
serviced channels array. * Changed the channels count to a
bridges count so it only needs to be incremented rather than
changed by two.
* main/frame.c, funcs/func_frame_trace.c: Improve func FRAME_TRACE
DTMF digit format.
* include/asterisk/bridging.h: Eliminate an unused lock in
ast_bridge_channel.
* main/channel.c: Eliminate a use of a C++ keyword as a variable.
new to new_frame
* channels/iax2: Add ignore properties to channels/iax2
* include/asterisk/channel.h, /: Make CHECK_BLOCKING() debug
message more useful. Change the displayed pthread value to hex
format so it can be easily matched with CLI core show threads or
gdb. ........ Merged revisions 380611 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380612 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_dahdi.c, /: chan_dahdi: Fix "dahdi show channels
group" for groups greater than 31. The variable type used was not
large enough to hold a group bit field. ........ Merged revisions
380572 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 380575 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-30 17:49 +0000 [r380460-380522] Matthew Jordan <mjordan@digium.com>
* /, configure, configure.ac: Support building Asterisk for
Raspberry Pi/Raspbian with hard-float support Building Asterisk
on Raspbian with hard-float support fails as it uses the string
'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'.
This patch modifies the configure script for Asterisk such that
it will match on any string beginning with 'linux-gnueabi', as
opposed to requiring an explicit match. (closes issue
ASTERISK-21006) Reported by: Christian Hesse Tested by: Christian
Hesse patches: linux-gnueabihf.patch uploaded by Christian Hesse
(license 6459) linux-gnueabihf-autoconf.patch uploaded by
Christian Hesse (license 6459) ........ Merged revisions 380520
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 380521 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: Unregister SIP provider API if module
load is declined A user in #asterisk ran into a problem where a
configuration error prevented the chan_sip module from being
loaded. Upon fixing their configuratione error, they could no
longer load the chan_sip module. This was because the
configuration checking happened after the SIP provider was
registered with the Asterisk core, and subsequent attempts to
load the SIP module failed as the provider was already
registered. Since we want to detect any failure in registering
chan_sip as early as possible (as that could be emblematic of a
deeper mismatch between module and Asterisk core), this patch
does not change the registration location, but does ensure that
if a module load is declined, we unregister the module as the SIP
api provider. ........ Merged revisions 380480 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: Perform case insensitive comparisons for
T.38 attributes RFC5347 section 2.5.2 states the following: ...
The attribute "T38MaxBitRate" was once incorrectly registered
with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
T.38 examples and common implementation practice, the form
"T38MaxBitRate" SHOULD be generated by implementations conforming
to this package. In general, it is RECOMMENDED that
implementations of this package accept lowercase, uppercase, and
mixed upper/lowercase encodings of all the T.38 attributes. ...
Asterisk currently does not perform case insensitive matching on
the T.38 attributes. This causes the T38MaxBitRate attribute to
be negotiated at 2400 baud instead of 14400 (or whatever value
you actually wanted). This patch makes it so that when we compare
T.38 attributes, we do so in a case insensitive fashion. Note
that while the issue reporter did not directly write the patch,
they contributed to it (and would have provided one themselves if
the license had gone through a tad faster), and hence get
attribution for it. Review:
https://reviewboard.asterisk.org/r/2298/ (closes issue
ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill
patches: -- uploaded by Eric Hill ........ Merged revisions
380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 380465 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_calendar_icalendar.c, /: Fix memory leak in
res_calendar_icalendar The ICalendar module had a systemic memory
leak on each fetch of data from the ICalendar source. The
previous fetched data was not being properly disposed. This patch
makes it so that before each fetch of data, we dispose of the
previously fetched data. (closes issue ASTERISK-21012) Reported
by: Joel Vandal Tested by: Joel Vandal ........ Merged revisions
380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 380452 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-29 22:58 +0000 [r380433] Sean Bright <sean@malleable.com>
* channels/iax2/parser.c (added), channels/iax2 (added),
channels/iax2-parser.h (removed),
channels/iax2/include/provision.h (added), channels/iax2/include
(added), channels/iax2/include/parser.h (added), channels/iax2.h
(removed), channels/iax2-provision.c (removed),
channels/iax2/provision.c (added), channels/Makefile,
channels/chan_iax2.c, channels/iax2-parser.c (removed),
channels/iax2/include/iax2.h (added), channels/iax2-provision.h
(removed): Move the ancillary iax2 source files into a separate
sub-directory. This patch just moves the IAX2 source and header
files into a separate iax2 sub-directory in the channels
directory, similar to how the sip source files are structured.
The only thing that was added was an #ifndef to protect
provision.h from multiple inclusion.
2013-01-29 20:19 +0000 [r380407] Joshua Colp <jcolp@digium.com>
* tests/test_sorcery.c, main/sorcery.c: Fix an issue where building
with DEBUG_FD_LEAKS enabled would not work due to sorcery using
calls called "open" and "close".
2013-01-29 18:02 +0000 [r380386] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_agent.c: chan_agent: Prevent multiple channels
from logging in as the same agent. Multiple channels logging in
as the same agent can result in dead channels waiting for a
condition signal that will never come because another channel
thread stole it. A symptom is chan_sip repeatedly generating
warning messages about rescheduling autodestruction of dialogs
with an agent channel owner. * Made only login_exec() (the app
AgentLogin) clear the agent_pvt->chan pointer to prevent multiple
channels from logging in as the same agent. agent_read(),
agent_call(), and agent_set_base_channel() no longer disconnect
the agent channel from the agent_pvt. This also eliminates the
need to keep checking for agent_pvt->chan being NULL. * Made
agent_hangup() not wake up the AgentLogin agent thread until it
is done. * Made agent_request() not able to get the agent until
he has logged in and any wrapup time has expired. * Made
agent_request() use ast_hangup() instead of agent_hangup() to
correctly dispose of a channel. * Removed
agent_set_base_channel(). Nobody calls it and it is a bad thing
in general. * Made only agent_devicestate() determine the current
device state of an agent. Note: Agent group device states have
never been supported. Review:
https://reviewboard.asterisk.org/r/2260/ ........ Merged
revisions 380364 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380384 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-29 17:46 +0000 [r380383] David M. Lee <dlee@digium.com>
* channels/sip/sdp_crypto.c, /: Corrected crypto tag in SDP ANSWER
for SRTP. (again) The original fix (r380043) for getting Asterisk
to respond with the correct tag overlooked some corner cases, and
the fact that the same code is in 1.8. This patch moves the
building of the crypto line out of sdp_crypto_process(). Instead,
it merely copies the accepted tag. The call to sdp_crypto_offer()
will build the crypto line in all cases now, using a tag of "1"
in the case of sending offers. (closes issue ASTERISK-20849)
Reported by: José Luis Millán Review:
https://reviewboard.asterisk.org/r/2295/ ........ Merged
revisions 380347 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380350 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-29 17:06 +0000 [r380349] Jonathan Rose <jrose@digium.com>
* main/features.c, /: call_parking: Make sure fallbacks are used
when lacking a flat channel exten A regression was introduced
which removed automatic fallback behavior from the PBX. This
behavior was used by call parking (or at least documented as how
the feature works) in order to select an extension when the flat
channel extension wasn't available from the comebackcontext.
Parking now handles the fallbacks internally in order to keep
behavior matching with how it is documented. (closes issue
ASTERISK-20716) Reported by: Chris Gentle Review:
https://reviewboard.asterisk.org/r/2296/ ........ Merged
revisions 380348 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-29 14:48 +0000 [r380299-380332] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Ensure that a declined media stream is
terminated with a '\r\n' In r369028, chan_sip's processing of
media streams in an SDP was modified to better handle multiple
offered media streams. Part of that change modified how streams
were declined. Previously, declined media streams were not
handled in an RFC compliant manner; now, we set the port number
to 0 in the media stream definition and proceed on with the next
media stream. Unfortunately, the formatting of the declined media
stream forgot to append a '\r\n' to the end of the media stream.
This is normally added to the accepted media streams later on in
the processing of the SDP. Since the declined media stream uses a
different buffer than the accepted media streams (and is a
malloc'd buffer as opposed to a struct ast_str), it's easier to
just slap the '\r\n' on the declined media stream buffer rather
than attempt to append it later on. So, that's what we do. And
now some devices (and probably some providers) will be a bit
happier (but probably not terribly happy, since we just rejected
something they offered). Review:
https://reviewboard.asterisk.org/r/2297/ (closes issue
ASTERISK-20908) Reported by: Dennis DeDonatis Tested by: Dennis
DeDonatis ........ Merged revisions 380331 from
http://svn.asterisk.org/svn/asterisk/branches/11
* autoconf/ast_check_pwlib.m4, /, configure,
include/asterisk/autoconfig.h.in: Update configure script to be
compatible with ptlib 2.10.9 With ptlib 2.10.9, the configure
script fails due to grep returning multiple matches for the
pattern it searches for. This patch updates the pattern matching
to return only the actual version for the symbol searched for,
PTLIB_VERSION. (closes issue ASTERISK-20980) Reported by: Stefan
Reuter patches: ASTERISK-20980-1.patch uploaded by Stefan Reuter
(license 5339) ........ Merged revisions 380297 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380298 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-28 21:09 +0000 [r380256] Sean Bright <sean@malleable.com>
* /, channels/iax2.h, channels/chan_iax2.c: Correct the number of
available call numbers in IAX2. There is currently an edge case
where call number 32768 might be allocated for a call, even
though the IAX2 protocol requires call numbers be only 15 bits.
This resulted in some unpredictable behavior when call number
32678 is chosen. This patch was mostly written by Richard Mudgett
via ReviewBoard. I'm just committing it. Review:
https://reviewboard.asterisk.org/r/2293/ ........ Merged
revisions 380254 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380255 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-28 01:58 +0000 [r380209-380212] Russell Bryant <russell@russellbryant.com>
* main/file.c, /: Change cleanup ordering in filestream destructor.
This patch came about due to a problem observed where wav files
had an empty header. The header is supposed to be updated in
wav_close(). It turns out that this was broken when the
cache_record_files option from asterisk.conf was enabled. The
cleanup code was moving the file to its final destination
*before* running the close() method of the file destructor, so
the header didn't get updated. Another problem here is that the
move was being done before actually closing the FILE *. Finally,
the last bug fixed here is that I noticed that wav_close() checks
for stream->filename to be non-NULL. In the previous cleanup
order, it's checking a pointer to freed memory. This doesn't
actually cause anything to break, but it's treading on dangerous
waters. Now the free() of stream->filename is happening after the
format module's close() method gets called, so it's safer.
Review: https://reviewboard.asterisk.org/r/2286/ ........ Merged
revisions 380210 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 380211 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/logger.c, CHANGES, configs/logger.conf.sample: Add
queue_log_realtime_use_gmt option to logger.conf Add an option
that lets you specify that the timestamps going into the realtime
queue log should be in GMT instead of local time. Review:
https://reviewboard.asterisk.org/r/2287/
2013-01-27 20:33 +0000 [r380194] Michael L. Young <elgueromexicano@gmail.com>
* apps/confbridge/conf_config_parser.c, /: Fix Some Configured
Conference Bridge Sounds Not Being Set The "sound_only_one" sound
was not being set even though it was configured. In looking into
this, I found that the "join" and "leave" prompts were not being
set either. (closes issue ASTERISK-20898) Reported by: Stephan
Tested by: Stephan Patches:
asterisk-20898-custom-sounds-ignored.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2289/ ........ Merged
revisions 380193 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-27 18:40 +0000 [r380165-380178] Joshua Colp <jcolp@digium.com>
* tests/test_sorcery.c: Add a unit test which confirms the apply
handler callback is called when it should be.
* main/sorcery.c: Fix a bug where the apply function was not
getting called.
2013-01-25 23:23 +0000 [r380142] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_multiplexed.c: bridge_multiplexed: Rename
variables so they are not the same as the struct name. * Rename
multiplexed_thread variables to muxed_thread. It is shorter and
my editer tagging works much better. Struct names and variable
names have different purposes and therefore should have different
names. * Renamed the multiplexed_threads container to
muxed_threads for consistency.
2013-01-25 20:46 +0000 [r380121] Jason Parker <jparker@digium.com>
* res/res_sorcery_memory.c, res/res_sorcery_config.c: Make sorcery
modules global, since they are required by other modules that are
global.
2013-01-25 20:00 +0000 [r380108-380109] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_multiplexed.c, main/bridging.c: Misc bridge code
improvements * Made multiplexed_bridge_destroy() check if
anything to destroy and cleared bridge_pvt pointer after
destruction. * Made multiplexed_add_or_remove() handling of the
chans array simpler. * Extracted bridge_channel_poke(). *
Simplified bridge_array_remove() handling of the bridge->array[].
The array does not have a NULL sentinel pointer. * Made
ast_bridge_new() not create a temporary bridge just to see if it
can be done. Only need to check if there is an appropriate bridge
tech available. * Made ast_bridge_new() clean up on allocation
failures. * Made destroy_bridge() free resources in the opposite
order of creation.
* bridges/bridge_simple.c, bridges/bridge_softmix.c,
bridges/bridge_multiplexed.c, main/bridging.c: More trivial
bridge code cleanup. * Breaking long lines * Word wrapping
comment blocks. * Removing redundant initializers. * Debug
message wording.
2013-01-25 14:23 +0000 [r380069-380082] Joshua Colp <jcolp@digium.com>
* res/res_sorcery_config.c: Add a missing '\' to a log message.
* configs/test_sorcery.conf.sample (added),
res/res_sorcery_memory.c (added), configs/sorcery.conf.sample
(added), include/asterisk/sorcery.h (added), tests/test_sorcery.c
(added), main/asterisk.c, main/sorcery.c (added),
res/res_sorcery_config.c (added): Merge the sorcery data access
layer API. Sorcery is a unifying data access layer which provides
a pluggable mechanism to allow object creation, retrieval,
updating, and deletion using different backends (or wizards).
This is a fancy way of saying "one interface to rule them all"
where them is configuration, realtime, and anything else that
comes along. Review: https://reviewboard.asterisk.org/r/2259/
2013-01-25 05:49 +0000 [r380057] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c, configs/skinny.conf.sample: Add force
dial keys to skinny. Adds a dial softkey when the device is in
DAFD. The softkey is greyed (unusable) until a possible dialplan
match is entered. Code includes updating transmit_selectsoftkeys
to allow the use of a button mask. Also add option to use # or *
as a dial now button. Original patch by snuffy cleaned up by
myself. Review: https://reviewboard.asterisk.org/r/2277/
2013-01-24 16:40 +0000 [r380044] David M. Lee <dlee@digium.com>
* /, channels/sip/sdp_crypto.c: Corrected crypto tag in SDP ANSWER
for SRTP. When Asterisk responds with an SDP ANSWER for SRTP, it
had the code to correctly fill in the crypto data, which was
overwritten by a call to sdp_crypto_offer. Corrected the
situation by changing sdp_crypto_offer to not replacing crypto
data if it already exists. (closes issue ASTERISK-20849) Reported
by: José Luis Millán Tested by: Iñaki Baz Castillo Patches:
fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407)
........ Merged revisions 380043 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-24 04:02 +0000 [r380029] Matthew Jordan <mjordan@digium.com>
* /, apps/app_confbridge.c: Correct documentation for
ConfbridgeList AMI action The documentation for ConfbridgeList
states that the Conference field is optional. That's not really
the case: if you fail to provide a Conference number, the command
will kick back an error. (closes issue AST-1090) Reported by:
John Bigelow ........ Merged revisions 380028 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-23 16:50 +0000 [r380004] Kinsey Moore <kmoore@digium.com>
* contrib/scripts/autosupport: Add support for DPMA to autosupport
This adds the ability to get the DPMA version, a listing of the
local firmware directory, and indexes of configured remote
directories. (closes issue AST-1070) Reported By: Malcolm
Davenport Tested By: Kinsey Moore <kmoore@digium.com>
2013-01-23 00:30 +0000 [r379966] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c, /: Attempt to be more helpful when using a bad
ao2 object pointer. Put the external obj pointer in the message
instead of the internal version. ........ Merged revisions 379963
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 379964 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-22 22:19 +0000 [r379950] Jonathan Rose <jrose@digium.com>
* /, res/res_fax_spandsp.c: res_fax_spandsp: fix t38 transmission
bug caused by not returning success This patch fixes the problem,
but the issue includes a test which is still being considered for
the automated test suite. (issue ASTERISK-20919) Reported by:
NITESH BANSAL Patches: patch_ast_fax_spandsp.patch uploaded by
NITESH BANSAL (license 6418) ........ Merged revisions 379949
from http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-22 20:58 +0000 [r379936] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: Remove a large block of commented out code
from chan_iax2. During the conversion to the newer CLI command
structure the old definitions were commented out. I think it's
safe to remove them completely now.
2013-01-22 19:29 +0000 [r379912] Jonathan Rose <jrose@digium.com>
* sounds/Makefile, /, apps/app_meetme.c: app_meetme: Use new
prompts for administrator menu The old prompts for the
administrator menu were inadequate. They didn't mention that the
menu had additional options through the 8 key and pressing the 8
key wouldn't reveal what those options were. This patch fixes all
of that while also organizing code pertaining to each individual
menu type which was previously all stored in one gigantic
function along with many of the basic conference functions.
(closes issue AST-996) Reported by: John Bigelow Review:
http://reviewboard.digium.internal/r/360/ ........ Merged
revisions 379885 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379892 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-22 16:48 +0000 [r379864] Richard Mudgett <rmudgett@digium.com>
* /: Remove stray property.
2013-01-22 15:16 +0000 [r379828-379830] Matthew Jordan <mjordan@digium.com>
* res/res_agi.c, main/file.c, main/app.c, CHANGES,
include/asterisk/frame.h, apps/app_playback.c,
apps/app_controlplayback.c, include/asterisk/file.h,
main/channel.c, funcs/func_frame_trace.c: Add ControlPlayback
manager action This patch adds the capability for asynchronous
manipulation of audio being played back to a channel though a new
AMI action "ControlPlayback". The ControlPlayback action supports
a number of operations, the availability of which depend on the
application being used to send audio to the channel. When the
audio playback was initiated using the ControlPlayback
application or CONTROL STREAM FILE AGI command, the audio can be
paused, stopped, restarted, reversed, or skipped forward. When
initiated by other mechanisms (such as the Playback application),
the audio can be stopped, reversed, or skipped forward. Review:
https://reviewboard.asterisk.org/r/2265/ (closes issue
ASTERISK-20882) Reported by: mjordan
* /, apps/app_meetme.c: Fix station ringback; trunk hangup issues
in SLA This patch fixes two bugs: * If an outbound call is made
from a SLA phone using SLAStation, then there is no ringtone
audible to the phone that originates the call. The indication of
the ringing was not being passed to the SLA station; this patch
fixes that by passing through the progress indications. * If an
SLA station hangs up before the called party answers, then the
channel to the called party continues to ring until a timeout
occurs. If the called party manages to answer, Asterisk attempts
to connect the called party to a non-existant MeetMe room. This
patch corrects the behavior by abandoning the call attempt if it
detects that the SLA station is no longer in use while attempting
to call the called party. Review:
https://reviewboard.asterisk.org/r/2275/ (closes issue
ASTERISK-20462) Reported by: dkerr patches:
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
5558) asterisk-11-bugid20462.patch uploaded by dkerr (license
5558) (closes issue ASTERISK-20440) Reported by: dkerr patches:
asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
5558) ........ Merged revisions 379825 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379826 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-22 00:36 +0000 [r379809] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_bridge.c, apps/app_confbridge.c: confbridge:
Minor fixes playing user counts to the conference. * Generate a
warning message if sound files do not exist when trying to play
the user count to the conference. Use the new helper routine
sound_file_exists() for consistency. * Put the new user into
autoservice when playing user counts to the conference. * Check
the return value of ast_bridge_impart(). ........ Merged
revisions 379808 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-21 20:41 +0000 [r379791] Matthew Jordan <mjordan@digium.com>
* /, contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.gentoo.asterisk,
contrib/init.d/rc.slackware.asterisk,
contrib/init.d/rc.archlinux.asterisk,
contrib/scripts/safe_asterisk, main/asterisk.c,
contrib/init.d/rc.suse.asterisk,
contrib/init.d/rc.mandriva.asterisk,
contrib/init.d/rc.debian.asterisk: Update init.d scripts to
handle stderr; readd splash screen for remote consoles When
r376428 was commited to re-order start up sequences to be more
tolerant of forking with thread primitives, a few items were
changed that caused changes in behavior on some distros. This
includes: * Not displaying the splash screen on a remote console.
* Displaying an error message on stderr when a remote console
cannot connect to a running instance of Asterisk. In the first
case, the splash screen was re-added (thanks to Michael L.
Young). In the second case, the various init.d scripts were
modified to pipe stderr to /dev/null, as the error message is
useful - if you execute a remote console or a remote console
command execution and it fail, it should tell you. Note that the
error message was always present, it just failed to be printed
prior to r376428. Much thanks to the folks who quickly reported
this problem, provided solutions, and promptly tested the various
init.d scripts on a variety of distros. (closes issue
ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L.
Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches:
asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
(license 6283) ........ Merged revisions 379760 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379777 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 379790 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-21 20:35 +0000 [r379753-379789] Richard Mudgett <rmudgett@digium.com>
* bridges/bridge_builtin_features.c, main/bridging.c: Better
protect bridge_channel state from other threads.
* main/bridging.c: Extract common bridging code into bridge_stop()
and bridge_force_out_all().
* bridges/bridge_builtin_features.c,
include/asterisk/bridging_features.h,
include/asterisk/bridging.h, main/bridging.c: Made some bridging
API calls void. Some bridging comments updated.
2013-01-21 18:47 +0000 [r379721] Kinsey Moore <kmoore@digium.com>
* codecs/codec_ilbc.c, /: Prevent segfault for interpolated iLBC
frames When iLBC is being used with a jitter buffer and the jb
has to interpolate frames, it generates frames with a null
pointer and a non-zero datalen. This is now handled properly.
(closes issue ASTERISK-20914) Reported By: John McEleney Patches:
ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
........ Merged revisions 379718 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379719 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-21 18:45 +0000 [r379703-379720] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: Trivial bridge code cleanup.
* include/asterisk/bridging_features.h,
include/asterisk/bridging.h,
include/asterisk/bridging_technology.h,
bridges/bridge_builtin_features.c: Bridge API comment tweaks.
2013-01-21 07:26 +0000 [r379678] Damien Wedhorn <voip@facts.com.au>
* /, channels/chan_skinny.c: Fix device call logging issues in
skinny Skinny device call logging (ie missed, place and received
calls) has issues because the incorrect sequence of callstates
is/can be sent to the device. This patch removes some extra
callstate updates driven by forces external to skinny and ensures
the needed intermediary callstate messages are sent. (closes
issue ASTERISK-20964) Reported by: wedhorn Tested by: snuffy,
myself Patches: ast11-skinny-calllog01.diff uploaded by wedhorn
(license 5019) ........ Merged revisions 379677 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-21 04:50 +0000 [r379644] Andrew Latham <lathama@gmail.com>
* contrib/scripts/install_prereq, /: Add LDAP libraries to install
script Add LDAP dev package to Debian/Ubuntu install list.
Existed in Redhat already. (issue ASTERISK-20886) ........ Merged
revisions 379643 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-21 04:17 +0000 [r379610-379612] Matthew Jordan <mjordan@digium.com>
* /, apps/app_minivm.c: Fix crash in app_minivm when mime encoding
string An incorrect string initializations was left in
ast_str_encode_mime from the patch that converted string
manipulations to use ast_str strings (r191140). The string
initialization causes a crash when ast_str_set is called on the
string later on in the function. (closes issue ASTERISK-18697)
Reported by: Chris Boot patches:
minivm-null-pointer-dereference-fix.patch uploaded by bootc
(license 6309) (issue ASTERISK-20854) Reported by: Chris Warr
Tested by: Chris Warr ........ Merged revisions 379608 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379609 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /: Re-add merge properties
2013-01-20 03:06 +0000 [r379583] Damien Wedhorn <voip@facts.com.au>
* /, channels/chan_skinny.c: Fix issues with skinny sessions Fixes
a couple of issues with the way skinny handles sessions by
ensuring sessions aren't used after being freed. Some other minor
changes. Review: https://reviewboard.asterisk.org/r/2272/
........ Merged revisions 379582 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-19 20:54 +0000 [r379549] Walter Doekes <walter+asterisk@wjd.nu>
* configure.ac, /, configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, main/strcompat.c: Add builtin roundf()
for systems lacking it. (closes issue ASTERISK-16854) Review:
https://reviewboard.asterisk.org/r/2276 Reported-by: Ovidiu Sas
........ Merged revisions 379547 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379548 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-19 00:19 +0000 [r379518] Matthew Jordan <mjordan@digium.com>
* main/asterisk.c, /: Fix astcanary startup problem due to wrong
pid value from before daemon call When Asterisk forks itself into
the background via a call to daemon, it must re-set the pid value
of the new process. Otherwise, astcanary gets the pid value of
the process before the fork, which prevents it from running.
Asterisk eventually starts lowering its priority, as it can no
longer communicate with the proverbial canary in the coal mine.
This patch ensures that the correct process identifier is used by
astcanary. Note that this is getting committed to 10 as a
regression fix. (closes issue ASTERISK-20947) Reported by: Jakob
Hirsch Tested by: mjordan patches:
asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
(license 6113) ........ Merged revisions 379509 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379510 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 379513 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-18 22:42 +0000 [r379495] David M. Lee <dlee@digium.com>
* configure, main/Makefile, configure.ac, Makefile: Up the minimum
OS X version to 10.6. * This allows us to remove some
special-case build logic. * 10.5 is down to less that 8% of the
OS X market share. 10.4 is down to under 2%. * Apple is no longer
releasing security updates for 10.5 and earlier.
2013-01-18 21:52 +0000 [r379479] Kinsey Moore <kmoore@digium.com>
* /, apps/app_confbridge.c: Fix regression in Confbridge user count
When the restructuring work got committed to Confbridge in
r375470 to fix many open issues, it caused a regression in the
reported count of users when conference information was requested
via CLI or manager. This corrects the user count and user
information displayed when listing conference information from
the CLI and manager. (closes issue ASTERISK-20938) Reported By:
Timo Teras Patches: confbridge-list.patch uploaded by Timo Teras
(license 5409) ........ Merged revisions 379478 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-18 21:35 +0000 [r379477] David M. Lee <dlee@digium.com>
* /, configure, main/Makefile, configure.ac, UPGRADE-11.txt,
UPGRADE.txt, makeopts.in, Makefile: Specify the -rpath linker
flag when prefix != /usr. This allows Asterisk to start without
having to specify the LD_LIBRARY_PATH. This can be disabled by
passing --disable-rpath to configure. (closes issue
ASTERISK-20407) Reported by: David M. Lee Review:
https://reviewboard.asterisk.org/r/2132/ ........ Merged
revisions 379475 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-18 18:25 +0000 [r379461] Jonathan Rose <jrose@digium.com>
* /, apps/app_voicemail.c: app_voicemail: Improve msg_id handling
app_voicemail will no longer issue error messages when it
retrieves an msg_id with a NULL value from realtime and will
instead simply populate the msg_id field with a newly generated
msg_id. In addition, this patch changes the way msg_ids are
generated to eliminate certain causes of duplicate IDs appearing
within a single system. In addition, when messages are copied,
they will now receive a new msg_id. (closes issue ASTERISK-20717)
Reported by: Alec Davis Review:
https://reviewboard.asterisk.org/r/2220/ ........ Merged
revisions 379460 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-18 15:42 +0000 [r379432] Mark Michelson <mmichelson@digium.com>
* main/threadpool.c (added), main/taskprocessor.c,
include/asterisk/threadpool.h (added), /,
include/asterisk/taskprocessor.h, tests/test_threadpool.c
(added), tests/test_taskprocessor.c (added): Add threadpool
support to Asterisk. This commit consists of two parts. Part one
changes the taskprocessor API to be less self-contained. Instead,
the taskprocessor is now more of a task queue that informs a
listener of changes to the queue. The listener then has the
responsibility of executing the tasks as it pleases. There is a
default listener implementation that functions the same way as
"classic" taskprocessors, in that it creates a single thread for
tasks to execute in. Old users of taskprocessors have not been
altered and still function the same way. Part two introduces the
threadpool API. A threadpool is a special type of taskprocessor
listener that has multiple threads associated with it. The
threadpool also has an optional listener that can adjust the
threadpool as conditions change. In addition the threadpool has a
set of options that can allow for the threadpool to grow and
shrink on its own as tasks are added and executed. Both set of
changes contain accompanying unit tests. (closes issue
ASTERISK-20691) Reported By: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2242
2013-01-18 05:31 +0000 [r379394] David M. Lee <dlee@digium.com>
* channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
channels/sip/reqresp_parser.c: Fix Record-Route parsing for large
headers. Record-Route parsing copied the header into a char[256]
array, which can be a problem if the header is longer than that.
This patch parses the header in place, without the copy, avoiding
the issue. In addition to the original patch, I added a unit test
for the new get_in_brackets_const function. (closes issue
ASTERISK-20837) Reported by: Corey Farrell Patches:
chan_sip-build_route-optimized-rev1.patch uploaded by Corey
Farrell (license 5909) (with minor changes by dlee) ........
Merged revisions 379392 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379393 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-17 02:32 +0000 [r379344] Matthew Jordan <mjordan@digium.com>
* addons/chan_mobile.c, /: Fix issue where chan_mobile fails to
bind to first available port Per the bluez API, in order to bind
to the first available port, the rc_channel field of the socket
addressing structure used to bind the socket should be set to 0.
Previously, Asterisk had set the rc_channel field set to 1,
causing it to connect to whatever happens to be on port 1. We
could probably not explicitly set rc_channel to 0 since we memset
the struct earlier, but explicitly setting it will hopefully
prevent someone from coming in and setting it to some explicit
port in the future. (closes issue ASTERISK-16357) Reported by:
challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin,
eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by
Nikolay Ilduganov (license 6253) ........ Merged revisions 379342
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 379343 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-16 22:51 +0000 [r379312] Mark Michelson <mmichelson@digium.com>
* /, main/manager.c: Further fix misinformation in the description
of manager MailboxStatus command. The description still claimed
that it returned the number of messages rather than whether there
were messages waiting. ........ Merged revisions 379310 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379311 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-16 21:13 +0000 [r379278] Jason Parker <jparker@digium.com>
* contrib/scripts/install_prereq, /: Reduce number of packages
install_prereq installs on Debian systems. 'search' will look for
any package containing the name provided, so we need to force a
more exact search. ........ Merged revisions 379276 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379277 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-16 18:09 +0000 [r379231-379233] Richard Mudgett <rmudgett@digium.com>
* /, main/logger.c: Reduce call-id logging resource usage. Since
there is no need for the call-id logging ao2 object to have a
lock, don't create it with one. ........ Merged revisions 379232
from http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_misdn.c, /: chan_misdn: Fix compile error. (issue
ASTERISK-15456) ........ Merged revisions 379226 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379230 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-16 17:46 +0000 [r379144-379229] Matthew Jordan <mjordan@digium.com>
* /, res/res_xmpp.c, res/res_jabber.c, doc/appdocsxml.dtd: Let
documentation reference links specify which module they're
linking to Again, since res_jabber/res_xmpp have duplicate APIs,
their documentation ref links have to specify which reference
they're referring to. The various documentation parsers can
interpret the module attribute however they want in order to
construct the appropriate links. ........ Merged revisions 379228
from http://svn.asterisk.org/svn/asterisk/branches/11
* /, res/res_xmpp.c, res/res_jabber.c, doc/appdocsxml.dtd: Multiple
revisions 379209-379210 ........ r379209 | mjordan | 2013-01-16
09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines Add module tags to
documentation for res_jabber/res_xmpp Since res_jabber/res_xmpp
provide the same APIs (app/func/manager/etc.), the XML
documentation for each needs to call out which module is
providing the documentation. The module attribute has been added
to the various XML fragments for this purpose. ........ r379210 |
mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
Update the dtd to actually *support* the module attribute in all
elements Mea culpa. ........ Merged revisions 379209-379210 from
http://svn.asterisk.org/svn/asterisk/branches/11
* addons/chan_mobile.c, /: Fix parsing SMSSRC for SMS messages The
parser for SMS messages would incorrectly parse out the from
number. The parsing would incorrectly start scanning for the from
number at the same index as the first double quote ("); this
would inadvertently cause it to treat the first double quote as
the terminating double quote for the from number as well. The
SMSSRC should now populate correctly. (closes issue
ASTERISK-16822) Reported by: menschentier Tested by: Jonas Falck
patches: fixSMSSRC.patch uploaded by jonax (license 6320) (closes
issue ASTERISK-19153) Reported by: Panos Gkikakis patches:
sms-sender-fix.diff uploaded by roeften (license 5884) ........
Merged revisions 379178 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379179 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_misdn.c, /: Set the INVALID_EXTEN channel variable
when chan_misdn forces the 'i' extension The chan_misdn channel
driver will send a channel with an invalid destination to the 'i'
extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it
bounces the channel to this extension. Dialplan writers
everywhere moaned at yet another inconsistency. This is yet
another example of why duplicating logic in multiple places
results in bugs that stick around in Jira for just under three
years. Yes: ASTERISK-15456 was created on January 18th, 2010.
Patch committed on January 15th, 2013. Ouch. (closes issue
ASTERISK-15456) Reported by: Thomas Omerzu patches:
chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license
5927) ........ Merged revisions 379145 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379146 from
http://svn.asterisk.org/svn/asterisk/branches/11
* CHANGES, addons/chan_mobile.c: Add busy detection to chan_mobile
From the patch author: "First this patch adds general support for
busy detection. It also adds support for the ECAM command at Sony
Ericsson phones and also signals busy when only early media was
received but the call got not answered." Review:
https://reviewboard.asterisk.org/r/323 (closes issue
ASTERISK-14527) Reported by: Artem Makhutov Tested by: Artem
Makhutov patches: busy-full5.patch uploaded by artem (license
5757)
2013-01-15 22:23 +0000 [r379128] Richard Mudgett <rmudgett@digium.com>
* main/bridging.c: Fix ast_bridge_features_register() not
registering builtin features. I broke. Ooops.
2013-01-14 21:47 +0000 [r379021-379070] David M. Lee <dlee@digium.com>
* include/asterisk/test.h: Fixed doc comment for ast_test_validate
* UPGRADE.txt, include/asterisk/manager.h, main/channel.c: Gently
reduce masquerade insanity Masquerades are an insane
implementation detail within Asterisk. It generates a number of
useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly
thorough review of masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ. While ultimately it
makes the most sense to abandon masquerades altogether, it will
take some time to completely irradicate. Even then, there may
always be code that's not worth rewriting to get rid of the
masquerade. This patch does two things to make masquerades
slightly less insane: * When swapping the names of the original
and clone channel, only emit a single rename event of original ->
original<ZOMBIE>. The original code issued three rename events to
accomplish the same end. * In addition to swapping the names of
the channels, also swap their uniqueid's. This allows the
'Uniqueid' field to be used as a stable identifier for a channel
from and external interface, such as AMI. Review:
https://reviewboard.asterisk.org/r/2266/
* /, channels/chan_sip.c: Fix XML encoding of 'identity display' in
NOTIFY messages, continued. When r378933 was merged into 1.8, it
should have also escaped remote_display, since it will have the
same XML encoding problem when the caller/callee roles are
reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter
........ Merged revisions 379001 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 379020 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-13 22:07 +0000 [r378985] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c, /: Reset RTP timestamp; sequence number
on SSRC change In r370252 for ASTERISK-18404, Asterisk's handling
of RTP was modified to better account for out of order RTP
packets. This was accomplished by using the RTP timestamp and
sequence number to check for out of order packets. However, when
a SSRC change occurs, the timestamp and sequence number will no
longer have any relation to the previously received packets. The
variables tracking the timestamp and sequence number therefore
have to be reset. (closes issue ASTERISK-20906) Reported by:
Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco
Brolman (license #6442) ........ Merged revisions 378967 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378984 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-12 06:43 +0000 [r378935] David M. Lee <dlee@digium.com>
* include/asterisk/utils.h, /, channels/chan_sip.c,
tests/test_xml_escape.c (added), main/utils.c: Fix XML encoding
of 'identity display' in NOTIFY messages. XML encoding in
chan_sip is accomplished by naively building the XML directly
from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML. This patch adds
an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the
local_display attribute in XML formatted NOTIFY messages. Several
things to note: * The Right Thing(TM) to do would probably be to
replace the ast_build_string stuff with building an ast_xml_doc.
That's a much bigger change, and out of scope for the original
ticket, so I refrained myself. * It is with great sadness that I
wrote my own ast_xml_escape function. There's one in libxml2, but
it's knee-deep in libxml2-ness, and not easily used to one-off
escape a string. * I only escaped the string we know is causing
problems (local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902) Reported by: Guenther Kelleter Tested by:
Guenther Kelleter Review:
http://reviewboard.digium.internal/r/365/ ........ Merged
revision 378919 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........ Merged revisions 378933 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378934 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-11 23:05 +0000 [r378918] Joshua Colp <jcolp@digium.com>
* res/res_xmpp.c, /: Retain XMPP filters across reconnections so
external modules continue to function as expected. Previously if
an XMPP client reconnected any filters added by an external
module were lost. This issue exhibited itself with chan_motif not
receiving and reacting to Jingle signaling. (closes issue
ASTERISK-20916) Reported by: kuj ........ Merged revisions 378917
from http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-11 22:31 +0000 [r378915] David M. Lee <dlee@digium.com>
* include/asterisk/json.h (added), makeopts.in, tests/test_json.c
(added), contrib/scripts/install_prereq, res/res_json.c (added),
include/asterisk/test.h, build_tools/menuselect-deps.in,
configure, include/asterisk/autoconfig.h.in, main/Makefile,
res/res_json.exports.in (added), configure.ac: Add JSON API for
Asterisk. This provides a JSON API by pulling in and wrapping the
Jansson JSON library[1]. The Asterisk API basically mirrors the
Jansson functionality, with a few minor tweaks. * Some names have
been asteriskified to protect the innocent. * Jansson provides
both reference-stealing and reference-borrowing versions of
several API's. The Asterisk API is exclusively reference-stealing
for operations that put elements into arrays and objects. * No
support for doubles, since we usually don't need that. * Coming
along for the ride is the ast_test_validate macro, which made the
unit tests much easier to write. [1]:
http://www.digip.org/jansson/ (issue ASTERISK-20887) (closes
issue ASTERISK-20888) Review:
https://reviewboard.asterisk.org/r/2264/
2013-01-10 02:40 +0000 [r378789-378889] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: * Simplify native bridge code in
ast_channel_bridge(). * Fix an unbalanced
manager_bridge_event(unlink) call if AST_SOFTHANGUP_UNBRIDGE is
set in ast_channel_bridge(). * Make ast_channel_bridge() use
common cleanup code when leaving the bridge.
* main/channel.c: * Removed some noop code and restructured an
else-if ladder in ast_generic_bridge(). * Trivial changes in
ast_channel_bridge().
* main/channel.c: * Simple optimization of bridge_playfile(). *
Squeezed some redundancy out of update_bridge_vars(). * Wrapped
long line in __ast_change_name_nolink().
* bridges/bridge_softmix.c, bridges/bridge_multiplexed.c: Trivial
misc bridge code changes. * softmix_bridge_thread() was
redundantly initializing an 8K buffer. * Promoted a debug message
to a warning in multiplexed_add_or_remove().
* main/logger.c: Fix logger.c function definition.
* bridges/bridge_multiplexed.c, main/bridging.c,
include/asterisk/bridging_features.h, bridges/bridge_simple.c:
Trivial misc bridge code changes.
* include/asterisk/test.h, main/test.c: Tweaked
__ast_test_suite_assert_notify() and
__ast_test_suite_event_notify() to be void functions.
* include/asterisk/test.h, main/test.c: * Whitespace changes. *
Made ast_test_init() match its prototype.
* main/udptl.c, main/rtp_engine.c: * Found some more places to use
ast_channel_lock_both(). * Minor optimization in
ast_rtp_instance_early_bridge().
2013-01-09 20:30 +0000 [r378735-378783] David M. Lee <dlee@digium.com>
* main/rtp_engine.c, /: Fix end condition in
ast_rtp_lookup_mime_multiple2. The erroneous end condition would
never include the AST_RTP_CISCO_DTMF flag in the debug output.
(closes issue ASTERISK-20772) Reported by: Xavier Hienne ........
Merged revisions 378776 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378780 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, include/asterisk/strings.h: Move declaration of
ast_regex_string_to_regex_pattern futher down strings.h. The
prior location is before the declaration of struct ast_str, which
causes compiler warnings. (closes issue ASTERISK-20852) Reported
by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller
(license 6302) ........ Merged revisions 378747 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, include/asterisk/causes.h: Replace errant tabs with spaces in
causes.h. (closes issue ASTERISK-20826) Reported by: snuffy
Patches: notabs.dif uploaded by snuffy (license 5024) ........
Merged revisions 378733 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378734 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-09 00:05 +0000 [r378688-378691] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, /: app_queue: Fix incorrect assertion. (issue
ASTERISK-16115) ........ Merged revisions 378689 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378690 from
http://svn.asterisk.org/svn/asterisk/branches/11
* CHANGES, apps/app_queue.c, /, configs/queues.conf.sample,
UPGRADE.txt: app_queue: Fix multiple calls to a queue member that
is in only one queue. When ringinuse=no queue members can receive
more than one call if these calls happen at nearly the same time.
* Fix so a queue member does not receive more than one call from
a queue. NOTE: This fix does not prevent multiple calls to a
member if the member is in more than one queue. * Did some
refactoring to eliminate some code redundancy. (issue
ASTERISK-16115) Reported by: nik600 Patches:
jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
uploaded by rmudgett Modified * Revert the -r341580 and -r341599
changes adding the queues.conf check_state_unknown option as it
was added in an attempt to fix this problem. The fix did not need
to be optional. The fix should not have tried to explicitly set
the device state. Setting the device state by something other
than the device introduces a race condition. I also could not see
how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to
app_queue. ........ Merged revisions 378663 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378683 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378687 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-06 21:37 +0000 [r378623-378634] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Skinny blob cleanup Cleanup of red blobs
in chan_skinny and possible other small formatting issues.
Review: https://reviewboard.asterisk.org/r/2262/
* channels/chan_skinny.c: Add group and namedgroup pickup to skinny
Above says it all. Code by snuff, cleaned up by me. Review:
https://reviewboard.asterisk.org/r/2246/
* /, channels/chan_skinny.c: Rewrite skinny dialing to remove
threaded simpleswitch This rewrite changes skinny dialing from
the threaded simpleswitch to a scheduled timeout approach. There
were some underlying issues with the threaded simple switch with
occasional corruption and possible segfaults. Review:
https://reviewboard.asterisk.org/r/2240/ ........ Merged
revisions 378622 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-04 23:14 +0000 [r378593] Jonathan Rose <jrose@digium.com>
* res/res_srtp.c, /: res_srtp: Prevent a crash from occurring due
to srtp_create failures in srtp_create Under some circumstances,
libsrtp's srtp_create function deallocates memory that it wasn't
initially responsible for allocating. Because we weren't
initially aware of this behavior, this memory was still used in
spite of being unallocated during the course of the
srtp_unprotect function. A while back I made a patch which would
set this value to NULL, but that exposed a possible condition
where we would then try to check a member of the struct which
would cause a segfault. In order to address these problems,
ast_srtp_unprotect will now set an error value when it ends
without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant
channel instead of trying to keep using the invalid session
address. (closes issue ASTERISK-20499) Reported by: Tootai
Review:
https://reviewboard.asterisk.org/r/2228/diff/#index_header
........ Merged revisions 378591 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378592 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-04 22:19 +0000 [r378585] Kinsey Moore <kmoore@digium.com>
* res/pjproject/aconfigure, res/pjproject/aconfigure.ac, /,
res/pjproject/build/common.mak: Fix pjproject compilation in
certain circumstances On a fresh checkout of Asterisk 11, running
make before ./configure could cause the pjproject subdirectory to
get in an odd state that would prevent compilation. This patch by
Tilghman prevents that from occurring. (closes issue
ASTERISK-20681) Reported by: Dinesh Ramjuttun Tested by: danilo
borges, Steve Lang patches: 20121208__ccar_solved.diff.txt
uploaded by Tilghman Lesher (license 5003) ........ Merged
revisions 378582 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-04 21:20 +0000 [r378565] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix SIP Notify Messages To Have The
Proper IP Address In The FROM Field On a multihomed server when
sending a NOTIFY message, we were not figuring out which network
should be used to contact the peer. This patch fixes the problem
by calling ast_sip_ouraddrfor() and then build_via() so that our
NOTIFY message contains the correct IP address. Also, a debug
message is being added to help follow the call-id changes that
occur. This was helpful for confirming that the IP address was
set properly since the call-id contains the IP address. It also
will be helpful for troubleshooting purposes when following a
call in the debug logs. (closes issue ASTERISK-20805) Reported
by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches:
asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2255/
........ Merged revisions 378554 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378559 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-04 21:18 +0000 [r378557] Joshua Colp <jcolp@digium.com>
* /, res/res_rtp_asterisk.c: Don't pass STUN packets through the
SRTP unprotect function. (closes issue AST-1036) Reported by:
jbigelow ........ Merged revisions 378553 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378555 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-04 16:44 +0000 [r378543] Andrew Latham <lathama@gmail.com>
* res/res_config_ldap.c: Doxygen Cleanups Baseline clean up of
formating to make room for extended documentation (issue
ASTERISK-20259)
2013-01-03 22:14 +0000 [r378516] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_queue.c, /: Fix Queue Log Reporting Every Call
COMPLETECALLER With "h" Extension Present When the "h" extension
is present within the context of the queue, all calls are being
reported COMPLETECALLER even when the agent is hanging up the
call. This patch checks to see if the agent hung-up or not
instead of only relying on checking if the queue (caller) channel
hung-up or not. It would appear that having the h extension in
the mix, the pbx goes to the h extension, "hanging-up" the queue
channel and triggering the reporting of COMPLETECALLER. (closes
issue ASTERISK-20743) Reported by: call Tested by: call, Michael
L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2256/ ........ Merged
revisions 378514 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378515 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-03 19:42 +0000 [r378488] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_agent.c: chan_agent: Fix wrapup time wait
response. * Made agent_cont_sleep() and agent_ack_sleep() stop
waiting if the wrapup time expires. agent_cont_sleep() had tried
but returned the wrong value to stop waiting. * Made
agent_ack_sleep() take a struct agent_pvt pointer instead of a
void pointer for better type safety. ........ Merged revisions
378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 378487 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-03 18:51 +0000 [r378460] Kinsey Moore <kmoore@digium.com>
* main/channel.c, /: Add missing test event This test event was
missing from channel.c causing the dial_LS_options test to fail
intermittently because of a race condition where most code paths
emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now. ........ Merged revisions 378455
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 378459 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-03 18:47 +0000 [r378429-378458] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_agent.c: chan_agent: Misc code cleanup. * Fix
off-nominal path resource cleanup in agent_request(). * Create
agent_pvt_destroy() to eliminate inlined versions in many places.
* Pull invariant code out of loop in add_agent(). * Remove
redundant module user references in login_exec(). * Remove unused
struct agent_pvt logincallerid[] member. ........ Merged
revisions 378456 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378457 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_agent.c: chan_agent: Fix agent_indicate()
locking. Avoid deadlock potential with local channels and
simplify the locking. ........ Merged revisions 378427 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378428 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-03 16:04 +0000 [r378414] Tilghman Lesher <tilghman@meg.abyt.es>
* apps/app_directory.c, contrib/realtime/mysql/voicemail.sql,
configs/voicemail.conf.sample: Add aliases to the Directory. This
is an interesting feature that allows additional strings to be
used to search the Directory, primarily intended to be used with
nicknames, but could be used with affiliations and the like.
Because the name field is used in more than one place (such as
email notifications), it is important that these additional
strings not be placed in the name field, but be specified
separately. Review: https://reviewboard.asterisk.org/r/2244/
2013-01-03 15:40 +0000 [r378412] Joshua Colp <jcolp@digium.com>
* /, res/res_xmpp.c: Prevent exhaustion of system resources through
exploitation of event cache This patch changes res_xmpp to no
longer cache events under certain circumstances. (issue
ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua
Colp Tested by: kmoore ........ Merged revisions 378411 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-03 15:37 +0000 [r378377-378410] Matthew Jordan <mjordan@digium.com>
* /, res/res_xmpp.c: Prevent crashes in res_xmpp when receiving
large messages Similar to r378287, res_xmpp was marshaling data
read from an external source onto the stack. For a sufficiently
large message, this could cause a stack overflow. This patch
modifies res_xmpp in a similar fashion to res_jabber by removing
the stack allocation, as it was unnecessary. (issue
ASTERISK-20658) Reported by: wdoekes ........ Merged revisions
378409 from http://svn.asterisk.org/svn/asterisk/branches/11
* addons/app_mysql.c: Clean up app_mysql's application entry points
to properly parse arguments When parsing arguments, application
entry points should not attempt to directly modify the parameters
to the function. This patch properly duplicates the passed in
parameters before attempting to parse them. (issue
ASTERISK-20658) Reported by: wdoekes patches:
issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license
5674)
* main/config.c, funcs/func_realtime.c, /: Prevent crashes from
occurring when reading from data sources with large values When
reading configuration data from an Asterisk .conf file or when
pulling data from an Asterisk RealTime backend, Asterisk was
copying the data on the stack for manipulation. Unfortunately, it
is possible to read configuration data or realtime data from some
data source that provides a large blob of characters. This could
potentially cause a crash via a stack overflow. This patch
prevents large sets of data from being read from an ARA backend
or from an Asterisk conf file. (issue ASTERISK-20658) Reported
by: wdoekes Tested by: wdoekes, mmichelson patches: *
issueA20658_dont_process_overlong_config_lines.patch uploaded by
wdoekes (license 5674) * issueA20658_func_realtime_limit.patch
uploaded by wdoekes (license 5674) ........ Merged revisions
378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 378376 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-02 21:23 +0000 [r378374] Richard Mudgett <rmudgett@digium.com>
* main/features.c, include/asterisk/channel.h, main/manager.c, /:
Fix AMI redirect action with two channels failing to redirect
both channels. The AMI redirect action can fail to redirect two
channels that are bridged together. There is a race between the
AMI thread redirecting the two channels and the bridge thread
noticing that a channel is hungup from the redirects. * Made the
bridge wait for both channels to be redirected before exiting. *
Made the AMI redirect check that all required headers are present
before proceeding with the redirection. * Made the AMI redirect
require that any supplied ExtraChannel exist before proceeding.
Previously the code fell back to a single channel redirect
operation. (closes issue ASTERISK-18975) Reported by: Ben Klang
(closes issue ASTERISK-19948) Reported by: Brent Dalgleish
Patches: jira_asterisk_19948_v11.patch (license #5621) patch
uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak
Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/
........ Merged revisions 378356 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378358 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-02 18:11 +0000 [r378288-378322] Matthew Jordan <mjordan@digium.com>
* main/event.c, apps/app_confbridge.c,
apps/confbridge/conf_state_empty.c, funcs/func_devstate.c,
res/res_calendar.c, include/asterisk/devicestate.h,
channels/chan_local.c, /, main/ccss.c, channels/chan_sip.c,
apps/app_meetme.c, main/channel_internal_api.c,
channels/chan_agent.c, main/devicestate.c,
include/asterisk/channel.h, res/res_jabber.c, apps/app_queue.c,
channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
channels/chan_skinny.c, include/asterisk/event_defs.h,
main/features.c: Prevent exhaustion of system resources through
exploitation of event cache Asterisk maintains an internal cache
for devices in the event subsystem. The device state cache holds
the state of each device known to Asterisk, such that consumers
of device state information can query for the last known state
for a particular device, even if it is not part of an active
call. The concept of a device in Asterisk can include entities
that do not have a physical representation. One way that this
occurred was when anonymous calls are allowed in Asterisk. A
device was automatically created and stored in the cache for each
anonymous call that occurred; this was possible in the SIP and
IAX2 channel drivers and through channel drivers that utilized
the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and
Motif). These devices are never removed from the system, allowing
anonymous calls to potentially exhaust a system's resources. This
patch changes the event cache subsystem and device state
management to no longer cache devices that are not associated
with a physical entity. (issue ASTERISK-20175) Reported by:
Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore
patches: event-cachability-3.diff uploaded by jcolp (license
5000) ........ Merged revisions 378303 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378320 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378321 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/http.c, res/res_jabber.c, channels/sip/include/sip.h, /,
channels/chan_sip.c: Resolve crashes due to large stack
allocations when using TCP Asterisk had several places where
messages received over various network transports may be copied
in a single stack allocation. In the case of TCP, since multiple
packets in a stream may be concatenated together, this can lead
to large allocations that overflow the stack. This patch modifies
those portions of Asterisk using TCP to either favor heap
allocations or use an upper bound to ensure that the stack will
not overflow: * For SIP, the allocation now has an upper limit *
For HTTP, the allocation is now a heap allocation instead of a
stack allocation * For XMPP (in res_jabber), the allocation has
been eliminated since it was unnecesary. Note that the HTTP
portion of this issue was independently found by Brandon Edwards
of Exodus Intelligence. (issue ASTERISK-20658) Reported by:
wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
uploaded by wdoekes (license 5674) ........ Merged revisions
378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 378286 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378287 from
http://svn.asterisk.org/svn/asterisk/branches/11
2013-01-01 19:02 +0000 [r378259] Andrew Latham <lathama@gmail.com>
* contrib/scripts/install_prereq: Add UUID packages now required to
configure In ASTERISK-20726 UUID was added to Asterisk. This
commit is to add the dependancies to the install script
2013-01-01 17:10 +0000 [r378248-378249] Sean Bright <sean@malleable.com>
* main/translate.c: Revert 378248. I changed the logic of this
function unitentionally, pointed out by file.
* main/translate.c: Bail out early when building an ast_trans_pvt
and the translator doesn't supply a 'newpvt'
2012-12-31 14:46 +0000 [r378220] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
without crypto info This ensures that Asterisk rejects encrypted
media streams (RTP/SAVP audio and video) that are missing
cryptographic keys and ensures that the incoming SDP is
consistent with RFC4568 as far as having a crypto attribute
present for any SAVP streams. Review:
https://reviewboard.asterisk.org/r/2204/ ........ Merged
revisions 378217 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378218 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378219 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-20 21:51 +0000 [r378166] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Give the causes[] a struct name. ........
Merged revisions 378164 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378165 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-18 17:48 +0000 [r378122] Kinsey Moore <kmoore@digium.com>
* main/channel.c, /: Add test events for time limit-related hangups
This patch adds hangup-related test events in order to support
testing of time-limited bridges. This aids in testing the S() and
L() bridge options. (issue SWP-4713) ........ Merged revisions
378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 378120 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378121 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-17 23:10 +0000 [r378081-378095] Richard Mudgett <rmudgett@digium.com>
* main/loader.c, /: Fix potential double free when unloading a
module. ........ Merged revisions 378092 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378093 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378094 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_local.c, /: Make chan_local module references tied
to local_pvt lifetime. The chan_local module references were
manually tied to the existence of the ;1 and ;2 channel links. *
Made chan_local module references tied to the existence of the
local_pvt structure as well as automatically take care of the
module references. * Tweaked the wording of the local_fixup()
failure warning message to make sense. Review:
https://reviewboard.asterisk.org/r/2181/ ........ Merged
revisions 378088 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378089 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378090 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_local.c: chan_local: Parse dial string
consistently. * Fix local_alloc() unexpected limitation of exten
and context length from a combined length of 80 characters to a
normal 80 characters each. * Made local_alloc() and
local_devicestate() parse the same way.
2012-12-17 20:59 +0000 [r378074] Jason Parker <jparker@digium.com>
* /, main/Makefile: Make libasteriskssl.so symlink use a relative
path. This was causing issues when using DESTDIR, since the path
to which the link pointed is not likely to exist (and not useful
to exist) on the target system. (issue ASTNOW-284) ........
Merged revisions 378073 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-17 20:34 +0000 [r378072] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: chan_local: Misc lock and ref tweaks. *
awesome_locking() does not need to thrash the pvt lock as much. *
local_setoption() does not need to check for NULL pvt on cleanup
since it will never be NULL. * Made ref the pvt before locking
for consistency.
2012-12-14 22:45 +0000 [r378064] Richard Mudgett <rmudgett@digium.com>
* channels/chan_agent.c: chan_agent: Remove some duplicated code.
No need to check for an agent twice. Santa does that.
2012-12-14 22:34 +0000 [r378063] Jonathan Rose <jrose@digium.com>
* CHANGES, main/features.c, UPGRADE.txt: Features: BRIDGE_FEATURES
variable automixmonitor support and use proper party
BRIDGE_FEATURES did not previously support the automixmonitor
feature. Now it does. In addition, the BRIDGE_FEATURES variable
would not apply features to the proper party based on whether the
feature option letter was in caps or in lowercase (both ways
would apply it to the caller). Now uppercase applies to the
caller while lowercase applies to the callee (like with the dial
option)
2012-12-14 21:35 +0000 [r378029-378039] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, /: app_queue: Revert bad ringinuse=no patch.
With the option ringinuse=no set, the patch committed for
ASTERISK-16115 causes non-SIP queue members to never be called
because the device state is checked after a channel is created to
determine if the member is busy. These queue members always get
the "Member %s is busy, cannot dial" message. Most channel
drivers other than chan_sip use the default device state
handling. The default device-state state is considered in use or
unknown if the channel exists or not respectively. (closes issue
ASTERISK-20801) Reported by: rmudgett Patches:
jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
patch uploaded by rmudgett ........ Merged revisions 378036 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 378037 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 378038 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/app_queue.c: app_queue: Make update_status() not return
anything.
2012-12-14 01:55 +0000 [r378006-378011] Damien Wedhorn <voip@facts.com.au>
* /, channels/chan_skinny.c: Fix skinny to recognise vmexten in
general section of conf Fixup the vmexten so if globally set in
general section will be honored by chan_skinny. Also get rid of
the 'global_' part of variable name to match regexten. (closes
issue ASTERISK-20790) Reported by: snuffy Tested by: snuffy,
myself Patches: skinny-vm.diff uploaded by snuffy (license 5024)
........ Merged revisions 378010 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_skinny.c: Add g722 codec support to skinny (closes
issue ASTERISK-20788) Reported by: snuffy Tested by: snuffy,
myself Patches: skinny-g722.diff uploaded by snuffy (license
5024)
2012-12-13 21:28 +0000 [r378002] Richard Mudgett <rmudgett@digium.com>
* apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c,
apps/confbridge/conf_state.c, /,
apps/confbridge/include/confbridge.h,
include/asterisk/bridging.h: confbridge: Fix MOH on simultaneous
user entry to a new conference. When two users entered a new
conference simultaneously, one of the callers hears MOH. This
happened if two unmarked users entered simultaneously and also if
a waitmarked and a marked user entered simultaneously. * Created
a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be
locked when actually starting/stopping MOH because there is a
small window between the conference join unsuspend MOH and
actually joining the mixing bridge. * Created the concept of
suspended MOH so it can be interrupted while conference join
announcements to the user and DTMF features can operate. *
Suspend any MOH until the user is about to actually join the
mixing bridge of the conference. This way any pre-join file
playback does not need to worry about MOH. * Made post-join
actions only play deferred entry announcement files. Changing the
user/conference state during that time is not protected or
controlled by the state machine. (closes issue ASTERISK-20606)
Reported by: Eugenia Belova Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/2232/ ........ Merged
revisions 377992 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377993 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-13 21:25 +0000 [r378001] Damien Wedhorn <voip@facts.com.au>
* /, channels/chan_skinny.c: Minor fixes for chan_skinny
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and
correct len of 2 strcmp in skinny_setdebug(). (see opticron's
review on https://reviewboard.asterisk.org/r/2240/) ........
Merged revisions 377991 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-13 21:20 +0000 [r378000] Sean Bright <sean@malleable.com>
* res/res_calendar_exchange.c: Make generate_exchange_uuid() always
return the passed ast_str pointer. I changed this code earlier to
return NULL if it wasn't able to generate a UUID, whereas the
earlier code would always return the ast_str that was passed in.
Switch back to returning the ast_str, only set it to the empty
string instead if UUID generation fails. We still do a validity
check later which will catch this and blow up if necessary.
2012-12-13 21:15 +0000 [r377994] David M. Lee <dlee@digium.com>
* /: Fixed svn merge property breakage from r377986
2012-12-13 18:28 +0000 [r377986] Damien Wedhorn <voip@facts.com.au>
* /, channels/chan_skinny.c: Fix skinny debug tab completion Review
the syntax of the 'skinny debug' command to show more than just
'show' for options to 'skinny debug' command. (closes issue
ASTERISK-20789) Reported by: snuffy Tested by: snuffy, myself
Patches: skinny-debug.diff uploaded by snuffy (license 5024)
........ Merged revisions 377985 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-13 16:43 +0000 [r377981] David M. Lee <dlee@digium.com>
* configure.ac, configure, include/asterisk/autoconfig.h.in: Bail
configure if it can't find libuuid.
2012-12-13 16:18 +0000 [r377977] Russell Bryant <russell@russellbryant.com>
* configure.ac, main/utils.c, configure,
include/asterisk/autoconfig.h.in: Remove compile time check
HAVE_DEV_URANDOM. The code was doing a runtime check, anyway. The
compile time check isn't always valid (cross-compiling,
packages). Review: https://reviewboard.asterisk.org/r/2245/
2012-12-13 15:40 +0000 [r377975] Mark Michelson <mmichelson@digium.com>
* main/taskprocessor.c: Re-add taskprocessor cleanup code that was
removed by the UUID merge.
2012-12-13 15:37 +0000 [r377974] Sean Bright <sean@malleable.com>
* res/res_calendar_exchange.c: Use the UUID API to generate and
validate UUIDs for res_calendar_exchange. Currently the
res_calendar_exchange module uses its own method of generating
UUIDs using ast_random(). Now that we have a UUID API we should
use that instead.
2012-12-13 15:37 +0000 [r377973] Mark Michelson <mmichelson@digium.com>
* res/res_clialiases.c: The UUID commit removed changes made in
res_clialiases.c This puts back in the changes that are designed
to work around a memory leak fix in the CLI code.
2012-12-13 15:24 +0000 [r377972] David M. Lee <dlee@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac: Fixed
configure.ac to look for proper uuid.h file Introduced in
r377846, the configure script was looking for uuid.h instead of
uuid/uuid.h.
2012-12-13 15:22 +0000 [r377971] Brent Eagles <beagles@digium.com>
* configs/sip.conf.sample, channels/sip/include/sip.h,
channels/chan_sip.c: This change adds a SIP peer configuration
feature to allow the peer's configured codecs to take precedence
on an outgoing call. This change introduces a new peer
configuration property named 'ignore_requested_pref' that causes
the requested codec to be ignored when determining the preferred
codec for an outgoing call leg. The consequence is that
Asterisk's usual efforts to prefer avoiding transcoding can be
overridden on a peer-by-peer basis where appropriate.
2012-12-13 14:28 +0000 [r377966] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Ensure Min-SE is included in outbound
INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
value is not 90 (the default) and session timers are not
disabled. This has the effect of Asterisk following RFC4028 more
closely with regard to 422 responses and preventing situations in
which Asterisk would be forced to temporarily accept a call to
tear it down based on a Session-Expires below the locally
configured Min-SE. (issue SWP-5051) Review:
https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
Moore Patch-by: Kinsey Moore ........ Merged revisions 377946
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 377947 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377948 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-12 22:43 +0000 [r377925] Rusty Newton <rnewton@digium.com>
* sounds/Makefile, /: Incremented EXTRA_SOUNDS_VERSION in
sounds/Makefile to 1.4.12 for new Extra Sounds releases See
CHANGES-* files in English extra 1.4.12 tarballs for new sound
prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
(closes AST-755) Reported by: John Bigelow ........ Merged
revisions 377922 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377923 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377924 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-12 04:43 +0000 [r377915] Michael L. Young <elgueromexicano@gmail.com>
* main/features.c: Convert Dynamic Features Buffer To Use ast_str
Currently, the buffer for the dynamic features list is set to a
fixed size of 128. If the list is bigger than that, it results in
the dynamic feature(s) not being recognized. This patch changes
the buffer from a fixed size to a dynamic one. (closes issue
ASTERISK-20680) Reported by: Clod Patry Tested by: Michael L.
Young Patches: asterisk-20680-dynamic-features-v2.diff uploaded
by Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2221/
2012-12-12 00:02 +0000 [r377906-377911] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix a potential deadlock in chan_sip
during transfers. The issue comes from the fact that transfers
may perform a redirecting update on a channel. The issue is that
lock inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process. The fix
is to move when the redirecting update occurs to a place where
neither the tech_pvt or the channel is locked so that the two can
be locked in the proper order. (closes issue ASTERISK-20708)
reported by Mark Michelson patches: ASTERISK-20708-3.patch
uploaded by Mark Michelson (License #5049) Tested by: Tim
Ringenbach at Asteria Solutions Group ........ Merged revisions
377910 from http://svn.asterisk.org/svn/asterisk/branches/11
* main/features.c: Add test events necessary for bridging tests to
be able to properly run.
2012-12-11 22:03 +0000 [r377884] Richard Mudgett <rmudgett@digium.com>
* main/file.c, main/http.c, main/aoc.c, main/image.c, main/cel.c,
main/timing.c, main/channel.c, main/data.c, main/stun.c, /:
Cleanup CLI commands on exit for several files. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
unregister-cli-multiple-all.patch (license #5909) patch uploaded
by Corey Farrell ........ Merged revisions 377881 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377882 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377883 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-11 21:53 +0000 [r377878-377880] Mark Michelson <mmichelson@digium.com>
* /: And remove svnmerge-integrated property.
* /: Remove automerge properties.
2012-12-11 21:22 +0000 [r377867] Richard Mudgett <rmudgett@digium.com>
* main/udptl.c, /: Cleanup udptl on exit. * Cleanup CLI commands on
exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
uploaded by Corey Farrell Modified ........ Merged revisions
377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377848 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377849 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-11 21:04 +0000 [r377844-377846] Mark Michelson <mmichelson@digium.com>
* configure.ac, include/asterisk/uuid.h (added),
main/taskprocessor.c, tests/test_uuid.c (added), main/asterisk.c,
main/uuid.c (added), res/res_clialiases.c, /, configure,
include/asterisk/autoconfig.h.in, main/Makefile: Add UUID support
to Asterisk. This provides a common API for dealing with unique
identifiers. The API provides methods to create, parse, copy, and
stringify UUIDs. An accompanying unit test is provided that tests
all operations. (closes issue ASTERISK-20726) reported by Matt
Jordan Review: https://reviewboard.asterisk.org/r/2217
* res/res_clialiases.c, /: Fix crash that can occur if CLI
registration fails for an aliased command. A recent memory leak
fix in main/cli.c causes an ast_cli_entry's command field to be
freed and NULLed if ast_cli_register() fails. res_clialiases was
ignoring the return value of ast_cli_register() and was then
passing the NULL command off to a a hash function. This resulted
in a crash. The fix is not to ignore the erroneous return value.
If ast_cli_register() fails, then we do not continue trying to
process the current alias. ........ Merged revisions 377840 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377842 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377843 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-11 20:46 +0000 [r377707-377841] Richard Mudgett <rmudgett@digium.com>
* main/taskprocessor.c, /: Cleanup taskprocessor on exit. * Cleanup
CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey
Farrell Patches: taskprocessor-cleanup-1_8-11-trunk.patch
(license #5909) patch uploaded by Corey Farrell
taskprocessor-cleanup-10-only.patch (license #5909) patch
uploaded by Corey Farrell Modified ........ Merged revisions
377837 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377838 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377839 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/pbx.c, /: Cleanup pbx on exit. * Cleanup CLI commands on
exit. * Unreference hints and statecbs containers on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
uploaded by Corey Farrell Modified ........ Merged revisions
377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377807 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377808 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/logger.c: Cleanup logger on exit. * Cleanup CLI commands,
destroy verbosers and logchannels lists on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
logger-cleanup-all.patch (license #5909) patch uploaded by Corey
Farrell Modified ........ Merged revisions 377771 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377772 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377773 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/indications.c: Cleanup indications on exit. * Made
ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to
select the tone zone being unregistered again. * Ringcadence is
no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
commands and destroy default_tone_zone on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
indications-cleanup-all.patch (license #5909) patch uploaded by
Corey Farrell Modified ........ Merged revisions 377740 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377741 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377742 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/event.c: Cleanup event on exit. * Cleanup CLI commands on
exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
event_shutdown-10-only.patch (license #5909) patch uploaded by
Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909)
patch uploaded by Corey Farrell ........ Merged revisions 377708
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 377709 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377710 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/dnsmgr.c, /: Cleanup dnsmgr on exit. * Cleanup dnsmgr thread
and CLI commands on exit. (issue ASTERISK-20649) Reported by:
Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909)
patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
(license #5909) patch uploaded by Corey Farrell Modified ........
Merged revisions 377704 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377705 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377706 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-10 16:56 +0000 [r377626-377658] Kinsey Moore <kmoore@digium.com>
* /, res/res_fax.c: Ensure ReceiveFax provides a CED tone via T.38
When using res_fax_digium, the T.38 CED tone was not being
provided properly which would cause some incoming faxes to fail.
This was not an issue with res_fax_spandsp since it does not
strictly honor the send_ced flag and sends the CED tone whenever
receiving a T.38 fax. (closes issue FAX-343) Reported-by:
Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions
377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377656 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377657 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: Handle Session-Expires less than local
Min-SE in 200 OK Ensure that a call is immediately torn down if a
Session-Expires value received in a 200 OK is less than the local
Min-SE. This also prevents Asterisk from allowing calls with
Session-Expires below the RFC4028-mandated minimum (90s). (closes
issue ASTERISK-20653) Review:
https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore
........ Merged revisions 377623 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377624 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377625 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-10 07:03 +0000 [r377579-377595] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Add firmware information to CLI devices
listing
* channels/chan_unistim.c, /: Fix codec mismatch Fix code to send
in both rx and tx open stream messages correct codecs. Found that
on phase 0/1 phones wrong codecs cause to no audio in some
situations. (issue ASTERISK-20183) ........ Merged revisions
377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377592 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377593 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_unistim.c, /: Remove trailing whitespaces in number
from incoming redial list. Reported by: Igor Olhovskiy ........
Merged revisions 377577 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-10 01:41 +0000 [r377506-377512] Tilghman Lesher <tilghman@meg.abyt.es>
* main/xmldoc.c, /: Improve documentation by making all of the
colors used readable, no matter what the background color is.
Dark blue on a black background is unreadable, as is yellow on a
light background. This patch turns on the bright attribute for
colors when on a dark background and turns *off* the bright
attribute when the -W command line option is used (indicating a
_light_ background). This ensures that text is readable in both
cases. Patch by: tilghman Review:
https://reviewboard.asterisk.org/r/2224 ........ Merged revisions
377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377510 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377511 from
http://svn.asterisk.org/svn/asterisk/branches/11
* addons/cdr_mysql.c, /: Remove some dead code and additionally
handle a case that wasn't handled. ........ Merged revisions
377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377504 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377505 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-09 01:23 +0000 [r377463] Joshua Colp <jcolp@digium.com>
* channels/chan_motif.c, /: Add missing support for "who hung up"
to chan_motif. (closes issue ASTERISK-20671) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/2208/ ........
Merged revisions 377462 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-08 00:30 +0000 [r377402-377434] Richard Mudgett <rmudgett@digium.com>
* contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP
allow/disallow in MySQL contrib script. Using the contrib
sippeers.sql script to create the sippeers MySQL table would
result in being unable to place calls if you set the disallow
value to all. (closes issue ASTERISK-20756) Reported by: Andre
Luis Patches: sippeers.patch patch uploaded by Andre Luis
........ Merged revisions 377431 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377432 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377433 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/astmm.c, /: MALLOC_DEBUG: Only wait if we want atexit
allocation dumps. ........ Merged revisions 377398 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377399 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377401 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-07 22:08 +0000 [r377384] Kinsey Moore <kmoore@digium.com>
* codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder
show" CLI command. In r306010 "Asterisk media architecture
conversion - no more format bitfields", the logic for
incrementing encoders and decoders when opening transcoder
channels was changed without making the corresponding change when
decrementing encoder / decoder channels. The result being that
when a channel was destroyed, codec_dahdi couldn't properly tell
if it was an encoder or decoder, and the default case is to
assume it was a decoder. This could result in negative numbers
for decoders in use like in: VOIP6*CLI> transcoder show 2/-2
encoders/decoders of 92 channels are in use. (closes issue
ASTERISK-19921) Patch-by: Shaun Ruffell ........ Merged revisions
377382 from http://svn.asterisk.org/svn/asterisk/branches/10
........ Merged revisions 377383 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-07 00:00 +0000 [r377356] Richard Mudgett <rmudgett@digium.com>
* apps/confbridge/conf_config_parser.c, /, apps/app_confbridge.c:
confbridge: Fix some resource leaks on conference teardown. *
Made destroy_conference_bridge() destroy a missed ast_mutex_t and
ast_cond_t. * Made join_conference_bridge() init the
ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can
destroy them unconditionally. * Made join_conference_bridge()
abort if the new conference could not be added to the conferences
container. * Made leave_conference() discard any post-join
actions if join_conference_bridge() had to abort early. * Made
the join_conference_bridge() diagnostic messages better describe
what happened. * Renamed leave_conference_bridge() to
leave_conference() and made it only take a conference user
pointer. The conference pointer was redundant. * Made
conf_bridge_profile_copy() use struct copy instead of memcpy(). *
No need to lock the conference in start_conf_record_thread()
since all of the callers already have it locked. ........ Merged
revisions 377354 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377355 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-06 17:29 +0000 [r377329-377341] Russell Bryant <russell@russellbryant.com>
* /: Recorded merge of revisions 377340 from
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Add CLI
tab completion to 'acl show'. The 'acl show' CLI command allows
you to show the details about a specific named ACL in acl.conf.
This patch adds tab completion to the command. Review:
https://reviewboard.asterisk.org/r/2230/
* main/named_acl.c: Minor code cleanup in named_acl.c. This patch
makes a few little cleanups to named_acl.c. A couple non-public
functions were made static and an opening brace for a function
was moved to its own line, per the coding guidelines.
* main/named_acl.c: Add CLI tab completion to 'acl show'. The 'acl
show' CLI command allows you to show the details about a specific
named ACL in acl.conf. This patch adds tab completion to the
command. Review: https://reviewboard.asterisk.org/r/2230/
2012-12-06 14:26 +0000 [r377324] Matthew Jordan <mjordan@digium.com>
* main/manager.c, /: Fix memory leak in 'manager show event' when
command entered incorrectly When the CLI command 'manager show
event' was run incorrectly and its usage instructions returned, a
reference to the event container was leaked. This would prevent
the container from being reclaimed when Asterisk exits. We now
properly decrement the count on the ao2 object using the nifty
RAII_VAR macro. Thanks to Russell for helping me stumble on this,
and Terry for writing that ridiculously helpful macro. ........
Merged revisions 377319 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-05 17:17 +0000 [r377263] Jonathan Rose <jrose@digium.com>
* /, res/res_srtp.c: res_srtp: Fix a crash caused by srtp_dealloc
on an already dealloced session When srtp_create fails, the
session may be dealloced or just not alloced. At the same time
though, the session pointer might not be set to NULL in this
process and attempting to srtp_dealloc it again will cause a
segfault. This patch checks for failure of srtp_create and sets
the session pointer to NULL if it fails. (closes issue
ASTERISK-20499) Reported by: tootai Review:
https://reviewboard.asterisk.org/r/2228/ ........ Merged
revisions 377256 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377261 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377262 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-05 16:51 +0000 [r377260] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Fix a SIP request memory leak with TLS
connections. During the TLS re-work in chan_sip some TLS specific
code was moved into a separate function. This function operates
on a copy of the incoming SIP request. This copy was never
deinitialized causing a memory leak for each request processed.
This function is now given a SIP request structure which it can
use to copy the incoming request into. This reduces the amount of
memory allocations done since the internal allocated components
are reused between packets and also ensures the SIP request
structure is deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763) Reported by: deti ........ Merged
revisions 377257 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377258 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377259 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-05 02:23 +0000 [r377214-377246] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/_private.h, main/asterisk.c, main/format.c:
Remove init_framer(). It no longer does anything.
* main/format.c, /: Fix registering core show codecs/codec CLI
commands twice. ........ Merged revisions 377241 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377244 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/confbridge/conf_config_parser.c, /: confbridge: Fix several
small issues. * Made func_confbridge_helper() allow an empty
value when setting options. You previously could not
Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the
dialplan. * Made func_confbridge_helper() handle its datastore
better if multiple threads attempt to set the first CONFBRIDGE
option value on the channel. * Made the func_confbridge_helper()
only output one diagnostic message concerning the option. * Made
the bridge video_mode able to repeatedly change in the config
file and CONFBRIDGE dialplan function. The video_mode option
values are an enum and not independent of each other. * Made
handle_cli_confbridge_show_bridge_profile() better handle the
video_mode option. * Simplified datastore handling code in
conf_find_user_profile() and conf_find_bridge_profile(). (closes
issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter
........ Merged revisions 377227 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377228 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_confbridge.c: confbridge: Update online XML
documentation. ........ Merged revisions 377212 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377213 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-04 13:01 +0000 [r377196] Russell Bryant <russell@russellbryant.com>
* contrib/scripts/install_prereq, /: Add libuuid to install_prereq
for Fedora. I ran this script and my build failed. pjproject
requires this. ........ Merged revisions 377195 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-03 23:00 +0000 [r377040-377168] Richard Mudgett <rmudgett@digium.com>
* /, main/asterisk.c: Cleanup ast_run_atexits() atexits list. *
Convert atexits list to a mutex instead of a rd/wr lock. The lock
is only write locked. * Move CLI verbose Asterisk ending message
to where AMI message is output in really_quit() to avoid further
surprises about using stuff already shutdown. (issue
ASTERISK-20649) Reported by: Corey Farrell ........ Merged
revisions 377165 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377166 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377167 from
http://svn.asterisk.org/svn/asterisk/branches/11
* include/asterisk/_private.h, main/stdtime/localtime.c,
main/asterisk.c, /: Cleanup core main on exit. * Cleanup time
zones on exit. * Make exit clean/unclean report consistent for
AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported by:
Corey Farrell Patches: core-cleanup-1_8-10.patch (license #5909)
patch uploaded by Corey Farrell core-cleanup-11-trunk.patch
(license #5909) patch uploaded by Corey Farrell Modified ........
Merged revisions 377135 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377136 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377137 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/config.c, /: Cleanup config cache on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
config-cleanup-all.patch (license #5909) patch uploaded by Corey
Farrell ........ Merged revisions 377104 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377105 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377106 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/cli.c, /: Cleanup CLI resources on exit and CLI command
registration errors. (issue ASTERISK-20649) Reported by: Corey
Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
#5909) patch uploaded by Corey Farrell Modified ........ Merged
revisions 377073 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377074 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377075 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/cdr.c, /: Cleanup CDR resources on exit. * Simplify
do_reload() return handling since it never returned anything
other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell
Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by
Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
uploaded by Corey Farrell Modified ........ Merged revisions
377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 377070 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377071 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/ccss.c: Fix CCSS CLI commands and logger level not
unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
Corey Farrell ........ Merged revisions 377037 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 377038 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 377039 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-03 16:45 +0000 [r377035] Olle Johansson <oej@edvina.net>
* res/res_rtp_asterisk.c: Formatting fixes
2012-12-03 14:56 +0000 [r377022] Joshua Colp <jcolp@digium.com>
* channels/chan_motif.c, /: Fix an RTP instance reference count
leak in chan_motif. When setting up an RTP instance the RTCP
portion of the instance keeps a reference to the instance itself.
In order to release this reference and stop RTCP the stop API
call must be called before destroying the instance. (closes issue
ASTERISK-20751) Reported by: joshoa ........ Merged revisions
377021 from http://svn.asterisk.org/svn/asterisk/branches/11
2012-12-03 14:46 +0000 [r376998-377018] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Move functions to AFTER the block of forward
declarations of functions. It was a mess. The first part of
chan_sip.c is constants, declarations, structures and stuff, then
forward declarations and then actual code. It's still a mess, but
a bit less messy ;-)
* channels/chan_sip.c, res/res_rtp_asterisk.c: Formatting changes
Found a large amount of missing {} in the code before patching in
another branch
2012-12-01 00:47 +0000 [r376984] Joshua Colp <jcolp@digium.com>
* configs/motif.conf.sample, /, channels/chan_motif.c: Tweak
extension used for incoming calls received on Motif. Based on
feedback from numerous individuals this patch tweaks incoming
calls to first look for an extension with the name of the
endpoint. If no such extension exists the call will silently fall
back to the "s" extension as it previously did. ........ Merged
revisions 376983 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-30 21:38 +0000 [r376953] Richard Mudgett <rmudgett@digium.com>
* /, channels/misdn/isdn_lib.c: chan_misdn: Fix sending
RELEASE_COMPLETE in response to SETUP. Fix sending a
RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
have a B channel available to assign to the call. (closes issue
ABE-2869) Reported by: Guenther Kelleter Patches:
setup-reject_2.diff (license #6372) patch uploaded by Guenther
Kelleter Modified ........ Merged revision 376949 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........ Merged revisions 376950 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376951 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376952 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-30 17:08 +0000 [r376922] Sean Bright <sean@malleable.com>
* /, funcs/func_volume.c: Minor spelling fix to the VOLUME
documentation. ........ Merged revisions 376919 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376920 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376921 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-30 16:56 +0000 [r376918] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix potential crashes during SIP attended
transfers. The principal behind this patch is simple. During a
transfer, we manipulate channels that are owned by a separate
thread than the one we currently are running in, so it makes
sense that we need to grab a reference to the channels so that
they cannot disappear out from under us. In the wild, crashes
were sometimes seen when the transferring party would hang up the
call before the transfer target answered the call. The most
common place to see the crash occur was when attempting to send a
connected line update to the transferer channel. (closes issue
ASTERISK-20226) Reported by Jared Smith Patches:
ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
Tested by: Jared Smith ........ Merged revisions 376901 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376916 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376917 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-29 23:01 +0000 [r376867-376871] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in
local_devicestate(). Regression introduced by ASTERISK-20390 fix.
(closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
rmudgett ........ Merged revisions 376868 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376869 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376870 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
........ Merged revisions 376864 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376865 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376866 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-29 21:58 +0000 [r376837] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Improve Code Readability And Fix Setting
natdetected Flag For 1.8, 10, 11 and trunk we are are improving
the code readability. For 11 and trunk, auto nat detection was
added. The natdetected flag was being set to 1 when the host
address in the VIA header did not specifiy a port. This patch
fixes this by setting the port on the temporary sock address used
to SIP_STANDARD_PORT in order for the sock address comparison to
work properly. (closes issue ASTERISK-20724) Reported by: Michael
L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2206/ ........ Merged
revisions 376834 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376835 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376836 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-29 17:16 +0000 [r376821] David M. Lee <dlee@digium.com>
* main/utils.c: Fixed ast_random's comment about locking. The
original comment was separated from the code at some point, and
didn't reflect the use of libc's other than glibc for Linux.
2012-11-29 16:44 +0000 [r376820] Pedro Kiefer <pedro@kiefer.com.br>
* channels/chan_sip.c: Fix chan_sip websocket payload handling
Websocket by default doesn't return an ast_str for the payload
received. When converting it to an ast_str on chan_sip the last
character was being omitted, because ast_str functions expects
that the given length includes the trailing 0x00. payload_len
only has the actual string length without counting the trailing
zero. For most cases this passed unnoticed as most of SIP
messages ends with \r\n. (closes issue ASTERISK-20745) Reported
by: Iñaki Baz Castillo Review:
https://reviewboard.asterisk.org/r/2219/
2012-11-29 00:48 +0000 [r376761-376791] Richard Mudgett <rmudgett@digium.com>
* main/astmm.c, main/asterisk.c, /: Add MALLOC_DEBUG atexit
unreleased malloc memory summary. * Adds the following CLI
commands to control MALLOC_DEBUG reporting of unreleased malloc
memory when Asterisk is shut down. memory atexit list on memory
atexit list off memory atexit summary byline memory atexit
summary byfunc memory atexit summary byfile memory atexit summary
off * Made check all remaining allocated region blocks atexit for
fence violations. * Increased the allocated region hash table
size by about three times. It still isn't large enough
considering the number of malloced blocks Asterisk uses. * Made
CLI "memory show allocations anomalies" use
regions_check_all_fences(). Review:
https://reviewboard.asterisk.org/r/2196/ ........ Merged
revisions 376788 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376789 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376790 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
"memory show allocations" misspelling of anomalies option. The
command will still accept the original misspelling. *
Miscellaneous tweaks to CLI "memory show allocations" command
output format. * Made CLI "memory show summary" summarize by line
number instead of by function if a filename is given. * Made CLI
"memory show summary" sort its output by filename or
function-name/line-number depending upon request. * Miscellaneous
tweaks to CLI "memory show summary" command output format.
........ Merged revisions 376758 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376759 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376760 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-28 16:47 +0000 [r376728] Jonathan Rose <jrose@digium.com>
* main/manager.c, /: manager: Make challenge work with
allowmultiplelogin=no Prior to this patch, challenge would yield
a multiple logins error if used without providing the username
(which isn't really supposed to be an argument to challenge) if
allowmultiplelogin was set to no because allowmultiplelogin finds
a user with a zero length login name. This check is simply
disabled for the challenge action when the username is empty by
this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
Patches: challenge_action_nomultiplelogin.diff uploaded by
Jonathan Rose (license 6182) ........ Merged revisions 376725
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 376726 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376727 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-28 00:13 +0000 [r376630-376691] Richard Mudgett <rmudgett@digium.com>
* UPGRADE.txt, main/pbx.c, /: Fix extension matching with the '-'
char. The '-' char is supposed to be ignored by the dialplan
extension matching. Unfortunately, it's treatment is not handled
consistently throughout the extension matching code. * Made the
old exten matching code consistently ignore '-' chars. * Made the
old exten matching code consistently handle case in the matching.
* Made ignore empty character sets. * Fixed ast_extension_cmp()
to return -1, 0, or 1 as documented. The only user of it in
pbx_lua.c was testing for -1. It was originally returning the
strcmp() value for less than which is not usually going to be -1.
* Fix character set sorting if the sets have the same number of
characters and start with the same character. Character set [0-9]
now sorts before [02-9a] as originally intended. * Updated some
extension label and priority already in use warnings to also
indicate if the extension is aliased. (closes issue
ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
Harzenetter Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/2201/ ........ Merged
revisions 376688 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376689 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376690 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_celgenuserevent.c, pbx/pbx_dundi.c,
addons/res_config_mysql.c: Remove unnecessary channel module
references. * Removed call to ast_module_user_hangup_all() in
res_config_mysql.c since it is effectively a noop. No channels
can attach a reference to that module. * Removed call to
ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
of unload_module() has already called it. * Removed redundant
channel module references in pbx_dundi.c. The registered dialplan
function callback dispatchers for the read/read2/write callbacks
already reference the module before calling. * pbx_dundi: Moved
unregistering CLI commands, DUNDi switch, and dialplan functions
to the first thing the unload_module() does. This will reduce the
chance of new channels using DUNDi services while the module is
being torn down. ........ Merged revisions 376657 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376658 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376659 from
http://svn.asterisk.org/svn/asterisk/branches/11
* include/asterisk/linkedlists.h, /: Made AST_LIST_REMOVE() simpler
and use better names. * Update doxygen of AST_LIST_REMOVE().
........ Merged revisions 376627 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376628 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376629 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-23 00:02 +0000 [r376589] Matthew Jordan <mjordan@digium.com>
* main/lock.c, /, main/logger.c, include/asterisk/lock.h:
Re-initialize logmsgs mutex upon logger initialization to prevent
lock errors Similar to the patch that moved the fork earlier in
the startup sequence to prevent mutex errors in the recursive
mutex surrounding the read/write thread registration lock, this
patch re-initializes the logmsgs mutex. Part of the start up
sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to
daemon in order to read startup parameters. When reading in a
conf file, log statements can be generated. Since this can't be
avoided, the mutex instead is re-initialized to ensure a reset of
any thread tracking information. This patch also includes some
additional debugging to catch errors when locking or unlocking
the recursive mutex that surrounds locks when the DEBUG_THREADS
build option is enabled. DO_CRASH or THREAD_CRASH will cause an
abort() if a mutex error is detected. (issue ASTERISK-19463)
Reported by: mjordan Tesetd by: mjordan ........ Merged revisions
376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 376587 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376588 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-21 18:33 +0000 [r376575] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c, main/astobj2.c, include/asterisk/test.h,
main/channel.c, include/asterisk/astobj2.h, main/test.c,
tests/test_astobj2.c: Add red-black tree container type to
astobj2. * Add red-black tree container type. * Add CLI command
"astobj2 container dump <name>" * Added ao2_container_dump() so
the container could be dumped by other modules for debugging
purposes. * Changed ao2_container_stats() so it can be used by
other modules like ao2_container_check() for debugging purposes.
* Updated the unit tests to check red-black tree containers.
(closes issue ASTERISK-19970) Reported by: rmudgett Tested by:
rmudgett Review: https://reviewboard.asterisk.org/r/2110/
2012-11-20 22:06 +0000 [r376562] David M. Lee <dlee@digium.com>
* res/res_http_websocket.c, /: Added missing newlines to websocket
ast_logs. Without these newlines, log messages just continue
tacking onto the same line, and do not flush immediately.
........ Merged revisions 376561 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-20 19:09 +0000 [r376551] Mark Michelson <mmichelson@digium.com>
* channels/sip/include/sip.h, /, channels/chan_sip.c: Add "Require:
timer" to 200 OK responses when appropriate. The method by which
the Require header is added to 200 responses is inspired by the
method that Olle Johansson uses in his darjeeling-prack branch.
(closes issue ASTERISK-20570) Reported by Matt Jordan, at the
behest of Olle Johansson Review:
https://reviewboard.asterisk.org/r/2172 ........ Merged revisions
376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 376522 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376550 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-20 17:39 +0000 [r376541] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_sip.c: Reduce CLI spam of "Extension Changed"
device state messages. Asterisk 11 follows RFC3265 that states
that after every subscribe or resubscribe a notify should be
sent. Thus the console if filled continuously with the following
after every subscribe; == Extension Changed 8512[phones] new
state IDLE for Notify User cisco1 In Asterisk 1.8 only changes
would be sent. Thus only when a device state changed was anything
emitted to the console. fix: Only print to console when device
state isn't forced. (closes issue ASTERISK-20706) Reported by:
alecdavis Tested by: alecdavis alecdavis (license 585) ........
Merged revisions 376540 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-19 20:03 +0000 [r376472] Walter Doekes <walter+asterisk@wjd.nu>
* main/indications.c, /, channels/chan_sip.c,
main/security_events.c: Fix most leftover non-opaque ast_str
uses. Instead of calling str->str, one should use
ast_str_buffer(str). Same goes for str->used as
ast_str_strlen(str) and str->len as ast_str_size(str). Review:
https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 376470 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376471 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-19 02:14 +0000 [r376416-376457] Matthew Jordan <mjordan@digium.com>
* tests/test_astobj2.c: Fix uninitialized in this function error
With some versions of gcc, n_buckets will be flagged as being
uninitialized before use. While its technically impossible (since
the switch statement, even without a default, accounts for all
possibilities), we'll initialize the variable to 0 anyway.
* main/asterisk.c, /, main/utils.c: Reorder startup sequence to
prevent lockups when process is sent to background Although it is
very rare and timing dependent, the potential exists for the call
to 'daemon' to cause what appears to be a deadlock in Asterisk
during startup. This can occur when a recursive mutex is obtained
prior to the daemon call executing. Since daemon uses fork to
send the process into the background, any threading primitives
are unsafe to re-use after the call. Implementations of pthread
recursive mutexes are highly likely to store the thread
identifier of the thread that previously obtained the mutex. If
the mutex was locked prior to the fork, a subsequent unlock
operation will potentially fail as the thread identifier is no
longer valid. Since the mutex is still locked, all subsequent
attempts to grab the mutex by other threads will block. This
behavior exhibited itself most often when DEBUG_THREADS was
enabled, as this compile time option surrounds the mutexes in
Asterisk with another recursive mutex that protects the storage
of thread related information. This made it much more likely that
a recursive mutex would be obtained prior to daemon and unlocked
after the call. This patch does the following: a) It backports a
patch from Asterisk 11 that prevents the spawning of the
localtime monitoring thread. This thread is now spawned after
Asterisk has fully booted. b) It re-orders the startup sequence
to call daemon earlier during Asterisk startup. This limits the
potential of threading primitives being accessed by
initialization calls before daemon is called. c) It removes calls
to ast_verbose/ast_log/etc. prior to daemon being called.
Developers should send error messages directly to stderr prior to
daemon, as calls to ast_log may access recursive mutexes that
store thread related information. d) It reorganizes when thread
local storage is created for storing lock information during the
creation of threads. Prior to this patch, the read/write lock
protecting the list of threads in ast_register_thread would
utilize the lock in the thread local storage prior to it being
initialized; this patch prevents that. On a very related note,
this patch will *greatly* improve the stability of the Asterisk
Test Suite. Review: https://reviewboard.asterisk.org/r/2197
(closes issue ASTERISK-19463) Reported by: mjordan Tested by:
mjordan ........ Merged revisions 376428 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376431 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376441 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/confbridge/conf_state.c, /: Add a test event that reports
changes in ConfBridge state This patch adds a test event to
ConfBridge that reports transitions between states in ConfBridge.
This is used by tests in the Asterisk Test Suite that verify
state changes based on the entering/leaving of conference
participants. ........ Merged revisions 376414 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376415 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-16 00:15 +0000 [r376341-376345] David M. Lee <dlee@digium.com>
* /, utils/extconf.c: Fixed extconf.c breakage introduced in
r376306. To quote wdoekes: > Note that I'm not confirming
legitimacy of having that file in tree at > all. Is anyone using
aelparse/conf2ael? ........ Merged revisions 376340 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376342 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376343 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /: Somehow I put in svn-1.6 merge information. Oops.
* utils/Makefile, tests/test_astobj2_thrash.c (added),
utils/utils.xml, /, utils/hashtest.c (removed),
tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed),
include/asterisk/hashtab.h: Migrate hashtest/hashtest2 to be unit
tests. Both hashtest and hashtest2 are manual testing apps that
thrash hash tables (hashtab and ao2 containers, respectively), by
spinning up several threads that randomly insert, delete, lookup
and iterate over the hash table. If the app doesn't crash, the
hash table probably passes the test. Those utils are not a part
of the typical Asterisk build, so they do not usually get
compiled. This all makes them less that useful. This patch
removes those manual test programs and replaces them with
Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
also attempts to make the tests more deterministic. * Rather than
spinning up some number of threads that operate on the hash table
randomly, spin up four threads that concurrenly add, remove,
lookup and iterate over the hash table. * Each thread checks the
state of the hash table both during and after execution, and
indicates a test failure if things are not as expected. * Each
thread times out after 60 seconds to prevent deadlocking the unit
test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
revisions 376306 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376315 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376339 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-15 23:10 +0000 [r376312] Jonathan Rose <jrose@digium.com>
* /, apps/app_meetme.c: app_meetme: Fix channels lingering when
hung up under certain conditions Channels would get stuck and
MeetMe would repeatedly display an Unable to write frame to
channel error in the conf_run function if hung up during certain
sound prompts such as during user count announcements. This patch
fixes that by reintroducing a hangup check in the meetme's main
loop (also in conf_run). (closes issue ASTERISK-20486) Reported
by: Michael Cargile Review:
https://reviewboard.asterisk.org/r/2187/ Patches:
meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan
Rose (license 6182) ........ Merged revisions 376307 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376308 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376310 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-15 14:35 +0000 [r376291] Brent Eagles <beagles@digium.com>
* main/channel.c, /: Patch to prevent stopping the active generator
when it is not the silence generator. This patch introduces an
internal helper function to safely check whether the current
generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator()
function has been modified to be implemented in terms of the new
function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad
........ Merged revisions 376217 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-15 02:29 +0000 [r376282] Rusty Newton <rnewton@digium.com>
* apps/app_voicemail.c, /: Patch to play correct sound file when a
voicemail's urgent status is removed We were attempting to play
"vm-urgent-removed", which didn't exist. Now we play
"vm-marked-nonurgent" which exists and is the correct sound file.
Previous behavior was silence and a warning on the CLI. (issue
ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
uploaded by Rusty Newton (license 5829) ........ Merged revisions
376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 376263 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376264 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-14 19:55 +0000 [r376235] Richard Mudgett <rmudgett@digium.com>
* pbx/pbx_spool.c, /: Fix call files when astspooldir is relative.
Future dated call files are ignored when astspooldir is relative
to the current directory. The queue_file() assumed that the qdir
needed to be prepended if the given filename did not start with a
'/'. If astspooldir is relative it is not going to start from the
root directory obviously so it will not start with a '/'. The
filename used in queue_file() ultimately results in qdir
prepended multiple times. * Made queue_file() not prepend qdir if
the filename contains a '/'. (closes issue ASTERISK-20593)
Reported by: James Le Cuirot Patches:
0004-Fix-future-call-files-from-relative-directories.patch
(license #6439) patch uploaded by James Le Cuirot ........ Merged
revisions 376232 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376233 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376234 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-13 19:42 +0000 [r376219] Jonathan Rose <jrose@digium.com>
* CHANGES, channels/chan_sip.c: chan_sip: Add SubscribeContext
field to SIPshowpeer AMI response The new field is will show up
within the response if the requested peer has a subscribe context
set. (closes issue ASTERISK-20626) Reported by: Jaco Kroon
Patches: asterisk-sip-ami-SubscrContext.patch uploaded by jkroon
(license 5671) -with modifications by jrose to conform to style
guidelines Review: https://reviewboard.asterisk.org/r/2195/
2012-11-12 20:46 +0000 [r376169] Joshua Colp <jcolp@digium.com>
* /, main/pbx.c: Properly check if the "Context" and "Extension"
headers are empty in a ShowDialPlan action. The code which
handles the ShowDialPlan action wrongly assumed that a non-NULL
return value from the function which retrieves headers from an
action indicates that the header has a value. This is incorrect
and the contents must be checked to see if they are blank.
(closes issue ASTERISK-20628) Reported by: jkroon Patches:
asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
........ Merged revisions 376166 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376167 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376168 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-12 20:18 +0000 [r376148] Michael L. Young <elgueromexicano@gmail.com>
* main/pbx.c, /: Fix Dynamic Hints Variable Substition - Underscore
Problem When adding a dynamic hint, if an extension contains an
underscore no variable subsitution is being performed. This patch
changes from checking if the extension contains an underscore to
checking if the extension begins with an underscore. (closes
issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
Steven T. Wheeler, Michael L. Young Patches:
asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2188/ ........ Merged
revisions 376142 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376143 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376144 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-11 17:15 +0000 [r376131] Joshua Colp <jcolp@digium.com>
* configs/sip.conf.sample, res/res_rtp_asterisk.c, /,
channels/chan_sip.c: Remove a fixed size limitation for producing
SDP and change how ICE support is disabled by default. With ICE
support enabled in chan_sip and a large number of interfaces on
the system it was possible for the produced SDP to be truncated
due to some fixed size buffers. These buffers have now been
changed so they will dynamically grow as needed. ICE support is
now also enabled by default in res_rtp_asterisk to provide a
smoother experience for chan_motif users where it is required. To
maintain the previous behavior in chan_sip it is no longer
enabled by default there. (closes issue ASTERISK-20643) Reported
by: coopvr ........ Merged revisions 376130 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-08 22:10 +0000 [r376092] Mark Michelson <mmichelson@digium.com>
* /, res/res_fax.c: Fix a "set but not used" warning on newer gccs.
Turns out the "helpful" setting of ms and res in this macro is
completely useless after the timeout antipattern fix. If you're a
new guy looking to write code, don't write a macro like this one.
........ Merged revisions 376087 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376088 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376089 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-08 21:12 +0000 [r376049-376061] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_ss7.c: chan_dahdi/SS7: Made reject incoming call
for an in-alarm or blocked channel. If a SS7 call comes in
requesting a CIC that is in-alarm, the call is accepted and
connects if the extension exists in the dialplan. The call does
not have any audio. * Made release the call immediately with
circuit congestion cause. (closes issue ASTERISK-20204) Reported
by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license
#5621) patch uploaded by rmudgett ........ Merged revisions
376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 376059 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376060 from
http://svn.asterisk.org/svn/asterisk/branches/11
* include/asterisk/utils.h, include/asterisk/astmm.h, /,
main/utils.c, main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG
enhancements. * Makes malloc() behave like calloc(). It will
return a memory block filled with 0x55. A nonzero value. * Makes
free() fill the released memory block and boundary fence's with
0xdeaddead. Any pointer use after free is going to have a pointer
pointing to 0xdeaddead. The 0xdeaddead pointer is usually an
invalid memory address so a crash is expected. * Puts the freed
memory block into a circular array so it is not reused
immediately. * When the circular array rotates out a memory block
to the heap it checks that the memory has not been altered from
0xdeaddead. * Made the astmm_log message wording better. * Made
crash if the DO_CRASH menuselect option is enabled and something
is found. * Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms. * Extracted region_check_fences() from
__ast_free_region() and handle_memory_show(). * Updated
handle_memory_show() CLI usage help. Review:
https://reviewboard.asterisk.org/r/2182/ ........ Merged
revisions 376029 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 376030 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376048 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-07 19:15 +0000 [r376015] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, main/pbx.c, main/rtp_engine.c, /,
apps/app_meetme.c, res/res_fax.c, apps/app_record.c,
channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
apps/app_queue.c, channels/sig_pri.c, channels/chan_iax2.c,
main/channel.c, channels/chan_dahdi.c, apps/app_waitforring.c,
channels/sig_analog.c, apps/app_jack.c, include/asterisk/time.h:
Multiple revisions 375993-375994 ........ r375993 | mmichelson |
2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix
misuses of timeouts throughout the code. Prior to this change, a
common method for determining if a timeout was reached was to
call a function such as ast_waitfor_n() and inspect the out
parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around. The
problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any.
When this happens thousands of times, the result is that the
timeout takes much longer than intended to be reached. As an
example, I had a situation where a 3 second timeout took multiple
days to finally end since most wakeups from ast_waitfor_n() were
under a millisecond. This patch seeks to fix this pattern
throughout the code. Now we log the time when an operation began
and find the difference in wall clock time between now and when
the event started. This means that sub-millisecond timeouts now
cannot play havoc when trying to determine if something has timed
out. Part of this fix also includes changing the function
ast_waitfor() so that it is possible for it to return less than
zero when a negative timeout is given to it. This makes it
actually possible to detect errors in ast_waitfor() when there is
no timeout. (closes issue ASTERISK-20414) reported by David M.
Lee Review: https://reviewboard.asterisk.org/r/2135/ ........
r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov
2012) | 3 lines Remove some debugging that accidentally made it
in the last commit. ........ Merged revisions 375993-375994 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375995 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 376014 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-06 19:05 +0000 [r375967] Richard Mudgett <rmudgett@digium.com>
* /, main/channel_internal_api.c, main/features.c,
include/asterisk/channel.h, include/asterisk/features.h,
main/channel.c: Fix stuck DTMF when bridge is broken. When a
bridge is broken by an AMI Redirect action or the ChannelRedirect
application, an in progress DTMF digit could be stuck sending
forever. * Made simulate a DTMF end event when a bridge is broken
and a DTMF digit was in progress. (closes issue ASTERISK-20492)
Reported by: Jeremiah Gowdy Patches: bridge_end_dtmf-v3.patch.txt
(license #6358) patch uploaded by Jeremiah Gowdy Modified to
jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch
(license #5621) patch uploaded by rmudgett Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2169/ ........ Merged
revisions 375964 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375965 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375966 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-06 12:15 +0000 [r375926] Joshua Colp <jcolp@digium.com>
* /, channels/chan_motif.c: Fix a bug where our Motif ICE
candidates were not quite proper, and make us more forgiving. An
issue was reported on the mailing list where calling would result
in an "Incomplete ICE-UDP candidate received on session" error
message. This is the result of the ICE-UDP candidate code not
placing a "network" attribute within the candidates. This is now
done. To increase compatibility though I have removed the
requirement for the "network" attribute to exist within ICE-UDP
candidates that are received since we don't actually require the
value. Reported on the mailing list by Jean-Denis Girard.
........ Merged revisions 375925 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-05 23:10 +0000 [r375896] Matthew Jordan <mjordan@digium.com>
* channels/chan_iax2.c, res/res_fax_spandsp.c,
res/res_timing_kqueue.c, main/timing.c, main/channel.c, /,
res/res_timing_pthread.c, res/res_timing_dahdi.c,
res/res_timing_timerfd.c, bridges/bridge_softmix.c,
funcs/func_jitterbuffer.c, include/asterisk/timing.h,
res/res_musiconhold.c: Refactor ast_timer_ack to return an error
and handle the error in timer users Currently, if an
acknowledgement of a timer fails Asterisk will not realize that a
serious error occurred and will continue attempting to use the
timer's file descriptor. This can lead to situations where errors
stream to the CLI/log file. This consumes significant resources,
masks the actual problem that occurred (whatever caused the timer
to fail in the first place), and can leave channels in odd
states. This patch propagates the errors in the timing resource
modules up through the timer core, and makes users of these
timers handle acknowledgement failures. It also adds some
defensive coding around the use of timers to prevent using bad
file descriptors in off nominal code paths. Note that the patch
created by the issue reporter was modified slightly for this
commit and backported to 1.8, as it was originally written for
Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license
6358) ........ Merged revisions 375893 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375894 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375895 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-05 21:42 +0000 [r375865] Richard Mudgett <rmudgett@digium.com>
* main/loader.c, /: Add safety NULL pointer check in module user
references. Made __ast_module_user_remove() check for NULL
pointers. ........ Merged revision 375860 from C.3 ........
Merged revisions 375862 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375863 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375864 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-05 18:00 +0000 [r375848] Jonathan Rose <jrose@digium.com>
* /, UPGRADE.txt: chan_sip: Document a change to user-field
encoding introduced with r303509 The change in question was added
to improve compliance with RFC3261, but at the time of commit, it
wasn't adequately documented in the UPGRADE notes. (closes issue
ASTERISK-20561) Reported by: Deniz Review:
https://reviewboard.asterisk.org/r/2177/ ........ Merged
revisions 375846 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375847 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-04 03:10 +0000 [r375730-375803] Matthew Jordan <mjordan@digium.com>
* main/manager.c, /: Don't attempt to purge sessions when no
sessions exist Manager's tcp/tls objects have a periodic function
that purge old manager sessions periodically. During shutdown,
the underlying container holding those sessions can be disposed
of and set to NULL before the tcp/tls periodic function is
stopped. If the periodic function fires, it will attempt to
iterate over a NULL container. This patch checks for whether or
not the sessions container exists before attempting to purge
sessions out of it. If the sessions container is NULL, we simply
return. Note that this error was also caught by the Asterisk Test
Suite. ........ Merged revisions 375800 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375801 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375802 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, res/res_fax.c: Only deref a reserved gateway session if we
actually reserved one Its perfectly acceptable to have a gateway
session unreserved when we go to first allocate one. Unreffing
the reserved gateway session - when its NULL - will result in an
assertion error. This problem was caught by the Asterisk Test
Suite (once we had enough of the debugging flags enabled)
........ Merged revisions 375797 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375798 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/manager.c: Properly clean up manager resources on exit
This patch does two things: 1) It properly unregisters the
manager CLI commands 2) It cleans up AMI users on exit. Prior to
this patch, the AMI users were not being disposed of properly,
resulting in a memory leak. (closes issue ASTERISK-20646)
Reported by: Corey Farrell patches: manager_shutdown.patch
uploaded by Corey Farrell (license 5909) ........ Merged
revisions 375793 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375794 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375795 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/db.c: Properly finalize prepared SQLite3 statements to
prevent memory leak The AstDB uses prepared SQLite3 statements to
retrieve data from the SQLite3 database. These statements should
be finalized during Asterisk shutdown so that the SQLite3
database can be properly closed. Failure to finalize the
statements results in a memory leak and a failure when closing
the database. This patch fixes those issues by ensuring that all
prepared statements are properly finalized at shutdown. (closes
issue ASTERISK-20647) Reported by: Corey Farrell patches:
astdb-sqlite3_close.patch uploaded by Corey Farrell (license
5909) ........ Merged revisions 375761 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375763 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/xmldoc.c, /: Fix memory leaks in XML documentation This
patch fixes two memory leaks: 1) When building XML documentation
items, the 'name' attribute was extracted from XML elements but
not properly freed after being copied into the item being built.
2) When unloading XML documentation, the doctree container
objects were not properly freed. This patch corrects these memory
leaks. Note that this patch was modified slightly for this
commmit, as the case where the 'name' attribute doesn't exist
also wasn't handled in the item construction. This patch also
checks for that attribute not existing. (closes issue
ASTERISK-20648) Reported by: Corey Farrell Tested by: mjordan
patches: xmldoc-memory_leak.patch uploaded by Corey Farrell
(license 5909) ........ Merged revisions 375756 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/cdr.c, /: Prevent multiple CDR batches from conflicting when
scheduling the CDR write The Asterisk Test Suite caught an error
condition where a scheduled CDR batch write can be deleted twice
if two channels attempt to post their CDRs at the same time. The
batch CDR mutex is locked while the CDRs are appended to the
current batch list; however, it is unlocked prior to actually
scheduling the CDR write. As such, two threads can attempt to
remove the currently scheduled batch write at the same time,
resulting in an assertion error. This patch extends the time that
the mutex is locked to encompass actually scheduling the write.
This prevents two threads from unscheduling the currently
scheduled write at the same time. ........ Merged revisions
375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 375728 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375729 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-02 21:03 +0000 [r375663] Damien Wedhorn <voip@facts.com.au>
* /, channels/chan_skinny.c: Fix for chan_skinny leaving RTP ports
open Skinny wasn't closing RTP sockets. This patch includes
ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes
the problem. Also add destroy for VRTP (which I believe is
unused, but exists). Review:
https://reviewboard.asterisk.org/r/2176/ ........ Merged
revisions 375660 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-02 21:01 +0000 [r375628-375662] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, channels/chan_misdn.c, /, main/ccss.c,
main/format_pref.c: Things don't need to be that const. ........
Merged revisions 375658 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375659 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375661 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Multiple
revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
primitives must be handled first. The frm->addr is a different
"address space" than the stack/instance address of other Lx
primitives. The test for B channel instance address could fail.
Patches: patch01_timers.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
chan_misdn: Free memory in error paths and other memory leaks.
The one line commented with BUG is not easily fixable because
there is no de-init function one can call. Patches:
patch02_memory.diff (license #6372) patch uploaded by Guenther
Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
L2 de-establish/establish * An NT-PTMP cannot de/establish L2
since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
is finally active in handle_l1. * L2 deactivation logging
cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
as "UNKN". * Removed unused functions and code for L2 handling.
Patches: patch03_L2estab.diff (license #6372) patch uploaded by
Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
prim via lower_id layer (3 or 1) simply does not work. For TE (3)
it returns an error (len=-6) which is not evaluated by
handle_l1(), so the L1 layer status ends up wrong. Instead PH
must be sent via L4, only then does it reach L1 without an error
message. And NT PH prims only reach L1 when they are sent to
layer 2 id. --> use upper_id to send PH primitives. * Check for
errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
improved. * The lower_id is now not used for anything, except:
Why is lower_id layer deleted when it wasn't created? I removed
this code since it looks very wrong. Patches:
patch04_l1activation.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
chan_misdn: Fix loss of B channels if L1 is down. If you make 2
calls out an NT PTMP port which is not connected to any phone,
the B channel associated with that call becomes unusable until
Asterisk is restarted. The problem is the EVENT_SETUP is queued
when L1 is not up in misdn_lib_send_event(). If L1 cannot be
activated the event won't be dequeued. It gets even worse when
the call is hung up. The queued EVENT_SETUP will be overwritten
by an EVENT_DISCONNECT. The reserved B channel then will never be
freed. If later someone connects a phone to the port, L1 will
eventually activate and the queued EVENT_DISCONNECT is sent down
the stack. However, it is ignored because it is the wrong call
state. The real fix would be that activation and queueing for a
new SETUP is done by the NT stack. But since it doesn't, the
workaround must be removed because it doesn't always work. Fix:
The event is no longer queued but immediately sent to the stack.
If L1 cannot be activated, the L3 state machine that was started
by the EVENT_SETUP will do its work, i.e. a timeout will release
the B channel properly. The SETUP possibly cannot be sent the
first time but is resent by T303 in case L1 could be activated.
Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
lines chan_misdn: Remove some calls to exit(). Try proper cleanup
when something goes wrong in misdn_lib_init(). Especially do not
call exit()! * Fix memory leak because stack_destroy() does not
free the stack struct. Patches: patch06_cleanup-init.diff
(license #6372) patch uploaded by Guenther Kelleter Modified JIRA
ABE-2888 ........ Merged revisions 375519-375524 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........ Merged revisions 375625 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375626 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375627 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-02 17:27 +0000 [r375614] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP
Origin Processing While looking at some debug logs, I noticed
that it was being reported that the SDP origin line was
unsupported or failed. Upon looking into this on my local
machine, I found that I too was getting this debug message yet
everything seemed to be getting processed properly. What was
discovered is, that, the variable to determine what is displayed
in the debug message for the SDP line that was processed, was not
being set for the origin line when the result was successful.
This patch fixes this and was tested on local machine. ........
Merged revisions 375594 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375601 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375613 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-11-01 15:03 +0000 [r375576] Jonathan Rose <jrose@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: chan_sip: Fix a
bug causing SIP reloads to remove all entries from the registry A
regression was introduced in chan_sip by changes to sip reload
introduced by r349097. That patch moved peer purging from the
beginning of the reload to after the general configuration was
finished. This patch fixes that by undoing the repositioning of
the original peer purging code and using a similar function after
performing general configuration that purges only autocreated
peers that were created when persist mode isn't enabled. (closes
issue ASTERISK-20611) Reported by: Alisher Review:
https://reviewboard.asterisk.org/r/2171/ ........ Merged
revisions 375575 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-31 18:01 +0000 [r375560] Joshua Colp <jcolp@digium.com>
* res/res_http_websocket.exports.in, /: Fix an issue with
res_http_websocket where the chan_sip WebSocket handler could not
be registered. On some systems the optional API support uses the
GCC compiler attribute "weakref" to provide its functionality.
This code changes the function names and prefixes "__" to the
front. The res_http_websocket exports file did not take this into
account, thereby not allowing those functions to be global and
ultimately found. (closes issue ASTERISK-20631) Reported by:
danjenkins ........ Merged revisions 375559 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-31 14:58 +0000 [r375533] Matthew Jordan <mjordan@digium.com>
* /, res/res_calendar_ews.c: Properly extract the Body information
of an EWS calendar item Unlike all other calendar modules,
res_calendar_ews fails to extract the Body information for a
calendar item. This is due, in part, to a quirk in the schema in
the XML - not only does a CalendarItem contain a Body element,
but the CalendarItem exists as a descendant of a different Body
element. The neon parser was erroneously skipping all Body
elements. This patch fixes that by bypassing Body elements that
are not a child of CalendarItem, and parsing the Body element out
if it is a child. Note that the original patch by Terry Wilson
only needed slight modifications to make it properly pull the
Body information out; as such, while I've linked to the patch
that I uploaded for Dmitry, I've attributed the patch to Terry.
(closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested
by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff
uploaded by Terry Wilson (license 6283) ........ Merged revisions
375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 375531 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375532 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-30 19:31 +0000 [r375511] Richard Mudgett <rmudgett@digium.com>
* /, bridges/bridge_softmix.c: Fix ConfBridge crash if no timing
module loaded. (closes issue ASTERISK-19448) Reported by: feyfre
Patches: smfix.patch (license #6099) patch uploaded by feyfre
Modified for coding guidelines. ........ Merged revisions 375496
from http://svn.asterisk.org/svn/asterisk/branches/10 ........
Merged revisions 375506 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-30 19:20 +0000 [r375472-375498] Jonathan Rose <jrose@digium.com>
* /, apps/app_mixmonitor.c: mixmonitor: Add a test event This test
event is being used to fix the mixmonitor_audiohook_inherit test.
........ Merged revisions 375484 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375485 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375486 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_confbridge.c: confbridge: Fix a bug which made
conferences not record with AMI/CLI commands When confbridge was
changed to handle conference status with a state machine in
r374658. The function responsible for starting recording for a
conference was refactored with the function actually responsible
for launching the recording thread being split into a function
with another name. The old function name was still used for
manually started recordings through AMI or CLI. This patch fixes
that by switching which function is used to start recording the
conference. (closes issue ASTERISK-20601) Reported by: Vilius
Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose
(license 6182) ........ Merged revisions 375470 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375471 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-29 21:38 +0000 [r375442-375443] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Prevent resetting of NATted realtime peer
address on reload. If a "sip reload" is issued for a SIP peer,
then his IP address will be cleared, thus resulting in forgetting
the public IP address. Asterisk will then attempt to route SIP
traffic to the private IP address. The fix here is to make "sip
reload" ignore realtime peers when "host = dynamic" is spotted.
Realtime peers can now only have their IP address reset if they
have gone from being not dynamic to being dynamic. (closes issue
ASTERISK-18203) reported by daren ferreira (closes issue
ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff
uploaded by JoshE (license #6075) ........ Merged revisions
375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 375417 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375437 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_mgcp.c, main/pbx.c, apps/app_osplookup.c,
channels/chan_sip.c, channels/chan_skinny.c,
funcs/func_strings.c, UPGRADE.txt: Make evaluation of channel
variables consistently case-sensitive. Due to inconsistencies in
how variable names were evaluated, the decision was made to make
all evaluations case-sensitive. See the UPGRADE.txt file or
https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity for
more details. (closes issue ASTERISK-20163) reported by Matt
Jordan Review: https://reviewboard.asterisk.org/r/2160
2012-10-29 21:02 +0000 [r375416] Matthew Jordan <mjordan@digium.com>
* UPGRADE.txt, apps/app_queue.c: Ensure that CDRs for a caller in a
Queue that is not answered is NO ANSWER. When a caller enters a
queue and no queue member answers the call, the current behaviour
can be a little odd depending on the paused status of the queue
members. If any queue member is paused, but not all, the CDR
disposition will be BUSY. If all queue members are paused, then
the CDR disposition is based instead on the disposition of the
call prior to entering the Queue. This patch modifies the
behaviour in the following ways: * If no queue members are
paused, the CDR disposition is whatever the disposition was prior
to going into Queue. If the call was answered this will be
ANSWERED; otherwise, it is NO ANSWER. * If some queue members are
pused, the CDR result is NO ANSWER. (This is a change in
behaviour, as the result would previously have been BUSY) * If
all queue members are paused, the CDR result is whatever the
result was prior to going into Queue. This is the same as the
behaviour prior to this patch. * If the caller hangs up, times
out, or presses '*' with the 'h' option, the CDR disposition is
again not set and is dependent on whether or not the caller was
Answered prior to entering Queue. This patch was based on one
provided by Thomas Arimont, but has been modified to accomodate
findings by the reviewers. Review:
https://reviewboard.asterisk.org/r/2064/ (closes issue AST-906)
Reported by: Thomas Arimont (closes issue ASTERISK-17776)
Reported by: Attila Megyeri
2012-10-29 19:31 +0000 [r375364-375391] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Fix the Park 'r' option when a channel parks
itself. When a channel uses the Park appliation to park itself
with the 'r' option, the channel hears music-on-hold instead of
the requested ringing. * Added a missing check for the 'r' option
when a channel parks itself. (closes issue ASTERISK-19382)
Reported by: James Stocks Patches by: dsessions Review:
https://reviewboard.asterisk.org/r/2148/ ........ Merged
revisions 375388 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375389 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375390 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_dahdi.c, /: chan_dahdi: Fix segfault dereferencing
a NULL tech_pvt. The tech support customer was using the AMI
Redirect action shortly after a call was placed. While the
channel tried to do an ast_read(), the masquerade resulting from
the channel redirect took place. The masquerade in the middle of
the ast_read() resulted in the segfault. (closes issue AST-1025)
Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
(license #5621) patch uploaded by rmudgett ........ Merged
revisions 375361 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375362 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375363 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-23 16:22 +0000 [r375291-375328] Jonathan Rose <jrose@digium.com>
* contrib/scripts/ast_tls_cert, /: ast_tls_cert script: Better
response for various exit conditions to openssl (closes issue
ASTERISK-20260) Reported by: Daniel O'Connor Patches:
ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
6419) ........ Merged revisions 375325 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375326 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375327 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/app.c: core: Fix a memory leak in app.c from an early
return ast_app_group_match_get_count allocates memory with the
regcomp function and we previously forgot to free it when bailing
out due to a regex compilation failure against category. (closes
issue AST-1018) Reported by: Guenther Kelleter Patches:
regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
........ Merged revisions 375299 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375300 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375301 from
http://svn.asterisk.org/svn/asterisk/branches/11
* codecs/gsm/src/code.c, /: GSM: Fix encoding problems with GSM
(closes issue ASTERISK-20457) Reported by: Richard Miller
Patches: code.patch uploaded by Richard Miller (license 5685)
........ Merged revisions 375272 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375273 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375288 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-18 21:49 +0000 [r375240-375249] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds UPGRADE
notes describing behavioral changes to rrmemory strategy caused
by 375216 (issue AST-989) Reported by: Thomas Arimont
* /, apps/app_queue.c: app_queue: Make ordering of
rrmemory/rrordered persist over add/remove members Prior to this
patch, adding, removing or reloading members to rrmemory would
cause the order to become completely jumbled. Now it behaves more
or less like rrordered other than the fact that it stores the
members on a hash table rather than a linked list. This patch
also prevents removal of members and member reloads from jumbling
rrordered queues. (issue AST-989) Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged
revisions 375216 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375217 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375219 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-18 20:31 +0000 [r375215] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_alarmreceiver.c: Fix XML Document Validation Failure Fix
documentation error when validating the xml in trunk caused by
r375150. Moved the description end tag down to below the
variablelist element end tag. Found when compiling with
--dev-mode-enabled. (issue ASTERISK-20289)
2012-10-18 20:13 +0000 [r375192] Richard Mudgett <rmudgett@digium.com>
* makeopts.in, Makefile, /, build_tools/make_version, configure,
include/asterisk/autoconfig.h.in, configure.ac: build_tools:
Allow Asterisk to report git SHAs in version string. Make git
more attractive for managing work-in-progress. Especially
convenient when a potential patch set needs to be tested on
multiple platforms since one can use git to keep all the test
environments in sync independent of a subversion server. Now the
Asterisk version will show the exact git SHA5 that was used when
building (still appended by "M" if there are local modifications)
from a git clone of the Asterisk repository so the developer can
more easily know what is actually under test. You will now get
this: $ asterisk -V Asterisk GIT-1698298 Instead of this: $
asterisk -V Asterisk UNKNOWN__and_probably_unsupported This has
zero impact for those not using git with the exception of an
extra test in the configure script to gather git's path. This is
necessary to prevent "sudo make install" from failing since git
may not be in the path in make's shell environment. (closes issue
ASTERISK-20483) Reported by: Shaun Ruffell Patches:
0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
(license #5417) patch uploaded by Shaun Ruffell Modified ........
Merged revisions 375189 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375190 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375191 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-18 14:17 +0000 [r375182] Andrew Latham <lathama@gmail.com>
* main/features.c, include/asterisk/module.h,
include/asterisk/doxygen/reviewboard.h, main/logger.c,
main/http.c, include/asterisk/doxygen/licensing.h, main/dsp.c,
main/udptl.c, main/dnsmgr.c, contrib/asterisk-ng-doxygen,
Makefile.rules, codecs/log2comp.h, main/cli.c, main/cdr.c,
include/asterisk/doxyref.h,
include/asterisk/doxygen/asterisk-git-howto.h, main/manager.c,
main/app.c, pbx/pbx_dundi.c, include/asterisk/doxygen/commits.h,
include/asterisk/udptl.h, include/asterisk/smdi.h,
main/asterisk.c, include/asterisk/doxygen/architecture.h,
include/asterisk.h, main/ccss.c, Makefile.moddir_rules,
main/cel.c, main/named_acl.c, main/enum.c, Makefile,
include/asterisk/paths.h, include/asterisk/doxygen/releases.h,
include/asterisk/compat.h: Doxygen Updates - Title update Update
and extend the configuration_file group and enable linking.
Commit other cleanups from multi-version Doxygen testing. Update
title that was left behind many years ago. (issue ASTERISK-20259)
2012-10-17 20:34 +0000 [r375175] Jonathan Rose <jrose@digium.com>
* main/manager.c: manager: remove curses dependent stuff from
r375103 Upon further examination, this code was causing
compliation problems on CentOS at the least (possibly on any
machine without curses) and also the local value of COLS is used
even with a remote console, so it is less than ideal. (issue
ASTERISK-20396) Reported by: Johan Wilfer
2012-10-17 19:02 +0000 [r375150] Pedro Kiefer <pedro@kiefer.com.br>
* apps/app_alarmreceiver.c, configs/alarmreceiver.conf.sample: Adds
new formats to app_alarmreceiver, ALAW calls support and enhanced
protection. Commiting this on behalf of Kaloyan Kovachev (license
5506). AlarmReceiver now supports the following DTMF signaling
types: - ContactId - 4x1 - 4x2 - High Speed - Super Fast We are
also auto-detecting which signaling is being received. So support
for those protocols should work out-the-box. Correctly identify
ALAW / ULAW calls. Some enhanced protection for broken panels and
malicious callers where added. (closes issue ASTERISK-20289)
Reported by: Kaloyan Kovachev Review:
https://reviewboard.asterisk.org/r/2088/
2012-10-17 19:01 +0000 [r375149] Kinsey Moore <kmoore@digium.com>
* main/tcptls.c, /: Ensure Asterisk fails TCP/TLS SIP calls when
certificate checking fails When placing a call to a TCP/TLS SIP
endpoint whose certificate is not signed by a configured CA
certificate, Asterisk would issue a warning and continue to
process the call as if there was not an issue with the
certificate. Asterisk now properly fails the call if the
certificate fails verification or if the certificate does not
exist when certificate checking is enabled (the default
behavior). (closes issue ASTERISK-20559) Reported by: kmoore
Review: https://reviewboard.asterisk.org/r/2163/ ........ Merged
revisions 375146 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375147 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375148 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-17 14:24 +0000 [r375110-375137] Walter Doekes <walter+asterisk@wjd.nu>
* res/res_rtp_asterisk.c, main/pbx.c, channels/chan_sip.c,
cdr/cdr_odbc.c: Change a few warnings to debug and the inverse.
Remove the "RTP Read too short" warning for RTP keepalives.
Remove the the warning about the application delimiter switch
from pipe to comma. (You should've done this by now.) Make
cdr_odbc report more when an insert fails. Make chan_sip warn
less when the peer wants SRTP (and we don't) or sends a zero port
to disable a media type. Review:
https://reviewboard.asterisk.org/r/2167 (closes issue
ASTERISK-20538)
* /, channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads.
Don't crash on large user input. Allow SIP headers without space.
Optimize code a bit. Review:
https://reviewboard.asterisk.org/r/2162 ........ Merged revisions
375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 375112 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375113 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_sip.c: Don't do SIP contact/route DNS if we're not
using the result. In many cases (for peers behind NAT or for TCP
sockets) we do not need to look up any hostname in the Contact
(or Route) when sending an in-dialog request. This should reduce
netsock2.c: getaddrinfo errors in certain scenarios. Review:
https://reviewboard.asterisk.org/r/2156
2012-10-16 20:45 +0000 [r375103] Jonathan Rose <jrose@digium.com>
* main/manager.c, CHANGES: manager: Change display of 'manager show
commands' and 'manager show command' manager show commands now
shows the full name of the command being displayed regardless of
size. The privilege column has also been removed from this
display. It will also now use the full length of the terminal if
curses is available. Manager show command will now always display
the privilege of the manager command within the CLI. (closes
ASTERISK-20396) Reported by: Johan Wilfer Review:
https://reviewboard.asterisk.org/r/2143/
2012-10-16 19:26 +0000 [r375081] Pedro Kiefer <pedro@kiefer.com.br>
* apps/app_alarmreceiver.c: Fixes two small regressions from
ASTERISK-20157 - receive_dtmf_digits had the wrong buffer length
- app_alarmreceiver should wait 100ms before sending the second
part of handshake (closes issue ASTERISK-20484) Reported by:
Jean-Philippe Lord Tested by: Jean-Philippe Lord, Pedro Kiefer
Patches: ASTERISK-20484_v2.diff uploaded by Kaloyan Kovachev
(license 5506)
2012-10-16 19:25 +0000 [r375080] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: Update sip_request_call SIP dial string
documentation. This was missed when merging review r1859.
........ Merged revisions 375074 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375078 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375079 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-16 14:09 +0000 [r375052] Joshua Colp <jcolp@digium.com>
* /, channels/chan_iax2.c: Remove a log message that was left in
accidentally from call-id logging development. ........ Merged
revisions 375051 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-15 21:25 +0000 [r375044] Mark Michelson <mmichelson@digium.com>
* include/asterisk/strings.h, channels/chan_iax2.c,
apps/app_dial.c, /, main/ccss.c: Fix some potential misuses of
ast_str in the code. Passing an ast_str pointer by value that
then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally passed
by value being invalidated if the ast_str had to be reallocated.
This fixes places in the code that do this. Only the example in
ccss.c could result in pointer invalidation though since the
other cases use a stack-allocated ast_str and cannot be
reallocated. I've also updated the doxygen in strings.h to
include notes about potential misuse of the functions mentioned
previously. Review: https://reviewboard.asterisk.org/r/2161
........ Merged revisions 375025 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 375026 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 375027 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-15 08:26 +0000 [r375017] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: Fix underscreen buttons warnings
apeared while transfer process ........ Merged revisions 375016
from http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-14 21:59 +0000 [r375003-375009] Andrew Latham <lathama@gmail.com>
* addons/chan_mobile.c, addons/app_mysql.c: Doxygen Updates Update
and extend the configuration_file group and enable linking.
(issue ASTERISK-20259)
* utils/extconf.c, utils/muted.c: Doxygen Updates Update and extend
the configuration_file group and enable linking. (issue
ASTERISK-20259)
* addons/Makefile, pbx/Makefile, formats/Makefile, sounds/Makefile,
funcs/Makefile, bridges/Makefile, agi/Makefile, codecs/Makefile,
utils/Makefile, tests/Makefile, cel/Makefile, main/Makefile:
Title update Update title that was left behind many years ago.
Used revision 6596 as my guide for what it should be. (issue
ASTERISK-20259)
* channels/chan_gtalk.c, channels/chan_console.c,
channels/Makefile, channels/chan_iax2.c, channels/chan_oss.c,
channels/chan_jingle.c, channels/chan_phone.c,
channels/chan_dahdi.c, channels/iax2-parser.h,
channels/chan_misdn.c, channels/chan_skinny.c,
channels/chan_motif.c, channels/chan_h323.c, channels/iax2.h,
channels/chan_alsa.c, channels/chan_mgcp.c, channels/chan_vpb.cc,
channels/chan_sip.c: Doxygen Updates - Title update Update and
extend the configuration_file group and enable linking. Update
title that was left behind many years ago. (issue ASTERISK-20259)
* cdr/Makefile, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c,
cdr/cdr_odbc.c, cdr/cdr_radius.c, cdr/cdr_custom.c,
cdr/cdr_manager.c, cdr/cdr_csv.c, cdr/cdr_syslog.c: Doxygen
Updates - Title update Update and extend the configuration_file
group and enable linking. Update title that was left behind many
years ago. (issue ASTERISK-20259)
* apps/Makefile, apps/app_meetme.c, apps/app_festival.c,
apps/app_fax.c, apps/app_skel.c, apps/app_alarmreceiver.c,
apps/app_amd.c, apps/app_confbridge.c, apps/app_followme.c,
apps/app_queue.c, apps/app_adsiprog.c, apps/app_voicemail.c:
Doxygen Updates - Title update Update and extend the
configuration_file group and enable linking to the application.
Update title that was left behind many years ago. (issue
ASTERISK-20259)
* res/res_jabber.c, res/res_config_sqlite.c, res/res_smdi.c,
res/res_curl.c, res/res_config_ldap.c, res/res_odbc.c,
res/res_clialiases.c, res/res_calendar.c,
res/res_config_sqlite3.c, res/res_config_pgsql.c, res/res_snmp.c,
res/res_limit.c, res/res_fax.c, res/res_phoneprov.c,
res/Makefile, res/res_xmpp.c, res/res_musiconhold.c: Doxygen
Updates - Title update Update and extend the configuration_file
group and enable linking to the resource. Update title that was
left behind many years ago. (issue ASTERISK-20259)
2012-10-14 12:23 +0000 [r374996] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, config.guess, config.sub: Update config.guess and config.sub:
2012-10-10 Update config.guess and config.sub to revision
fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
64bit). config.guess:timestamp='2012-09-25'
config.sub:timestamp='2012-10-10' ........ Merged revisions
374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 374991 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374995 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-13 19:58 +0000 [r374940-374970] Andrew Latham <lathama@gmail.com>
* CREDITS: Update CREDITS Update Jean-Denis and add myself (issue
ASTERISK-20259)
* Makefile: Multiplatform Makefile Update Paul Belanger pointed out
that using sed in the Makefile is an issue with multiple
platforms. We are cleaning up the Doxygen config as a following
step so I just switched the sed inplace changes to be an echo
append instead. (issue ASTERISK-20259)
* main/app.c, apps/app_dial.c: Doxygen Clean ups Add app_skel.c as
an example in app.c and fix some formating for the "Dial Privacy
scripts" so it actually shows up in the Doxygen output. (issue
ASTERISK-20259)
* Makefile: Test for Asterisk Version info Doxygen uses the
ASTERISKVERSION as a sub header. If a SVN export is done and no
.svn or .version file exists it defualts to
UNKNOWN__and_probably_unsupported which is honest but not great
for the online docs. During the "make progdocs" I added a test
for this and just warned and ommitted the version. (issue
ASTERISK-20259)
* contrib/asterisk-ng-doxygen: Correct output directory During
testing I used an alternate output directory and mistakenly
committed it. Matt Jordan noticed and I reverted. This is the
correct setting for local output to match with all branches.
(issue ASTERISK-20259)
* static-http/ajamdemo.html, static-http/astman.css: Add
licens/copyright header Begin update of static-http files and
general clean ups. This only adds the standard header to the
files. (issue ASTERISK-20503)
* configure, configure.ac, makeopts.in, Makefile: Add check for
Doxygen The autoconf configuration system had a test for DOT but
not for Doxygen. I added the test for Doxygen and did an overhaul
of the Makefile check to a much simpler process. (issue
ASTERISK-20259)
2012-10-12 21:58 +0000 [r374933] Kinsey Moore <kmoore@digium.com>
* /, apps/app_voicemail.c: Avoid a segfault on invalid format names
If a format name was not found by ast_getformatbyname, a NULL
pointer would be passed into ast_format_rate and immediately
dereferenced. This ensures that a valid pointer is used since the
structure is already allocated on the stack. (closes issue
DPH-523) Reported-by: Steve Pitts ........ Merged revisions
374932 from http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-12 16:31 +0000 [r374924] Mark Michelson <mmichelson@digium.com>
* main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
Do not use a FILE handle when doing SIP TCP reads. This is used
to solve an issue where a poll on a file descriptor does not
necessarily correspond to the readiness of a FILE handle to be
read. This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead. Because TCP does not
guarantee that an entire message or even just one single message
will arrive during a read, a loop has been introduced to ensure
that we only attempt to handle a single message at a time. The
tcptls_session_instance structure has also had an overflow buffer
added to it so that if more than one TCP message arrives in one
go, there is a place to throw the excess. Huge thanks goes out to
Walter Doekes for doing extensive review on this change and
finding edge cases where code could fail. (closes issue
ASTERISK-20212) reported by Phil Ciccone Review:
https://reviewboard.asterisk.org/r/2123 ........ Merged revisions
374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 374906 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374914 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-11 23:40 +0000 [r374879-374897] Andrew Latham <lathama@gmail.com>
* contrib/scripts/install_prereq: Append Doxygen to Debian packages
list Add Doxygen to the Debian install list. I will check for
other platforms like Red Hat (issue ASTERISK-20259)
* static-http/mantest.html: Update JQuery URL to recent version The
JQuery URL to version 1.4 will be removed within the life span of
Asterisk 11. This is a compatible upgrade by using the URL for
1.8. (issue ASTERISK-20503)
* main/manager.c, include/asterisk/module.h: Continue to group
config files (issue ASTERISK-20259)
* CREDITS: CREDITS clean up As discussed online
http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html
the credits file needs some cleaning. This is 95% whitespace with
a few additions found in file headers. Further additions should
be added here instead of in the file being updated. (issue
ASTERISK-20259)
* contrib/asterisk-ng-doxygen: Revert Local testing Config Revert a
local testing config that I made. This was not intended to be
committed. Thank you Matt Jordan for noticing this. (issue
ASTERISK-20259)
2012-10-11 21:19 +0000 [r374852-374878] Joshua Colp <jcolp@digium.com>
* /, channels/chan_motif.c: Fix a bug where audio on Google Voice
would not work due to ignoring candidates. Instead of ignoring
parts of the message that are not known just ignore the ones we
know may be present and that would cause a problem. ........
Merged revisions 374877 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_motif.c: Fix an issue where outgoing calls would
fail to establish audio due to ICE negotiation failures. This
change removes the requirement for ufrag and pwd in the transport
stanza and also makes us the controlling agent. (closes issue
ASTERISK-20554) Reported by: mmichelson ........ Merged revisions
374850 from http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-11 15:49 +0000 [r374849] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.exports.in (removed), main/sip_api.c (added),
/, channels/chan_sip.c, include/asterisk/sip_api.h: Don't make
chan_sip export global symbols. During testing, it was discovered
that having chan_sip export global symbols was problematic. The
biggest problem was that load order was affected. Trying to use
realtime could be problematic since in all likelihood the
necessary realtime driver(s) would not be loaded before chan_sip.
In addition, it was found that it was impossible to use the
Digium Phone Module for Asterisk since it must be loaded before
chan_sip since it must hook into chan_sip's configuration
parsing. The solution is to use a virtual table in the same
manner that other modules in Asterisk do, like app_voicemail.
(closes issue ASTERISK-20545) Reported by: kmoore ........ Merged
revisions 374842 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-11 15:44 +0000 [r374846] Matthew Jordan <mjordan@digium.com>
* main/cdr.c, /: Fix incorrect billing duration reported when batch
mode is enabled Similar to r369351, the billing duration can be
skewed when batch mode is enabled. This happened much more rarely
than the duration, as it only occured when the call was answered
(thereby indicating an actual answer time) and immediately hung
up on (indicating a billsec of 0). Since a billing time of '0'
can either mean that the call immediately ended or that the CDR
was improperly answered, we have to use additional information to
know whether or not we can trust the CDR billsec value. Prior to
this patch, we looked to see if we had a valid answer time. If we
did, and billsec was zero, we used the current time to calculate
what billsec value we could from the CDR being written. If batch
mode is enabled, this will incorrectly report a billsec value
being much greater than the actual duration of the call. Instead
of relying on the presence of an answer time to know whether or
not we can re-calculate the billsec for the CDR, we now also use
the presence of the CDR's end time to know if we need to
re-calculate or whether we can trust the billsec value that we
have. This prevents erroneous jumps in the billsec value, while
still making sure that in the worst case, some billing time will
be calculated. (closes issue AST-1016) Reported by: Thomas
Arimont Tested by: Thomas Arimont ........ Merged revisions
374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 374844 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374845 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-11 13:34 +0000 [r374834] Joshua Colp <jcolp@digium.com>
* /, channels/chan_motif.c: Consider the Google Talk content stanza
name (jin:content) valid. ........ Merged revisions 374833 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-10 21:05 +0000 [r374805] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, /: app_queue: Made pass connected line updates
from the caller to ringing queue members. Party A calls Party B
Party B puts Party A on hold. Party B calls a queue. Ringing
queue member D sees Party B identification. Party B transfers
Party A to the queue. Queue member D does not get a connected
line update for Party A. Queue member D answers the call and
still sees Party B information. However, if Party A later
transfers the call to Party C then queue member D gets a
connected line update for Party C. * Made pass connected line
updates from the caller to queue members while the queue members
are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
(closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
rmudgett ........ Merged revisions 374801 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........ Merged revisions 374802 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374803 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374804 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-10 13:40 +0000 [r374793] Kinsey Moore <kmoore@digium.com>
* main/manager.c, /: Fix segfault regression from r370681 Due to
usage of ast_hook_send_action, AMI action handling code should be
able to handle a NULL mansession->session. This would cause a
crash on NULL dereference if action_originate was called from
ast_hook_send_action. (closes issue ASTERISK-20544) ........
Merged revisions 374792 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-09 22:24 +0000 [r374778] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c, /: Fix execution of 'i' extension due to
uninitialized variable. The fix for ASTERISK-18243 added code
that could potentially use dst_exten[] uninitialized. As a result
the 'i' exten may not be executed when it should. (closes issue
ASTERISK-20455) Reported by: Richard Miller Patches:
pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard
Miller Made some cosmetic modifications. ........ Merged
revisions 374758 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374763 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374771 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-09 21:35 +0000 [r374757] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Improve logging for DTLS-SRTP failure
situations. (closes issue ASTERISK-20487) Reported by: mjordan
........ Merged revisions 374756 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-08 22:31 +0000 [r374717-374730] Richard Mudgett <rmudgett@digium.com>
* configs/chan_dahdi.conf.sample, /: dahdi.conf.sample: Add
description for "buffers" setting. This contains an edited
version of the patch originally created by John Bigelow. (closes
issue ASTERISK-14435) Reported by: John Bigelow Patches:
buffers.patch (license #5091) patch uploaded by John Bigelow
0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch
(license #5417) patch uploaded by Shaun Ruffell Modified ........
Merged revisions 374727 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374728 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374729 from
http://svn.asterisk.org/svn/asterisk/branches/11
* pbx/pbx_spool.c, /: Fix deletion of unopenable spool files. If
scan_service() cannot open the spool file, it logs a message
saying that it will delete the file and calls remove_from_queue()
to do it. However, remove_from_queue() fails to delete the spool
file because struct outgoing has not yet been fully initialized.
* Merged allocating a new struct outgoing and init_outgoing()
into new_outgoing(). Allocation is initialization. * Made
apply_outgoing() not initialize the spool filename in struct
outgoing. * Made apply_outgoing() call ast_trim_blanks() and
ast_skip_blanks() rather than manually inlining them. * Reduced
indentation levels in apply_outgoing(). * Fixed a garbled comment
in remove_from_queue(). * Reworked scan_service() to simplify it.
(closes issue ASTERISK-17231) Reported by: David Chappell
Patches: spool_open_failure.diff (license #4997) patch uploaded
by David Chappell Started with this patch. ........ Merged
revisions 374686 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some
memory leaks on off nominal paths in init_outgoing() when merging
into the new_outgoing() function dealing with o->capabilities.
........ Merged revisions 374695 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374708 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-08 20:39 +0000 [r374633-374677] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c, /, configs/rtp.conf.sample: Disable ICE
support by default Since there are a number of legacy devices out
there that fail to handle ICE candidates properly (which is a
nice way of saying something much uglier), disable it by default.
Support for ICE candidates can be enabled in rtp.conf using the
icesupport setting. ........ Merged revisions 374676 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/confbridge/conf_state_multi.c (added),
apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c
(added), apps/confbridge/conf_state_empty.c (added),
apps/confbridge/conf_state.c (added),
apps/confbridge/conf_state_single.c (added),
apps/confbridge/conf_state_inactive.c (added),
apps/confbridge/conf_state_single_marked.c (added), /,
apps/confbridge/include/confbridge.h,
apps/confbridge/include/conf_state.h (added): Resolve issues in
ConfBridge regarding marked, waitmarked, and unmarked users
Thank's to Neil Tallim (flan)'s tireless testing, issue
reporting, and patches it became clear that app_confbridge had
some complex logic in how it handled interactions between marked,
waitmarked, and unmarked users. In particular, there were some
areas in which the interactions between the users resulted in
inconsistent behavior, and app_confbridge was missing logic in
how to handle some corner cases. Some areas included: * Poor
handling of mixing unmarked and waitmarked users *
Inconsistencies in how MOH and muting was applied to various
users * Handling of various announcements for different user
profile options flan's patches seem to fix the various issues,
but highlighted how hard the code could be to maintain. In an
attempt to make things easier to maintain and to more fully
enumerate the various cases that exist, this patch breaks up the
logic into a state machine-like setup. Please note that the
various state transitioned are documented on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes
Review: //https://reviewboard.asterisk.org/r/2072/ Note that for
the following issues, mjordan uploaded the patch, although it was
written by twilson. Any contributor license discrepency is due to
that. (closes issue ASTERISK-19562) Reported by: flan Tested by:
flan, mjordan, jrose patches:
bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
twilson (license 6283) (closes issue ASTERISK-19726) Reported by:
flan Tested by: flan patches:
bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
twilson (license 6283) (closes issue ASTERISK-20181) Reported by:
Jonathan White Tested by: Jonathan White patches:
bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
twilson (license 6283) ........ Merged revisions 374652 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374657 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/pjproject/pjlib/src/pj/sock_linux_kernel.c,
res/pjproject/pjlib/include/pj/sock.h,
res/pjproject/pjlib/src/pj/sock_symbian.cpp, /,
res/pjproject/pjlib/src/pj/sock_bsd.c: pjproject: Fix for Solaris
builds. Do not undef s_addr. pjproject, in order to solve build
problems on Windows [1], undefines s_addr in one of it's headers
that is included in res_rtp_asterisk.c. On Solaris s_addr is not
a structure member, but defined to map to the real strucuture
member, therefore when building on Solaris it's possible to get
build errors like: [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o
In file included from
/export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
from res_rtp_asterisk.c:51:
/export/home/admin/asterisk-11-svn/include/asterisk/network.h: In
function `inaddrcmp':
/export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
error: structure has no member named `s_addr'
/export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
error: structure has no member named `s_addr' res_rtp_asterisk.c:
In function `ast_rtp_on_ice_tx_pkt': res_rtp_asterisk.c:706:
warning: dereferencing type-punned pointer will break
strict-aliasing rules res_rtp_asterisk.c:710: warning:
dereferencing type-punned pointer will break strict-aliasing
rules res_rtp_asterisk.c: In function
`rtp_add_candidates_to_ice': res_rtp_asterisk.c:1085: error:
structure has no member named `s_addr' make[2]: ***
[res_rtp_asterisk.o] Error 1 make[1]: *** [res] Error 2 make[1]:
Leaving directory `/export/home/admin/asterisk-11-svn' gmake: ***
[_cleantest_all] Error 2 Unfortunately, in order to make this
work, I also had to make sure pjproject only used the typdef
pj_in_addr and not the struct pj_in_addr so that when building
Asterisk I could "typedef struct in_addr pj_in_addr". It's
possible then that the library and users of those interfaces in
Asterisk have a different idea about the type of the argument,
while on the surface it looks like they are all 32 bit big endian
values. [1] http://trac.pjsip.org/repos/changeset/484 (issues
ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang,
mjordan patches:
0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch
uploaded by Shaun Ruffell (license 5417) ........ Merged
revisions 374642 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/acl.c, /: Trivial patch to make 'best_score' defined for all
architectures. Fixes trivial build error on Solaris: acl.c: In
function `get_local_address': acl.c:196: error: `best_score'
undeclared (first use in this function) acl.c:196: error: (Each
undeclared identifier is reported only once acl.c:196: error: for
each function it appears in.) make[2]: *** [acl.o] Error 1 (issue
ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang
patches:
0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch
by Shaun Ruffell (license 5417) ........ Merged revisions 374632
from http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-06 03:22 +0000 [r374612-374623] Matthew Jordan <mjordan@digium.com>
* /, res/res_xmpp.c: Handle capability stanzas that fail to provide
node or version information While XEP-0115 states that the node
and ver attributes are both required, some devices fail to
provide either field. Prior to this patch, failure to provide the
node or ver attribute would cause a crash in res_xmpp. While
failing to provide the node or ver attribute is technically
invalid, since this information is not utilized by Asterisk
except for reporting purposes, for interoperability reasons, we
continue to process the capability stanza anyways. (closes issue
ASTERISK-20495) Reported by: Martin W Tested by: Martin W
patches: 20495.patch uploaded by Martin W (license #6434)
........ Merged revisions 374622 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_xmpp.c, main/message.c, /: Update documentation for
MessageSend application/command's From field for XMPP When using
the channel technology agnostic application/AMI command
MessageSend, the "From" field is technically optional for the SIP
channel driver. However, if being sent by the XMPP resource
module (either res_xmpp or res_jabber), the "From" field is
necessary, and must correspond to a defined account. This patch
updates the documentation for this application/AMI command to
reflect this. (closes issue ASTERISK-20405) Reported by: Leif
Madsen ........ Merged revisions 374611 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-05 20:33 +0000 [r374588] David M. Lee <dlee@digium.com>
* main/manager.c, /: Multiple revisions 374570,374581 ........
r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) |
22 lines Improve AMI long line error handling In AMI's parser,
when it receives a long line (> 1024 characters), it discards
that line, but continues to process the message normally.
Typically, this is not a problem because a) who has lines that
long and b) usually a discarded line results in an invalid
message. But if that line is specifying an optional field, then
the message will be processed, you get a 'Response: Success', but
things don't work the way you expected them to. This patch
changes the behavior when a line-too-long parse error occurs. *
Changes the log message to avoid way-too-long (and truncated
anyways) log messages * Adds a 'parsing' status flag to Response:
Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line
is too long * Responds with an appropriate error if parsing !=
MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581
| dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
I've committed too much. Reverting part of r374570. ........
Merged revisions 374570,374581 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374586 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374587 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-05 18:42 +0000 [r374539] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c,
channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Merged
revisions 374515-374535 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
(Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
Made setup_bc() static. Patches: patch1_unused-code.diff (license
#6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
(Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
states Patches: patch2_unused-states.diff (license #6372) patch
uploaded by Guenther Kelleter JIRA ABE-2882 ................
r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
| 16 lines chan_misdn: Remove unnecessary null pointer checks and
checks for stack->nt * cleanup_bc() is always called with valid
bc (or it would've crashed before). * Value of stack->nt is known
in advance at some places. * Rename handle_event() to
handle_event_te(), handle_frm() to handle_frm_te(). Patches:
patch3_checks.diff (license #6372) patch uploaded by Guenther
Kelleter Modified JIRA ABE-2882 ................ r374518 |
rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Fix spelling in log messages Patches:
patch4_spelling.diff (license #6372) patch uploaded by Guenther
Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
emptied, cleaned and set not in use, although
misdn_lib_send_event() already did the same. This is bad. When
it's not in use we are not allowed to touch it. * Moved log
message in front of the resulting actions and fixed it to match
the case. Patches: patch5_bccleanup.diff (license #6372) patch
uploaded by Guenther Kelleter JIRA ABE-2882 ................
r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
| 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
etc., really bad stuff. * Fix return codes of cb_events() for
EVENT_SETUP to use caller's cleanup mechanisms. * Move
cl_queue_chan() call after bearer check. Patches:
patch6_leaks.diff (license #6372) patch uploaded by Guenther
Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
chan_misdn: We must initialize cause on sending a DISCONNECT. We
must initialize cause on sending a DISCONNECT, so it is later
correctly indicated to ast_channel in case the answer
(RELEASE/RELEASE_COMPLETE) does not include one. Patches:
patch7_hangupcause.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2882 ................ r374522 |
rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Remove unused code for upqueue Patches:
patch8_unused-upqueue.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2882 ................ r374523 |
rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Improve debugging (port number, messages fixed, dups
removed) Patches: patch9_debug.diff (license #6372) patch
uploaded by Guenther Kelleter JIRA ABE-2882 ................
r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
| 8 lines chan_misdn: Better debug: we can print_bc_info even if
there's no ast leg. Patches: patch10_debug-bc-2.diff (license
#6372) patch uploaded by Guenther Kelleter Modified. JIRA
ABE-2882 ................ r374534 | rmudgett | 2012-10-05
12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
setup_bc() is called too early for an incoming SETUP on TE. This
prevents the B channel from being setup for HDLC mode when
requested by the bearer capability and config option hdlc=yes. It
violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
connect to the channel until a CONNECT ACKNOWLEDGE message has
been received." * Call setup_bc() on receipt of
CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
Guenther Kelleter Modified. JIRA ABE-2881 ................
r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
| 2 lines chan_misdn: Remove some more deadcode. ................
........ Merged revisions 374536 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374537 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374538 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-04 20:21 +0000 [r374478-374493] Alec L Davis <sivad.a@paradise.net.nz>
* /, configs/dsp.conf.sample, CHANGES, main/dsp.c: dsp.c User
Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
a recompile, allow values to be adjusted in dsp.conf For binary
distributions allows easy adjustment for wobbly GSM calls, and
other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Reported by:
alecdavis Tested by: alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2144/ ........ Merged
revisions 374479 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374481 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374485 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/dsp.c, /: dsp.c fix incorrect DTMF Digit_Duration. it's
always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if
hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2145/ ........ Merged
revisions 374475 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374476 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374477 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-04 15:48 +0000 [r374429] David M. Lee <dlee@digium.com>
* /, res/res_agi.c, main/db.c: Fix DBDelTree error codes for AMI,
CLI and AGI The AMI DBDelTree command will return Success/Key
tree deleted successfully even if the given key does not exist.
The CLI command 'database deltree' had a similar problem, but was
saved because it actually responded with '0 database entries
removed'. AGI had a slightly different error, where it would
return success if the database was unavailable. This came from
confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted
(including 0 for deleting nothing). * Changed some poorly named
res variables to num_deleted * Specified specific errors when
calling ast_db_deltree (database unavailable vs. entry not found
vs. success) * Fixed similar bug in AGI database deltree, where
'Database unavailable' results in successful result (closes issue
AST-967) Reported by: John Bigelow Review:
https://reviewboard.asterisk.org/r/2138/ ........ Merged
revisions 374426 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374427 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374428 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-04 13:49 +0000 [r374414] Joshua Colp <jcolp@digium.com>
* include/asterisk/rtp_engine.h, main/rtp_engine.c,
channels/chan_sip.c: Add support for applying direct media ACLs
between differing channel technologies. Review:
https://reviewboard.asterisk.org/r/2122/
2012-10-04 04:50 +0000 [r374387] Alec L Davis <sivad.a@paradise.net.nz>
* CHANGES, main/dsp.c, /, configs/dsp.conf.sample: dsp.c User
configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
Asterisk's DTMF Specifications are based on AT&T specs, which may
not be compatible in other countries. Various countries have
different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies. Power
level difference between frequencies for different
Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
= Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
(2006-03) Now allow 4 variables to be individually configured in
dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
specifications Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
(closes issue ASTERISK-20442) Reported by: tbsky Tested by:
tbsky,alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2141/ ........ Merged
revisions 374384 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374385 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374386 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-04 02:16 +0000 [r374302-374338] Matthew Jordan <mjordan@digium.com>
* /, res/res_jabber.c: Check for presence of buddy in info/dinfo
handlers The res_jabber resource module uses the ASTOBJ library
for managing its ref counted objects. After calling
ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
the object has to be checked to see if the buddy existed. Prior
to this patch, the buddy object was not checked for NULL; with
this patch in both aji_client_info_handler and aji_dinfo_handler
the pointer is checked before used and, if no buddy object was
found, the handlers return an error code. This patch does not
take the approach that our JID can be used to log in from another
resource. If that approach is desired, an improvement could be
made to this patch to create the buddy on the fly. This patch
seeks only to prevent Asterisk from crashing. FYI: In Asterisk
11+, you really should be using res_xmpp. It does not have this
problem, as it moved to the astobj2 library. Note that multiple
people have proposed patches for this issue; the patch being
committed here is based on those. (closes issue ASTERISK-19532)
Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
fix-jabber uploaded by Karsten Wemheuer (license #5930)
xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
(license #6157) (closes issue ASTERISK-19557) Reported by:
ulugutz ........ Merged revisions 374335 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374336 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374337 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/ccss.c: Destroy the generic_monitors container after the
core_instances in ccss For each item in core_instances disposed
of in the shutdown of ccss, any generic monitor instances
referenced by the objects will be removed from generic_monitors
during their destruction. Hilarity ensues if generic_monitors no
longer exists. Thanks to the Asterisk Test Suite's generic_ccss
test for complaining loudly when it ran into this. ........
Merged revisions 374300 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374301 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-02 23:23 +0000 [r374269-374279] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c: Missed an astobj2.c debug tag.
* main/astobj2.c: * Add ref debug tags to astobj2.c ref usage. *
Make container nodes not show up in the ref debug log.
2012-10-02 21:26 +0000 [r374197-374259] Matthew Jordan <mjordan@digium.com>
* main/asterisk.c, /: Ensure Shutdown AMI event is still fired
during Asterisk shutdown Richard pointed out that having the
manager dispose of itself gracefully during shutdown meant that
the Shutdown event will no longer get fired. This patch moves the
AMI event just prior to running the atexit callbacks. ........
Merged revisions 374230 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374231 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374248 from
http://svn.asterisk.org/svn/asterisk/branches/11
* utils/hashtest2.c: Modify hashtest2 to compile after r374213.
Someone, somewhere, may care. Because hashtest2 has to provide
symbols for things in asterisk that items it includes may use,
when astobj2 decided to use ast_register_atexit it needed to
provide a declaration for that as well. Otherwise - no linky. On
a related note, ASTERISK-20505 was filed to convert
hashtest/hashtest2 into actual unit tests, so we don't run into
this problem again.
* main/astobj2.c, main/message.c, /: Fix findings from check-in on
r374177 Richard pointed out two problems with the check-in from
r374177: * The ast_msg_shutdown function declaration doesn't
match the prototype in main/message.c. * The ref/alloc function
usage in astobj2 (in trunk) can use the ao2_t_* variants of the
functions to allow the REF_DEBUG flag to enable/disable their
debug counterparts. ........ Merged revisions 374210 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374211 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/features.c, main/config_options.c, main/event.c,
main/message.c, main/asterisk.c, main/db.c, main/xmldoc.c,
main/format.c, main/udptl.c, main/pbx.c, /, main/ccss.c,
include/asterisk/astobj2.h, channels/chan_agent.c,
res/res_xmpp.c, main/taskprocessor.c, res/res_musiconhold.c,
main/named_acl.c, main/cel.c, main/astobj2.c, main/format_pref.c,
main/indications.c, main/channel.c, main/data.c, main/manager.c:
Fix a variety of ref counting issues This patch resolves a number
of ref leaks that occur primarily on Asterisk shutdown. It adds a
variety of shutdown routines to core portions of Asterisk such
that they can reclaim resources allocate duringd initialization.
Review: https://reviewboard.asterisk.org/r/2137 ........ Merged
revisions 374177 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374178 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374196 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-01 23:39 +0000 [r374164-374167] Andrew Latham <lathama@gmail.com>
* main/asterisk.c, addons/app_mysql.c, include/asterisk/doxyref.h,
contrib/asterisk-ng-doxygen, main/http.c: Doxygen Cleanup Start
adding configuration file linking and pages. Add module loading
doxygen block. Breaking up commits to keep it easy to track
(issue ASTERISK-20259)
* channels/chan_motif.c, channels/chan_alsa.c,
channels/chan_console.c, channels/chan_gtalk.c,
channels/chan_iax2.c, channels/chan_oss.c, channels/chan_mgcp.c,
channels/chan_jingle.c, channels/chan_dahdi.c,
channels/chan_misdn.c, channels/chan_vpb.cc, channels/chan_sip.c,
channels/chan_skinny.c: Doxygen Cleanup Start adding
configuration file linking and pages. Add module loading doxygen
block. Breaking up commits to keep it easy to track (issue
ASTERISK-20259)
* res/res_calendar.c, res/res_clialiases.c,
res/res_config_sqlite3.c, res/res_smdi.c, res/res_snmp.c,
res/res_fax.c, res/res_phoneprov.c, res/res_musiconhold.c,
res/res_xmpp.c, res/res_config_ldap.c, res/res_curl.c,
res/res_config_sqlite.c, res/res_timing_kqueue.c, res/res_odbc.c:
Doxygen Cleanup Start adding configuration file linking and
pages. Add module loading doxygen block. Breaking up commits to
keep it easy to track (issue ASTERISK-20259)
* apps/app_alarmreceiver.c, apps/app_amd.c, apps/app_confbridge.c,
apps/app_followme.c, apps/app_queue.c, apps/app_adsiprog.c,
apps/app_voicemail.c, apps/app_meetme.c, apps/app_festival.c,
apps/app_skel.c: Doxygen Cleanup Start adding configuration file
linking and pages. Add module loading doxygen block. (issue
ASTERISK-20259)
2012-10-01 20:36 +0000 [r374134-374151] Sean Bright <sean@malleable.com>
* main/db.c, include/asterisk/astdb.h, /, tests/test_db.c,
apps/app_queue.c: app_queue: Support persisting and loading of
long member lists. Greenlight in #asterisk brought up that he was
receiving an error message "Could not create persistent member
string, out of space" when running app_queue in Asterisk 10.
dump_queue_members() made an assumption that 8K would be enough
to store the generated string, but with queues that have large
member lists this is not always the case. This patch removes the
limitation and uses ast_str instead of a fixed sized buffer. The
complicating factor comes from the fact that ast_db_get requires
a buffer and buffer size argument, which doesn't let us pull back
more than what we pass in, so I introduced a new
ast_db_get_allocated() which returns an ast_strdup()'d copy of
the value from astdb. As an aside, I did some testing on the
maximum size of data that we can store in the BDB library we
distribute and was able to store a 10MB string and retrieve it
with no problems, so I feel this is a safe patch. Review:
https://reviewboard.asterisk.org/r/2136/ ........ Merged
revisions 374108 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 374135 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374150 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/db.c, /: Use ast_copy_string instead of strncpy to guarantee
a NUL terminated string. ........ Merged revisions 374132 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374133 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-01 17:05 +0000 [r374109] Richard Mudgett <rmudgett@digium.com>
* main/cli.c: Change core show help output format. The CLI "core
show help" output leaves something to be desired. 1) The command
is truncated to a maximum of 30 characters. 2) The output columns
are mirrored from the 31st column. Current output format: logger
mute Toggle logging output to a console logger reload Reopens the
log files logger rotate Rotates and reopens the log files logger
set level {DEBUG|NOTICE Enables/Disables a specific logging level
for this console logger show channels List configured log
channels New format: logger mute -- Toggle logging output to a
console logger reload -- Reopens the log files logger rotate --
Rotates and reopens the log files logger set level
{DEBUG|NOTICE|WARNING|ERROR|VERBOSE|DTMF} {on|off} --
Enables/Disables a specific logging level for this console logger
show channels -- List configured log channels Review:
https://reviewboard.asterisk.org/r/2133/
2012-10-01 16:26 +0000 [r374107] Mark Michelson <mmichelson@digium.com>
* /, apps/confbridge/conf_config_parser.c: Don't destroy confbridge
config when error is encountered during a reload. Not panicking
means that the old config is kept. (closes issue ASTERISK-20458)
Reported by: Leif Madsen Patches: ASTERISK-20458.patch uploaded
by Mark Michelson(license #5049) Tested by Leif Madsen ........
Merged revisions 374106 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-10-01 12:29 +0000 [r374096] Joshua Colp <jcolp@digium.com>
* include/asterisk/speech.h, res/res_speech.c,
apps/app_speech_utils.c: Add support for retrieving engine
specific settings using the speech API and from dialplan. (closes
issue ASTERISK-17136) Reported by: kenner
2012-09-29 03:56 +0000 [r374086] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Fix ref leak when adding ICE candidates
to an SDP There was a missing decrement to the reference count
for the current ICE candidate when local candidates are being
added to an outbound SDP. This patch corrects that. ........
Merged revisions 374085 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-28 22:11 +0000 [r374075] Richard Mudgett <rmudgett@digium.com>
* res/res_agi.c: Include channel uniqueid in "AsyncAGI" and
"AGIExec" events. * Added AMI event documentation for AsyncAGI
and AGIExec events. (closes issue ASTERISK-20318) Reported by:
Dan Cropp Patches: res_agi_patch.txt (license #6422) patch
uploaded by Dan Cropp modified for trunk.
2012-09-28 19:37 +0000 [r374060] Jonathan Rose <jrose@digium.com>
* res/res_jabber.c, /: res_jabber: Remove CLI command 'jabber test'
The opinion of development was that it is both improper to have
Matt's personal email address used in the source and that the
command wouldn't be useful without it. (closes issue AST-467)
Reported by: Malcolm Davenport ........ Merged revisions 374032
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 374045 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 374059 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-28 18:27 +0000 [r374030] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, UPGRADE.txt,
main/app.c, apps/app_senddtmf.c: Add pause one second W dial
modifier. * The following dialplan applications now recognize 'W'
to pause sending DTMF for one second in addition to the
previously existing 'w' that paused sending DTMF for half a
second. Dial, ExternalIVR, and SendDTMF. * The chan_dahdi analog
port dialing and deferred DTMF dialing for PRI now distinguishes
between 'w' and 'W'. The 'w' pauses dialing for half a second.
The 'W' pauses dialing for one second. * Created dahdi_dial_str()
in chan_dahdi that eliminated a lot of duplicated dialing code
and diagnostic messages for the channel driver. (closes issue
ASTERISK-20039) Reported by: Jeremiah Gowdy Patches:
jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by
Jeremiah Gowdy Expanded patch to add support in chan_dahdi.
Tested by: rmudgett
2012-09-28 13:04 +0000 [r374020] Brent Eagles <beagles@digium.com>
* res/res_xmpp.c, main/message.c, /: Reset hangup flags on channels
created through messages and cleanup globals in res_xmpp on
unload. This patch fixes an issue where hangup flags were not
being reset on a channel, affecting subsequent use of that
channel. The patch also adds some additional cleanup to res_xmpp
to fix an issue with reloading the module. (closes
ASTERISK-20360) Reported by: Noah Engelberth Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2134/ ........ Merged
revisions 374019 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-28 12:17 +0000 [r373992] Joshua Colp <jcolp@digium.com>
* /, res/res_agi.c: Update documentation to make it explicit that
"stream file" will not restart musiconhold. (issue
ASTERISK-17367) Reported by: oej ........ Merged revisions 373989
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 373990 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373991 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-28 03:06 +0000 [r373979] Matthew Jordan <mjordan@digium.com>
* CHANGES, apps/app_senddtmf.c: Add Duration header for PlayDTMF
AMI Action This patch adds an optional header to the PlayDTMF AMI
action, Duration. It allows the duration of the DTMF digit to be
played on the channel to be specified in milliseconds. (closes
issue ASTERISK-18172) Reported by: Renato dos Santos patches:
send-dtmf.patch uploaded by Renato dos Santos (license #6267)
Modified slightly for this commit for Asterisk 12.
2012-09-27 22:43 +0000 [r373965-373967] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c: Tweak app_dial documentation.
* main/app.c: Cleanup ast_dtmf_stream() * Made ast_dtmf_stream()
wait after starting the silence generator rather than before. *
Made ast_dtmf_stream() put the peer in autoservice for the whole
time things are being done to the chan.
* apps/app_senddtmf.c, /: Fix SendDTMF crash and channel reference
leak using channel name parameter. The SendDTMF channel name
parameter has two issues. 1) Crashes if the channel name does not
exist. 2) Leaks a channel reference if the channel is the current
channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
documentation. * Renamed app to senddtmf_name and tweaked the
type. ........ Merged revisions 373945 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373946 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373954 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-27 17:12 +0000 [r373915] Joshua Colp <jcolp@digium.com>
* res/res_http_websocket.c, /, channels/chan_sip.c,
include/asterisk/http_websocket.h: Make res_http_websocket an
optional dependency on supported platforms for chan_sip. (closes
issue ASTERISK-20439) Reported by: sruffell Patches:
0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded
by sruffell (license 5417) ........ Merged revisions 373914 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-27 17:02 +0000 [r373913] Kinsey Moore <kmoore@digium.com>
* apps/app_voicemail.c, CHANGES: Add VoicemailRefresh AMI Action
Currently, if there are modifications to mailboxes that Asterisk
is not aware of, the user needs to add "pollmailboxes" to their
mailbox configuration, which repeatedly polls the subscribed
mailboxes for changes. This results in a lot of extra work for
the CPU. This patch introduces the AMI command VoicemailRefresh
which permits external applications to trigger the refresh
themselves. The refresh can apply to a specified mailbox only, an
entire context, or all configured mailboxes. Even a refresh
performed on every mailbox would not consume as much CPU as the
pollmailboxes option, given that pollmailboxes runs continuously
and this only runs on demand. (closes issue ASTERISK-17206)
(closes issue ASTERISK-19908) Reported-by: Jeff Hutchins
Reported-by: Tilghman Lesher Patch-by: Tilghman Lesher
2012-09-27 16:53 +0000 [r373881-373912] Joshua Colp <jcolp@digium.com>
* /, main/loader.c: loader: Ensure dependent modules are properly
initialized. If an Asterisk module specifies a dependency in
ast_module_info.nonoptreq, it is possible for Asterisk to skip
calling the modules's .load function. Asterisk was loading and
linking the module via load_dynamic_module() but was not adding
the module to the resource_heap. Therefore the module was not
initialized based on it's priority along with the other modules
in the heap. Now use load_resource() instead of
load_dynamic_module() for non-optional requirement. This will add
the module to the resource_heap so the module can be properly
initialized in the correct order. This is required if there are
any module global data structures initialized in the .load()
callback for the module on platforms which do not support weak
references. (issue ASTERISK-20439) Reported by: sruffell Patches:
0001-loader-Ensure-dependent-modules-are-properly-initial.patch
uploaded by sruffell (license 5417) ........ Merged revisions
373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 373910 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373911 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_local.c, /: Fix an issue where Local channels
dialed by app_queue are considered in use immediately. The
chan_local channel driver returns a device state of in use even
if a created Local channel has not yet been dialed. This fix
changes the logic to return a state of not in use until the
channel itself has been dialed. (closes issue ASTERISK-20390)
Reported by: tim_ringenbach Review:
https://reviewboard.asterisk.org/r/2116/ ........ Merged
revisions 373878 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373879 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373880 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-26 21:17 +0000 [r373852] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Move handling of 408 response so there is
no misleading warning message. (closes issue ASTERISK-20060)
Reported by: Walter Doekes ........ Merged revisions 373848 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373849 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373850 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-26 18:23 +0000 [r373835] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_meetme.c: Fixed meetme tab completion and command
documentation. * Removed unnecessary case sensitivity in meetme
list, lock, unlock, mute, unmute, and kick commands. * Separated
meetme lock/unlock, mute/unmute, and kick commands into their own
registered commands to simplify tab completion and parameter
checking. meetme_lock_cmd(), meetme_mute_cmd(), and
meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
AST-1006) Reported by: John Bigelow Tested by: rmudgett ........
Merged revisions 373815 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373816 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373818 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-26 08:31 +0000 [r373805] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_queue.c, /: app_queue: 'agent available' hint, cleanup
restart, and initial state Fix previously untested senarios; 1).
On queue initialisation set queue_avail devstate to INUSE.
Previously was unavailable, which indicated an agent was
available. 2). When removing members, if there are no other
members available, set queue_avail to INUSE. Previously, if a
member interface had become 'unavailable', they were never going
to be removed, particularly when persistant queues is enabled.
3). When adding a member, check that they are available, if they
are set queue_avail to NOT_INUSE. Previously on reloaded, members
may have been 'unavailable'. 4). When pausing or unpausing a
member, set appropriate queue availability. alecdavis (license
585) Reported by: Alec Davis Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/2129/ ........ Merged
revisions 373804 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 23:10 +0000 [r373740-373776] Mark Michelson <mmichelson@digium.com>
* /, main/say.c: Fix saying of date in Dutch. The Dutch say the
date before the month. (closes issue ASTERISK-20353) Reported by:
Teun Ouwehand ........ Merged revisions 373773 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373774 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373775 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_agent.c, configs/agents.conf.sample: Remove dead
code and documentation for nonexistent feature. multiplelogin was
removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
was removed. (closes issue AST-948) reported by Steve Pitts
........ Merged revisions 373768 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373769 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373770 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/app_voicemail.c, /: Fix error where improper IMAP greetings
would be deleted. (closes issue ASTERISK-20435) Reported by:
fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
uploaded by Michael L. Young (License #5026) (with suggested
modification made by me) ........ Merged revisions 373735 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373737 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373738 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 20:14 +0000 [r373708] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Fix T.38 support when used with
chan_local in between. Users of the T.38 API can indicate
AST_T38_REQUEST_PARMS on a channel to request that the channel
indicate a T.38 negotiation with the parameters present on the
channel. The return value of this indication is expected to be
AST_T38_REQUEST_PARMS upon success but with chan_local involved
this could never occur. This fix changes chan_local to always
return AST_T38_REQUEST_PARMS for this situation. If the
underlying channel technology on the other side does not support
T.38 this would have been determined ahead of time using
ast_channel_get_t38_state and an indication would not occur.
(closes issue ASTERISK-20229) Reported by: wdoekes Patches:
ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
https://reviewboard.asterisk.org/r/2070/ ........ Merged
revisions 373705 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373706 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373707 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 19:29 +0000 [r373701] Mark Michelson <mmichelson@digium.com>
* include/asterisk/channel.h, CHANGES, channels/sig_pri.c,
funcs/func_callerid.c, include/asterisk/callerid.h,
main/channel.c, channels/chan_misdn.c, channels/chan_sip.c,
main/callerid.c: Allow for redirecting reasons to be set to
arbitrary strings. This allows for the REDIRECTING dialplan
function to be used to set the reason to any string. The SIP
channel driver has been modified to set the redirecting reason
string to the value received in a Diversion header. In addition,
SIP 480 response reason text will set the redirecting reason as
well. (closes issue AST-942) reported by Malcolm Davenport
(closes issue AST-943) reported by Malcolm Davenport Review:
https://reviewboard.asterisk.org/r/2101
2012-09-25 19:08 +0000 [r373691] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample, channels/sip/include/sip.h, /,
channels/chan_sip.c: Properly handle UAC/UAS roles for SIP
session timers The SIP session timer mechanism contains a
mandatory 'refresher' parameter (included in the Session-Expires
header) which is used in the session timer offer/answer signaling
within a SIP Invite dialog. It looks like asterisk is
interpreting the uac resp. uas role only as the initial role of
client and server (caller is uac, callee is uas). The standard
rfc 4028 however assigns the client role to the ((RE)-Invite)
requester, the server role to the ((RE)-Invite) responder. This
patch has Asterisk track the actual refresher as "us" or "them"
as opposed to relying on just the configured "uas" or "uac"
properties. (closes issue AST-922) Reported by: Thomas Airmont
Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged
revisions 373652 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373665 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373690 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 18:33 +0000 [r373689] Kinsey Moore <kmoore@digium.com>
* /, apps/app_queue.c: "show" completion option for "queue"
shouldn't appear twice When tab-completing CLI commands starting
with "queue", "show" appeared twice in the list due to the way
that Asterisk's tab completion functions and the order in which
the commands were registered. The registration order has been
altered to resolve this issue. (closes issue AST-940)
Reported-by: Steve Pitts ........ Merged revisions 373666 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373675 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373688 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 17:22 +0000 [r373636-373656] Richard Mudgett <rmudgett@digium.com>
* /, codecs/ilbc/iLBC_encode.c, codecs/ilbc/iLBC_decode.c: Fix
valgrind found memcpy issues in codec_ilbc. Valgrind found
codec_ilbc using memcpy instead of memmove for overlapping memory
blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
#5674) patch uploaded by Walter Doekes ........ Merged revisions
373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 373645 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373650 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, codecs/Makefile: Make rebuild GSM, ilbc, or lpc10 codecs if
the respective sources change. ........ Merged revisions 373618
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 373633 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373635 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 16:45 +0000 [r373608-373634] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: chan_sip: Set Quality of Service for
video rtp instance (closes issue ASTERISK-20201) Reported by:
ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license
6008) ........ Merged revisions 373617 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373631 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373632 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_agi.c: res_agi: async_agi responsiveness improvement on
datastore problems This patch changes get_agi_cmd so that the
return can be checked to differentiate between an empty list
success and something that triggered an error. This in turn
allows launch_asyncagi to detect these errors and break free from
the command processing loop so that the async agi can be ended
more cleanly (closes issue ASTERISK-20109) Reported by: Jeremiah
Gowdy Patches: jgowdy-7-9-2012.diff uploaded by Jeremiah Gowdy
(license 6358) (Modified by me to fix some logical issues and
apply to trunk) Review: https://reviewboard.asterisk.org/r/2117/
2012-09-25 14:13 +0000 [r373583] Mark Michelson <mmichelson@digium.com>
* funcs/func_presencestate.c, /: "He who go through turnstile
sideways is going to Bangkok" ........ Merged revisions 373582
from http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 13:29 +0000 [r373581] Kinsey Moore <kmoore@digium.com>
* configs/res_odbc.conf.sample, /: Fix documentation for default
username in res_odbc This was previously stated to be "root", but
is actually the name of the context if unspecified. (closes issue
ASTERISK-20258) Reported by: Stefan x ........ Merged revisions
373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 373579 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373580 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-25 12:12 +0000 [r373553] Joshua Colp <jcolp@digium.com>
* /, res/res_rtp_multicast.c: Fix an issue where a caller to
ast_write on a MulticastRTP channel would determine it failed
when in reality it did not. When sending RTP packets via
multicast the amount of data sent is stored in a variable and
returned from the write function. This is incorrect as any
non-zero value returned is considered a failure while a return
value of 0 is success. For callers (such as ast_streamfile) that
checked the return value they would have considered it a failure
when in reality nothing went wrong and it was actually a success.
The write function for the multicast RTP engine now returns -1 on
failure and 0 on success, as it should. (closes issue
ASTERISK-17254) Reported by: wybecom ........ Merged revisions
373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 373551 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373552 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-24 22:14 +0000 [r373503] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Be consistent, send From: "Anonymous"
<sip:anonymous@anonymous.invalid> When setting
CALLERID(pres)=unavailable in the dialplan, the From header in
the SIP message contains "Anonymous"
<sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
should use a lowercase a in the userpart of the URI. * Make the
From header use a lowercase A in the userpart of the anonymous
URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
patch uploaded by Antti Yrjola ........ Merged revisions 373500
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 373501 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373502 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-24 21:19 +0000 [r373479] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c, funcs/func_audiohookinherit.c, /:
func_audiohookinherit: Document some missed sources. This patch
also mentions that AUDIOHOOK_INHERIT can be used to transfer
MixMonitor audiohooks. There is also wiki that addresses
audiohooks and the use of AUDIOHOOK_INHERIT at the following
link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
(closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........
Merged revisions 373467 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373468 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373470 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-24 21:15 +0000 [r373471] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Fix potential reentrancy problems in
chan_sip. Asterisk v1.8 and later was not as vulnerable to this
issue. * Made find_call() lock each private as it processes the
found dialogs. (Primary cause of ABE-2876) * Made the other
functions that traverse the dialogs container lock each private
as it examines them. * Fix race condition in sip_call() if the
thread that sent the INVITE is held up long enough for a response
to be processed. The p->initid for the INVITE retransmission
could be added after it was canceled by the response processing.
* Made __sip_destroy() clean up resource pointers after freeing.
This is primarily defensive in case someone has a stale private
pointer. * Removed redundant memset() in reqprep(). The call to
init_req() already does the memset() and is the first reference
to req in reqprep(). * Removed useless set of req.method in
transmit_invite(). The calls to initreqprep() and reqprep() have
to do this because they memset() the req. JIRA ABE-2876
.......... Merged -r373423 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........ Merged revisions 373424 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373466 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373469 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-24 19:23 +0000 [r373414-373456] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Fix a deadlock caused by a race condition
between removing a hint and reloading the dialplan and
subscribing to the removed hint. If conditions were right it was
possible for both the PBX core and chan_sip to deadlock by both
having a lock that the other wants. In the case of the PBX core
it had the contexts lock and wanted a SIP dialog lock, while in
the case of chan_sip it had the SIP dialog lock and wanted the
contexts lock. This fix unlocks the SIP dialog before getting the
extension state so that the other thread will not block on trying
to lock it. Once the extension state is retrieved the SIP dialog
is locked again and life carries on. As the SIP dialog is
reference counted it is not possible for it to go away after
unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
........ Merged revisions 373438 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373440 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373454 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_format_attr_h264.c, /, channels/chan_sip.c: Fix an issue
with H.264 format attribute comparison and fix an issue with
improper SDP being produced. The H.264 format attribute module
compares two format attribute structures to determine if they are
compatible or not. In some instances it was possible for this
check to determine that both structures were incompatible when
they actually should be considered compatible. This check has now
been made even more permissive by assuming that if no attribute
information is available the two structures are compatible. If
both structures contain attribute information a base level
comparison of the H.264 IDC value is done to see if they are
compatible or not. The above issue uncovered a secondary issue in
chan_sip where the SDP being produced would be incorrect if the
formats were considered incompatible. This has now been fixed by
checking that all information required to produce the SDP is
available instead of assuming it is. (closes issue
ASTERISK-20464) Reported by: Leif Madsen ........ Merged
revisions 373413 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-24 12:42 +0000 [r373404] Brent Eagles <beagles@digium.com>
* res/res_rtp_asterisk.c, /, configs/rtp.conf.sample:
res_rtp_asterisk: Make TURN and STUN server configurations
consistent. This patch removes the turnport configuration
property and changes the turnaddr property to be a combined
host[:port] configuration string. The patch also modifies the
documentation in the example configuration to reflect the
property changes and adds some additional text indicating how the
STUN port is configured. (closes issue ASTERISK-20344) Reported
by: beagles Tested by: beagles Review:
https://reviewboard.asterisk.org/r/2111/ ........ Merged
revisions 373403 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-22 20:43 +0000 [r373384] Andrew Latham <lathama@gmail.com>
* Makefile, cel/cel_odbc.c, include/asterisk/doxyref.h,
main/manager.c, doc/README.txt, include/asterisk/xmpp.h,
apps/app_minivm.c, cel/cel_sqlite3_custom.c,
include/asterisk/format.h, main/audiohook.c,
include/asterisk/pbx.h, res/res_timing_kqueue.c,
addons/chan_mobile.c, main/asterisk.c, main/xmldoc.c,
channels/chan_mgcp.c, apps/app_voicemail.c, utils/refcounter.c,
res/res_config_pgsql.c, main/pbx.c, main/ccss.c,
channels/chan_sip.c, tests/test_gosub.c,
include/asterisk/doxygen/mantisworkflow.h (removed),
contrib/asterisk-ng-doxygen, channels/chan_agent.c, main/astfd.c,
apps/app_queue.c, codecs/speex/speex_resampler.h,
res/res_config_sqlite.c: Doxygen Updates Janitor Work *
Whitespace, doc-blocks, spelling, case, missing and incorrect
tags. * Add cleanup to Makefile for the Doxygen configuration
update * Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation *
remove mantisworkflow... * update documentation README * Add
markup to Tilghman's email and talk with him about updating his
email, he knows... * no code changes on this commit other than
the mentioned Makefile change (issue ASTERISK-20259)
2012-09-21 19:35 +0000 [r373369] Jonathan Rose <jrose@digium.com>
* /, channels/iax2-provision.c: iax2-provision: Fix improper return
on failed cache retrieval (closes issue ASTERISK-20337) reported
by: John Covert Patches: iax2-provision.c.patch uploaded by John
Covert (license 5512) ........ Merged revisions 373342 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373343 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373368 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-21 18:22 +0000 [r373320-373341] Andrew Latham <lathama@gmail.com>
* contrib/asterisk-ng-doxygen: Update Doxygen Config Comments This
annoying update is almost totally whitespace and updated config
comments. I did add Python to the documented file types. (issue
ASTERISK-20259)
* include/asterisk/localtime.h, apps/app_ices.c, cdr/cdr_pgsql.c,
res/res_xmpp.c, res/res_jabber.c, cdr/cdr_radius.c,
include/asterisk/doxygen/releases.h, include/asterisk/doxyref.h,
res/res_smdi.c, main/manager.c, main/tdd.c,
include/asterisk/bridging_features.h, main/ast_expr2f.c,
cdr/cdr_sqlite.c, apps/app_skel.c, include/asterisk/sip_api.h,
channels/chan_motif.c, main/http.c, apps/app_confbridge.c,
include/asterisk/doxygen/commits.h, res/res_config_ldap.c,
res/res_curl.c, main/strings.c, res/res_config_pgsql.c,
codecs/codec_speex.c, res/res_crypto.c, main/acl.c,
channels/chan_console.c, res/res_config_curl.c,
channels/chan_jingle.c, include/asterisk/app.h,
include/asterisk/res_odbc.h, channels/chan_misdn.c,
include/asterisk/doxygen/asterisk-git-howto.h,
include/asterisk/xmpp.h, include/asterisk/jabber.h,
channels/chan_h323.c, include/asterisk/doxygen/reviewboard.h,
channels/sip/include/sdp_crypto.h, main/asterisk.c,
main/xmldoc.c, include/asterisk/doxygen/architecture.h,
include/asterisk/acl.h, cel/cel_pgsql.c, funcs/func_speex.c,
cel/cel_radius.c, apps/app_meetme.c, main/ccss.c, res/res_snmp.c,
include/asterisk/doxygen/mantisworkflow.h, main/sha1.c,
channels/sip/reqresp_parser.c: Doxygen Updates - janitor work
Doxygen updates including mistakes, misspellings, missing
parameters, updates for Doxygen style. Some missing txt file
links are removed but their content or essense will be included
in some later updates. A majority of the txt files were removed
in the 1.6 era but never noted. The HR and EXTREF are simple
changes that make the documentation more compatable with more
versions of Doxygen. Further updates coming. (issue
ASTERISK-20259)
* README: Start work on documentation janitor project with a little
commit. This adds a link to the Asterisk wiki at
https://wiki.asterisk.org to the README file. (issue
ASTERISK-20259)
2012-09-21 15:41 +0000 [r373319] Jonathan Rose <jrose@digium.com>
* /, apps/app_queue.c: app_queue: Make queue reload members and
variants of that work Prior to this patch, 'queue reload members'
cli command did not work at all. This also affects the manager
function 'QueueReload' when supplied with the 'members: yes'
field. (closes issue AST-956) Reported by: John Bigelow ........
Merged revisions 373298 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373300 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373318 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-21 09:11 +0000 [r373275-373284] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c: dsp.c: remove more whitespace mentioned in review2107
* main/dsp.c: dsp.c ast_dsp_call_progress use local short variable
in loop, plus other cleanup janitor cleanup. No functional
change. 1). ast_dsp_call_progress: use 'short samp' instead of
s[x] inside loop. apply same casting as other _init, dsp->energy
= (int32_t) samp * (int32_t) samp 2). ast_dtmf_detect_init: move
repeated setting of s->energy to outside of loop. do
goertzel_init loop first before setting s->lasthit and
s->current_hit, consistant with ast_dsp_digitreset() 3).
ast_mf_detect_init: do goertzel_init loop first before setting
s->hits[] and s->current_hit, consistant with
ast_dsp_digitreset() 4). Don't chain init different variables, as
the type may change Review
https://reviewboard.asterisk.org/r/2107/
2012-09-20 19:16 +0000 [r373247] Joshua Colp <jcolp@digium.com>
* /, apps/app_meetme.c: Fix incorrect MeetME conference bridge
reference count decrementing and sometimes premature destruction.
When using the 'e' or 'E' option to MeetMe the configured
conference bridges are loaded and examined to see if any are
empty. If no conference bridges are empty the caller is prompted
to enter the number of one. This operation left around a pointer
to the last created conference bridge still containing
participants. When the caller that was not able to find any empty
conference bridge hung up this pointer was disposed of and the
reference count of the conference bridge decremented. If there
was only a single participant in the conference bridge it was
ultimately destroyed prematurely. (closes issue AST-994) Reported
by: John Bigelow ........ Merged revisions 373242 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373245 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373246 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-20 18:44 +0000 [r373239] Matthew Jordan <mjordan@digium.com>
* CHANGES, apps/app_queue.c, configs/extensions.conf.sample, /: Add
queue monitoring hints This patch adds support for hints on a
queue. Hints can be added using the nomenclature 'Queue:name',
where name is the name of the queue being monitored. This nifty
feature was done by Alec Davis. Review:
https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis
Tested by: alecdavis patches: review1619.diff2 by alecdavis
(license 585) ........ Merged revisions 373235 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-20 18:27 +0000 [r373234] Joshua Colp <jcolp@digium.com>
* channels/sip/include/sip.h, res/res_rtp_asterisk.c,
main/rtp_engine.c, /, channels/chan_sip.c, configure,
include/asterisk/autoconfig.h.in, configure.ac,
configs/sip.conf.sample, include/asterisk/rtp_engine.h: Add
support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As
mentioned on the review for this, WebRTC has moved towards
choosing DTLS-SRTP as the mechanism for key exchange for SRTP.
This commit adds support for this but makes it available for
normal SIP clients as well. Testing has been done to ensure that
this introduces no regressions with existing behavior and also
that it functions as expected. Review:
https://reviewboard.asterisk.org/r/2113/ ........ Merged
revisions 373229 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-20 18:02 +0000 [r373222] Matthew Jordan <mjordan@digium.com>
* apps/app_queue.c: Support all ways a member can be available for
'agent available' hints Alec's patch in r373188 added the ability
to subscribe to a hint for when Queue members are available. This
patch modifies the check that determines when a Queue member is
available by refactoring the availability checks in
num_available_members into a shared function is_member_available.
This should now handle the ringinuse option, as well as device
state values other than AST_DEVICE_NOT_INUSE.
2012-09-20 17:22 +0000 [r373221] Richard Mudgett <rmudgett@digium.com>
* apps/app_directed_pickup.c, funcs/func_channel.c,
main/features.c, include/asterisk/channel.h,
include/asterisk/features.h, main/channel.c, /: Named call pickup
groups. Fixes, missing functionality, and improvements. *
ASTERISK-20383 Missing named call pickup group features:
CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() -
Needs to also select from named pickup groups. * ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail
even though there was a call it could have picked up. In a call
pickup race when there are multiple calls to pickup and two
extensions try to pickup a call, it is conceivable that the loser
will not pick up any call even though it could have picked up the
next oldest matching call. Regression because of the named call
pickup group feature. * See ASTERISK-20386 for the implementation
improvements. These are the changes in channel.c and channel.h. *
Fixed some locking issues in CHANNEL(). (closes issue
ASTERISK-20383) Reported by: rmudgett (closes issue
ASTERISK-20384) Reported by: rmudgett (closes issue
ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/2112/ ........ Merged
revisions 373220 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-20 13:04 +0000 [r373212] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Correct handling of unknown SDP stream
types When the patch to handle arbitrary SDP stream arrangements
went into Asterisk, it also included an ability to transparently
decline unknown stream types. The scanf calls used were not
checked properly causing this part of the functionality to be
broken. (closes issue ASTERISK-20203) ........ Merged revisions
373211 from http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-20 11:05 +0000 [r373203] Sean Bright <sean@malleable.com>
* res/res_curl.c: When trying to unload res_curl.so, warn about all
dependent modules. Before this, attempting to unload res_curl.so
would warn you about the first module it found that was
dependent. We now warn about all of the loaded modules instead.
2012-09-20 10:41 +0000 [r373188-373202] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c: dsp.c: remove whitespace mentioned in review2107
Related https://reviewboard.asterisk.org/r/2107/
* CHANGES, apps/app_queue.c, configs/extensions.conf.sample:
app_queue: Support an 'agent available' hint Sets INUSE when no
free agents, NOT_INUSE when an agent is free. modifes
handle_statechange() scan members loop to scan for a free agent
and updates the Queue:queuename_avial devstate. Previously exited
early if the member was found in the queue. Now Exits later when
both a member was found, and a free agent was found. alecdavis
(license 585) Reported by: Alec Davis Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2121/
2012-09-18 20:19 +0000 [r373134-373142] Sean Bright <sean@malleable.com>
* main/logger.c: Make the casing of CALL_ID in debug messages
consistent to satisfy my OCD.
* main/manager.c, /: Don't crash when passing a NULL message to
__astman_get_header. Before this commit, __astman_get_header
would blindly dereference the passed in 'struct message *' to
traverse the header list. There are cases, however, such as
'*CLI> sip qualify peer foo' where the message pointer is NULL,
so we need to check for that. ........ Merged revisions 373131
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 373132 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373133 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-18 15:50 +0000 [r373120] David M. Lee <dlee@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
makeopts.in, Makefile, include/asterisk/utils.h: Add
-fnested-functions compile flag, if needed. In order to use
nested functions on some versions of GCC (e.g. GCC on OS X), the
-fnested-functions flag must be passed to the compiler. This
patch adds detection logic to ./configure to add the flag if
necessary. It also adds a comment to utils.h as to why the nested
function needs a prototype. (closes issue ASTERISK-20399)
Reported by: David M. Lee Review:
https://reviewboard.asterisk.org/r/2102/ ........ Merged
revisions 373119 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-15 00:32 +0000 [r373108] Richard Mudgett <rmudgett@digium.com>
* channels/sig_ss7.c, /: Made companding law for SS7 calls only
determined by SS7 signaling type. For SS7, the companding law for
a call was chosen inconsistently depending upon ss7type (ITU vs
ANSI) and the DAHDI companding default (T1 vs E1). For incoming
calls, the companding law was determined by ss7type. For outgoing
calls, the companding law was determined by the DAHDI default.
With the wrong combination you would get A-law/u-law conflicts.
An A-law/u-law conflict sounds like bad static on the line. SS7
ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
with T1 line: ok * Fix the companding law used to be determined
by the SS7 signaling type only. ........ Merged revisions 373090
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 373101 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373107 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-14 19:53 +0000 [r373080] Matthew Jordan <mjordan@digium.com>
* main/libasteriskssl.c, main/tcptls.c, /, channels/chan_sip.c:
Resolve memory leaks in TLS initialization and TLS client
connections This patch resolves two sources of memory leaks when
using TLS in Asterisk: 1) It removes improper initialization (and
multiple re-initializations) of portions of the SSL library.
Asterisk calls SSL_library_init and SSL_load_error_strings during
SSL initialization; collectively this obviates the need for
calling any of the following during initialization or client
connection handling: * ERR_load_crypto_strings (handled by
SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
SSL_library_init) 2) Failure to completely clean up all memory
allocated by Asterisk and by the SSL library for TLS clients.
This included not freeing the SSL_CTX object in the SIP channel
driver, as well as not clearing the error stack when the TLS
client exited. Note that these memory leaks were found by Thomas
Arimont, and this patch was essentially written by him with some
minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
Arimont (license 5525) Review:
https://reviewboard.asterisk.org/r/2105 ........ Merged revisions
373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 373062 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373079 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-13 20:05 +0000 [r373046-373048] David M. Lee <dlee@digium.com>
* /, main/Makefile: Fixed make clean when configured
--disable-asteriskssl ........ Merged revisions 373047 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/channel.c, /, include/asterisk/channel.h: Fix timeouts for
ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
its timeout to ast_waitfor_nandfds, expecting it to decrement the
timeout by however many milliseconds were waited. This is a
problem if it consistently waits less than 1ms. The timeout will
never be decremented, and we wait... FOREVER! This patch makes
ast_waitfordigit_full manage the timeout itself. It maintains the
previously undocumented behavior that negative timeouts wait
forever. (closes issue ASTERISK-20375) Reported by: Mark
Michelson Tested by: Mark Michelson Review:
https://reviewboard.asterisk.org/r/2109/ ........ Merged
revisions 373024 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 373025 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 373029 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-12 21:02 +0000 [r372997] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c, main/channel.c, include/asterisk/astobj2.h,
tests/test_astobj2.c: Enhance astobj2 to support other types of
containers. The new API allows for sorted containers, insertion
options, duplicate handling options, and traversal order options.
* Adds the ability for containers to be sorted when they are
created. * Adds container creation options to handle duplicates
when they are inserted. * Adds container creation option to
insert objects at the beginning or end of the container traversal
order. * Adds OBJ_PARTIAL_KEY to allow searching with a partial
key. The partial key works similarly to the OBJ_KEY flag. (The
real search speed improvement with this flag will come when
red-black trees are added.) * Adds container traversal and
iteration order options: Ascending and Descending. * Adds an
AST_DEVMODE compile feature to check the stats and integrity of
registered containers using the CLI "astobj2 container stats
<name>" and "astobj2 container check <name>". The channels
container is normally registered since it is one of the most
important containers in the system. * Adds ao2_iterator_restart()
to allow iteration to be restarted from the beginning. * Changes
the generic container object to have a v_method table pointer to
support other types of containers. * Changes the container nodes
holding objects to be ref counted. The ref counted nodes and
v_method table pointer changes pave the way to allow other types
of containers. * Includes a large astobj2 unit test enhancement
that tests the new features. (closes issue ASTERISK-19969)
Reported by: rmudgett Review:
https://reviewboard.asterisk.org/r/2078/
2012-09-12 20:54 +0000 [r372996] Joshua Colp <jcolp@digium.com>
* channels/chan_motif.c, /: Skip any non-content information when
looking for and handling content. This fixes a bug with Jitsi and
conference calling. Jitsi implements XEP-0298 which places some
conference-info information in the session-initiate request which
chan_motif did not expect to occur. ........ Merged revisions
372995 from http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-12 18:33 +0000 [r372976-372985] Jonathan Rose <jrose@digium.com>
* /, res/res_xmpp.c: res_xmpp: Fix a segfault caused by bodyless
messages (closes issue ASTERISK-20361) Reported by: Noah
Engelberth Review: https://reviewboard.asterisk.org/r/2108/
........ Merged revisions 372984 from
http://svn.asterisk.org/svn/asterisk/branches/11
* configs/logger.conf.sample, main/logger.c: logger: Add
rotatestrategy option of 'none' which does not perform rotations
With this option in use, it may be necessary to regulate your log
files externally. (closes issue ASTERISK-20189) Reported by: Jaco
Kroon Patches: asterisk-logger-norotate-trunk.patch uploaded by
Jaco Kroon (license 5671)
2012-09-12 15:21 +0000 [r372943] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Add channel name to a warning to make
debugging easier. The "autodestruct with owner in place" message
is typically indicative of a channel reference leak. Printing out
the name of the channel in the message may be helpful when trying
to debug the issue. ........ Merged revisions 372932 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372933 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372937 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-12 14:22 +0000 [r372931] David M. Lee <dlee@digium.com>
* /, main/Makefile: Fixed r372696 when configured
--disable-asteriskssl; properly install libasteriskssl.dylib on
OS X. I didn't realize that libasteriskssl.c was still compiled,
even when you disable asteriskssl; it simple gets statically
linked into asterisk. ........ Merged revisions 372930 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-11 22:40 +0000 [r372918] Jonathan Rose <jrose@digium.com>
* channels/chan_local.c, /: chan_local: Switch from using a random
4 digit hex identifier to unique id Changes chan_local channels
to use an 8 digit hex identifier generated atomically and
sequentially in order to eliminate the chance of having multiple
channels with the same name during high call volume situations.
(issue ASTERISK-20318) Reported by: Dan Cropp Review:
https://reviewboard.asterisk.org/r/2104/ ........ Merged
revisions 372902 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372916 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372917 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-11 21:17 +0000 [r372887-372891] Mark Michelson <mmichelson@digium.com>
* include/asterisk/_private.h, main/message.c, main/asterisk.c, /:
Fix inability to shutdown gracefully due to an unending channel
reference. message.c makes use of a special message queue channel
that exists in thread storage. This channel never goes away due
to the fact that the taskprocessor used by message.c does not get
shut down, meaning that it never ends the thread that stores the
channel. This patch fixes the problem by shutting down the
taskprocessor when Asterisk is shut down. In addition, the thread
storage has a destructor that will release the channel reference
when the taskprocessor is destroyed. (closes issue AST-937)
Reported by Jason Parker Patches: AST-937.patch uploaded by Mark
Michelson (License #5049) Tested by Jason Parker ........ Merged
revisions 372885 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372888 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/features.c, /: Fix bad channel application data reference.
When channels get bridged due to an AMI bridge action or a DTMF
attended transfer, the two channels that get bridged have their
application data pointing to the other channel's name. This means
that if one channel is hung up but the other moves on, it means
that the channel that moves on will have its application data
pointing at freed memory. (issue ASTERISK-20335) Reported by:
aragon ........ Merged revisions 372840 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372841 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372886 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-11 18:09 +0000 [r372874] David M. Lee <dlee@digium.com>
* Makefile, /: Corrects the astsbindir setting when installing the
sample asterisk.conf. (closes issue ASTERISK-20406) ........
Merged revisions 372863 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372864 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-11 14:43 +0000 [r372808-372832] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt, CHANGES: chan_sip: Fix CHANGES and UPGRADE.txt for
r372808 (issue AST-969) Reported by John Bigelow
* channels/chan_sip.c: chan_sip: Change SIPQualifyPeer to improve
initial response time Prior to this patch, The acknowledgement
wasn't produced until after executing the sip_poke_peer action
actually responsible for qualifying the peer. Now the response is
given immediately once it is known that a peer will be qualified
and a SIPqualifypeerdone event is issued when the process is
finished. Thanks to OEJ for identifying the problem and helping
to come up with a solution. (issue AST-969) Reported by John
Bigelow Review: https://reviewboard.asterisk.org/r/2098/
2012-09-10 21:00 +0000 [r372796-372807] Kinsey Moore <kmoore@digium.com>
* channels/chan_iax2.c, /: Ensure iax2 debug output is displayed
when expected When IAX2 debug was changed from iax_showframe to
iax_outputframe, some instances were missed (or added afterward).
This was causing debug output to not be displayed when expected.
(closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
John Covert ........ Merged revisions 372804 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372805 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372806 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/devicestate.c, channels/chan_gtalk.c, res/res_jabber.c,
channels/chan_jingle.c, include/asterisk/doxygen/architecture.h:
Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk,
chan_jingle, and res_jabber are now deprecated in favor of using
chan_motif and res_xmpp. They are a feature-equivalent
replacement and are written to be more easily maintainable.
(closes issue ASTERISK-20298) Review:
https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen
........ Merged revisions 372795 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-10 19:22 +0000 [r372787] David M. Lee <dlee@digium.com>
* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Eliminate
"type-punned pointer" build warning. Removes
"res_rtp_asterisk.c:706: warning: dereferencing type-punned
pointer will break strict-aliasing rules" warning from the build
on 32-bit platforms. The problem is that 'size' was referenced
aliased to both (pj_size_t *) and (pj_ssize_t *). Now just make a
copy of size that is the right type so there isn't any pointer
aliasing happening. It also adds comments and asserts regarding
what looks like an inappropriate use of pj_sock_sendto, but is
actually totally fine. (closes issue ASTERISK-20368) Reported by:
Shaun Ruffell Tested by: Michael L. Young Patches:
0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch
uploaded by Shaun Ruffell (license 5417) slightly modified by
David M. Lee. ........ Merged revisions 372777 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-10 18:58 +0000 [r372755-372769] Jonathan Rose <jrose@digium.com>
* /, apps/app_meetme.c: app_meetme: Document that 'p' option will
continue in dialplan. (closes issue AST-991) Reported by John
Bigelow ........ Merged revisions 372765 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372767 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372768 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/channel.c: Masquerade: Retain parkinglot settings made by
CHANNEL function. Prior to this patch, the user would have a
parkinglot set on a channel that was parked and when the channel
was retrieved, any attempt by that channel to park would simply
use the default. This patch makes parkinglot values set in this
way be retained through the masquerade. (closes issue AST-990)
Reported by: Nick Huskinson Patches:
masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
(license 6182) ........ Merged revisions 372736 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372737 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372754 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-09 01:28 +0000 [r372712] Matthew Jordan <mjordan@digium.com>
* channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when
needed In r356604, SRTP handling was fixed to accomodate multiple
crypto keys in an SDP offer and the ability to re-create an SRTP
session when the crypto keys changed. In certain circumstances -
most notably when a phone is put on hold after having been
bridged for a significant amount of time - the act of re-creating
the SRTP session causes problems for certain models of phones.
The patch committed in r356604 always re-created the SRTP session
regardless of whether or not the cryptographic keys changed.
Since this is technically not necessary, this patch modifies the
behavior to only re-create the SRTP session if Asterisk detects
that the remote key has changed. This allows models of phones
that do not handle the SRTP session changing to continue to work,
while also providing the behavior needed for those phones that do
re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported
by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 372710 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372711 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-08 06:18 +0000 [r372699] David M. Lee <dlee@digium.com>
* /, main/Makefile: Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and
tcptls.c. Without this flag, those files will compile with the
system installed OpenSSL headers (if they exist). This is a real
bummer if a different path was specified using --with-ssl=
(closes issue ASTERISK-20392) ........ Merged revisions 372682
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Recorded merge of revisions 372695 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........
Recorded merge of revisions 372696 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-07 23:10 +0000 [r372623-372658] Richard Mudgett <rmudgett@digium.com>
* /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup().
(closes issue ASTERISK-20349) Reported by: Brent Eagles ........
Merged revisions 372655 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372656 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372657 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, funcs/func_math.c: Remove annoying unconditional debug message
from INC/DEC functions. (closes issue AST-1001) Reported by:
Guenther Kelleter ........ Merged revisions 372628 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372629 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372630 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/app_queue.c, /: Fix exception path typo in app_queue.c
try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
Pepper Patches: fix-local-channel-locking.patch (license #6350)
patch uploaded by Jeremy Pepper ........ Merged revisions 372624
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 372625 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372626 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers
ServerEmail and MailCommand reported values. The AMI action
VoicemailUsersList VoicemailUserEntry event headers ServerEmail
and MailCommand did not report the global values if they were not
overridden. The VoicemailUserEntry event header ServerEmail was
not populated with the global value if the voicemail user did not
override it. The VoicemailUserEntry event header MailCommand was
never populated with a value. * Removed unused struct ast_vm_user
member mailcmd[]. (closes issue AST-973) Reported by: John
Bigelow Tested by: rmudgett ........ Merged revisions 372620 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372621 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372622 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-07 21:04 +0000 [r372610-372612] David M. Lee <dlee@digium.com>
* res/pjproject/pjmedia/lib, codecs/ilbc,
res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
res/pjproject/third_party/gsm/lib,
res/pjproject/third_party/gsm/bin, res/pjproject/pjnath/lib,
res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/lib,
res/pjproject/pjsip/bin, res/pjproject/pjsip-apps/bin,
res/pjproject/third_party/lib, res/pjproject/third_party/bin,
res/pjproject/lib, res/pjproject/pjlib/lib, /: svn:ignore
cleanup. * pjproject bin and lib directories should pretty much
ignore everything * Ignore *.o in codecs/ilbc ........ Merged
revisions 372611 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, res/Makefile: Fix parallel make for res_asterisk_rtp. Fixes a
build regression introduced in r369517 "Add support for
ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1]
http://svnview.digium.com/svn/asterisk?view=revision&revision=369517
When compiling asterisk in parallel like: $ make -j 10 It's
possible to get errors like the following:
.pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing
separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep]
Error 1 make[2]: ***
[/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a]
Error 2 make[3]: warning: jobserver unavailable: using -j1. Add
`+' to parent make rule. This is because the build system is
trying to build each of the libraries in pjproject in parallel.
Now the build will build pjproject in a single job and link the
results into res_asterisk_rtp. Parallel builds, on one test
system, saves ~1.5 minutes from a default Asterisk build: Single
job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null
2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys
0m15.970s Parallel make: $ git clean -fdx >/dev/null && time (
./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real
1m2.353s user 2m39.120s sys 0m18.850s (closes issue
ASTERISK-20362) Reported by: Shaun Ruffel Patches:
0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch
uploaded by Shaun Ruffel (License #5417) ........ Merged
revisions 372609 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-07 02:27 +0000 [r372538-372584] Matthew Jordan <mjordan@digium.com>
* /, apps/app_minivm.c: Free ast_str objects when temp file fails
to be created in MiniVM The previous commit (r372554) was from a
patch that was written before r366880, which ensured that ast_str
objects allocated in the sendmail routine were free'd in off
nominal paths. This commit frees the string objects in the off
nominal path introduced in r372554. (issue ASTERISK-17133)
Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372582 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372583 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_minivm.c: Fix file descriptor leak and pointer scope
issue in MiniVM when sending mail When MiniVM sends an e-mail and
it has the volgain option set, it will spawn sox in a separate
process to handle the manipulation of the sound file. In doing
so, it creates a temporary file. There are two problems here: 1)
The file descriptor returned from mkstemp is leaked 2) The
finalfilename character pointer points to a buffer that loses
scope once volgain processing is finished. Note that in r316265,
Russell fixed some gcc warnings by using the return value of the
mkstemp call. A warning was placed in minivm that the file
descriptor was going to be leaked. This patch reverts that
change, as it handles the leak and 'uses' the file descriptor
returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
Cohen (license #5035) ........ Merged revisions 372554 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372555 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372556 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_queue.c: Update QueueMemberStatus event documentation
to include member status values The Status: header in a
QueueMemberStatus event (and other QueueMember* events) is the
numeric value of the device state corresponding to that Queue
Member. As those values are not exactly obvious, listing them in
the documentation is useful. Matt Riddell reported this
indirectly through the wiki page. (closes issue ASTERISK-20243)
Reported by: Matt Riddell ........ Merged revisions 372531 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-06 22:14 +0000 [r372524] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, /: Fix loss of MOH on an ISDN channel when
parking a call for the second time. Using the AMI redirect action
to take an ISDN call out of a parking lot causes the MOH state to
get confused. The redirect action does not take the call off of
hold. When the call is subsequently parked again, the call no
longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
repeated AST_CONTROL_HOLD frames if it is already in a state
where it is supposed to be sending MOH. The MOH may have been
stopped by other means. (Such as killing the generator.) This
simple fix is done rather than making the AMI redirect action
post an AST_CONTROL_UNHOLD unconditionally when it redirects a
channel and thus potentially breaking something with an
unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
rmudgett ........ Merged revisions 372521 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........ Merged revisions 372522 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372523 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-06 21:43 +0000 [r372520] Kinsey Moore <kmoore@digium.com>
* /, apps/app_queue.c: Ensure listed queues are not offered for
completion When using tab-completion for the list of queues on
"queue reset stats" or "queue reload
{all|members|parameters|rules}", the tab-completion listing for
further queues erroneously listed queues that had already been
added to the list. The tab-completion listing now only displays
queues that are not already in the list. (closes issue AST-963)
Reported-by: John Bigelow ........ Merged revisions 372517 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372518 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372519 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-06 15:57 +0000 [r372474] Jonathan Rose <jrose@digium.com>
* /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how
directmediapermit/deny ACL works r366547 introduced a change to
the directmedia ACL for chan_sip which modified the behavior
significantly. Prior to the patch, this option would bridge peers
with directmedia if a peer's IP address matched its own
directmedia ACL. After that patch, the peer would check the
bridged peer's ACL instead. This change has been present since
1.8.14.0. That patched failed to document the change in
Upgrade.txt, so this patch adds mention of that change to
UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
........ Merged revisions 372471 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372472 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372473 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-06 14:31 +0000 [r372447] Kinsey Moore <kmoore@digium.com>
* /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue
show" Previously, tabbing at the end of "queue show" produced a
list of available queues about which information could be shown,
but did not include an alternative command, "rules", to access
information about queue rules. The "rules" item should now be
shown in the list of tab-completable items. (closes issue
AST-958) Reported-by: John Bigelow ........ Merged revisions
372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 372445 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372446 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-06 02:52 +0000 [r372393-372420] Matthew Jordan <mjordan@digium.com>
* /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when
neighboring peer is unreachable Consider a scenario where DUNDi
peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
and where PBX2 and PBX3 are also neighbors. If the connection is
temporarily broken between PBX1 and PBX3, PBX1 should not include
PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
message, as it cannot send messages to PBX3. If it does, PBX2
will assume that PBX3 already received the message and fail to
forward the message on to PBX3 itself. This patch fixes this by
only including peers in a DPDISCOVER message that are reachable
by the sending node. This includes all peers with an empty
address (00:00:00:00:00:00) and that are have been reached by a
qualify message. This patch also prevents attempting to qualify a
dynamic peer with an empty address until that peer registers. The
patch uploaded by Peter was modified slightly for this commit.
(closes issue ASTERISK-19309) Reported by: Peter Racz patches:
dundi_routing.patch uploaded by Peter Racz (license 6290)
........ Merged revisions 372417 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372418 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372419 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_followme.c: Allow configured numbers for FollowMe to
be greater than 90 characters When parsing a 'number' defined in
followme.conf, FollowMe previously parsed the number in the
configuration file into a buffer with a length of 90 characters.
This can artificially limit some parallel dial scenarios. This
patch allows for numbers of any length to be defined in the
configuration file. Note that Clod Patry originally wrote a patch
to fix this problem and received a Ship It! on the JIRA issue.
The patch originally expanded the buffer to 256 characters.
Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the
application. (closes issue ASTERISK-16879) Reported by: Clod
Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
by Clod Patry (license #5138) Slightly modified for this commit.
........ Merged revisions 372390 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372391 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372392 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-05 19:44 +0000 [r372374] Richard Mudgett <rmudgett@digium.com>
* /: Recorded merge of revisions 372373 from
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Fix
compile error. ........ Merged revisions 372372 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-09-05 19:26 +0000 [r372344-372371] Kinsey Moore <kmoore@digium.com>
* main/manager.c, /: Correct documentation for ModuleLoad AMI
action The documentation incorrectly listed 'rtp' as a reloadable
subsystem and left out many other reloadable subsystems. It is
now also documented that subsystems may only be reloaded, not
loaded or unloaded. (closes issue AST-977) Reported-by: John
Bigelow ........ Merged revisions 372354 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372358 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372365 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/pbx.c, /: Ensure counts generated in
manager_show_dialplan_helper are correct When
manager_show_dialplan_helper was written, the counter increment
for the total number of contexts was placed with the extensions
increment instead of in the enclosing loop. This function should
now generate correct context counts. (closes issue AST-970)
Reported-by: John Bigelow ........ Merged revisions 372337 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372338 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372340 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-05 18:56 +0000 [r372343] Alec L Davis <sivad.a@paradise.net.nz>
* /, main/dsp.c: dsp.c: in ast_mf_detect_init incorrectly sets
goertzel samples to 160, should be MF_GSIZE Remove unused
goertzel_state_t member 'samples'. Related
https://reviewboard.asterisk.org/r/2097/
2012-09-05 17:38 +0000 [r372329] Richard Mudgett <rmudgett@digium.com>
* res/res_rtp_asterisk.c, /: Multiple revisions 372327-372328
........ r372327 | rmudgett | 2012-09-05 12:33:11 -0500 (Wed, 05
Sep 2012) | 15 lines Fix RTP/RTCP read error message confusion.
The RTP/RTCP read error message can report "fail: success" when
the read failure is because of an ICE failure. * Changed
__rtp_recvfrom() to generate a PJ ICE message when ICE fails. *
Changed RTP/RTCP read error message to indicate an unspecified
error when errno is zero. (closes issue ASTERISK-20288) Reported
by: Joern Krebs Patches: jira_asterisk_20288_err_msg.patch
(license #5621) patch uploaded by rmudgett (modified) ........
r372328 | rmudgett | 2012-09-05 12:35:20 -0500 (Wed, 05 Sep 2012)
| 1 line Fix coding guidelines issue with a recent commit.
........ Merged revisions 372327-372328 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-05 16:24 +0000 [r372310-372319] Mark Michelson <mmichelson@digium.com>
* main/rtp_engine.c, /, include/asterisk/rtp_engine.h,
res/res_rtp_asterisk.c: Re-fix sending unnegotiated payloads
during a P2P RTP bridge. The previous fix still would look in the
static_RTP_PT table, which is inappropriate since we specifically
want to find a codec that has been negotiated. (closes issue
ASTERISK-20296) reported by NITESH BANSAL Patches:
codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
........ Merged revisions 372311 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/app_alarmreceiver.c: Add fixes and cleanup to
app_alarmreceiver. This work comes courtesy of Pedro Kiefer
(License #6407) The work was posted to review board by Kaloyan
Kovachev (License #5506) (closes issue ASTERISK-16668) Reported
by Grant Crawshay (closes issue ASTERISK-16694) Reported by Fred
van Lieshout (closes issue ASTERISK-18417) Reported by Kostas
Liakakis (closes issue ASTERISK-19435) Reported by Deon George
(closes issue ASTERISK-20157) Reported by Pedro Kiefer (closes
issue ASTERISK-20158) Reported by Pedro Kiefer (closes issue
ASTERISK-20224) Reported by Pedro Kiefer Review:
https://reviewboard.asterisk.org/r/2075
2012-09-05 14:44 +0000 [r372302] Matthew Jordan <mjordan@digium.com>
* /, apps/app_voicemail.c: Fix memory leaks in app_voicemail when
using IMAP storage or realtime config This patch fixes two memory
leaks: 1. When find_user is called with NULL as its first
parameter, the voicemail user returned is allocated on the heap.
The inboxcount2 function uses find_user in such a fashion when
counting new messages, and fails to free the resulting voicemail
user object. 2. When populate_defaults is called on a voicemail
user, it wipes whatever flags have been set on the object by
copying over the global flags object. If the VM_ALLOCED flag was
ste on the voicemail user prior to doing so, that flag is
removed. This leaks the voicemail user when free_user is later
called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
Patch slightly modified for this commit. Review:
https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 372288 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372289 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-05 14:12 +0000 [r372290] Darren Sessions <dmsessions@gmail.com>
* channels/chan_sip.c, configs/res_ldap.conf.sample: LDAP Realtime
Peers Cannot Register Prior to 1.8, it was not necessary for an
explicit "type" to be set for an asterisk LDAP realtime peer. Now
the routine find_peer actually checks the type field during
registration and fails to find the peer if it is not set. The
attached patch makes the realtime type equal whatever type is
being searched for if the type is 0 upon return from routine
build_peer. (closes issue ASTERISK-17222) Reported by: John
Covert Patch by: David Vossel Tested by: Darren Sessions Review:
https://reviewboard.asterisk.org/r/2095/
2012-09-05 12:18 +0000 [r372267] Michael L. Young <elgueromexicano@gmail.com>
* res/res_rtp_asterisk.c, /: Fix breakage caused by last merge.
Missing a variable for 11 and trunk. ........ Merged revisions
372266 from http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-05 07:43 +0000 [r372215-372242] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c, /: dsp.c: Fix multiple issues when no-interdigit
delay is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss
detector to original -r349249 method with some changes, remove
unnecessary; 1. reseting of hits=0, when no signal, only need to
set it once. 2. incrementing of hits, when the hit is the same as
the current hit. 3. setting of lasthit, when it's the same as
before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
spelling mistakes (closes issue ASTERISK-19610) alecdavis
(license 585) Reported by: Jean-Philippe Lord Tested by:
alecdavis Review: https://reviewboard.asterisk.org/r/2085/
........ Merged revisions 372239 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372240 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372241 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/dsp.c: dsp.c: optimize goerztzel sample loops, in
dtmf_detect, mf_detect and tone_detect use a temporary short int
when repeatedly used to call goertzel_sample. alecdavis (license
585) Reported by: alecdavis Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/2093/ ........ Merged
revisions 372212 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372213 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372214 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-05 04:55 +0000 [r372200] Michael L. Young <elgueromexicano@gmail.com>
* res/res_rtp_asterisk.c, /: Fix Incrementing Sequence Number For
Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
place to increment the sequence number for retransmitted DTMF end
packets. With the introduction of the RTP engine API in 1.8, the
sequence number was no longer being incremented. This patch fixes
this regression as well as cleans up a few lines that were not
doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
Bansal Tested by: Michael L. Young Patches:
01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2083/ ........ Merged
revisions 372185 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372198 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372199 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-05 02:26 +0000 [r372176] Matthew Jordan <mjordan@digium.com>
* cel/cel_pgsql.c, /: Fix memory leak when CEL is successfully
written to PostgreSQL database PQClear is not called when the
result object of a call to PQExec has a status of
PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
handled properly, so this memory leak only occurred when CEL
records were successfully written. This patch properly clears the
result in the nominal code path. (closes issue ASTERISK-19991)
Reported by: Etienne Lessard Tested by: Etienne Lessard patches:
mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
#6394) ........ Merged revisions 372158 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372165 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372175 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-09-04 19:30 +0000 [r372148-372149] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt: app_queue: PAUSEALL/UNPAUSEALL logged only if
interface is a queue member Adding UPGRADE.txt entry for r372148
(issue AST-946) Reported by: John Bigelow
* CHANGES, apps/app_queue.c: app_queue: Only log
PAUSEALL/UNPAUSEALL when 1+ memebers changed. Prior to this
patch, if pause or unpause was issued on an interface without
specifying a specific queue, a PAUSEALL or UNPAUSEALL event would
be logged in the queue log even if that interface wasn't a member
of any queues. This patch changes it so that these events are
only logged when at least one member of any queue exists for that
interface. (closes issue AST-946) Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2079/
2012-09-04 15:50 +0000 [r372136-372138] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix issue where SIP devices were not
notified when custom devices changed to "ringing". The problem
had to do with logic used when checking for what the oldest
ringing channel was. The problem was that if no channel was
found, then no notification would be sent. For custom device
states, there is no associated channel, so no notification would
get sent. This fixes the issue by still sending the notification
even if no associated channel can be found for a ringing device
state change. (closes issue ASTERISK-20297) Reported by Noah
Engelberth ........ Merged revisions 372137 from
http://svn.asterisk.org/svn/asterisk/branches/11
* apps/app_confbridge.c, /, main/config_options.c: Prevent crash
from using app_page with no confbridge.conf file provided. Also
prevents other potential crashes when using aco API with
uninitialized aco_info structs. (closes issue ASTERISK-20305)
reported by Noah Engelberth Tested by Noah Engelberth Review:
https://reviewboard.asterisk.org/r/2086 ........ Merged revisions
372135 from http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-31 21:15 +0000 [r372119] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c, /: Prevent local RTP bridges from sending
inappropriate formats to participants. A change for Asterisk 11
caused a check for failure to incorrectly check the return value.
This resulted in the possibility of transmitting media that a
party had not negotiated. If this media happened to be G.729,
then this could potentially result in one-way audio if no G.729
translators are installed. (closes issue ASTERISK-20296) reported
by NITESH BANSAL ........ Merged revisions 372118 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-30 20:54 +0000 [r372051-372092] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Prevent crash on shutdown due to refcount
error on queues container. When app_queue is unloaded, the queues
container has its refcount decremented, potentially to 0. Then
the taskprocessor responsible for handling device state changes
is unreferenced. If the taskprocessor happens to be just about to
run its task, then it will create and destroy an iterator on the
queues container. This can cause the refcount on the queues
container to increase to 1 and then back to 0. Going back to 0 a
second time results in double frees. This failure was seen
periodically in the testsuite when Asterisk would shut down.
........ Merged revisions 372089 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 372090 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372091 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, apps/app_queue.c: Help prevent ringing queue members from
being rung when ringinuse set to no. Queue member status would
not always get updated properly when the member was called, thus
resulting in the member getting multiple calls. With this change,
we update the member's status at the time of calling, and we also
check to make sure the member is still available to take the call
before placing an outbound call. (closes issue ASTERISK-16115)
reported by nik600 Patches: app_queue.c-svn-r370418.patch
uploaded by Italo Rossi (license #6409) ........ Merged revisions
372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 372049 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372050 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-30 16:25 +0000 [r371964-372029] Matthew Jordan <mjordan@digium.com>
* channels/chan_iax2.c, /: AST-2012-013: Resolve ACL rules being
ignored during calls by some IAX2 peers When an IAX2 call is made
using the credentials of a peer defined in a dynamic Asterisk
Realtime Architecture (ARA) backend, the ACL rules for that peer
are not applied to the call attempt. This allows for a remote
attacker who is aware of a peer's credentials to bypass the ACL
rules set for that peer. This patch ensures that the ACLs are
applied for all peers, regardless of their storage mechanism.
(closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
mjordan, Alan Frisch ........ Merged revisions 372028 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/manager.c, /, README-SERIOUSLY.bestpractices.txt:
AST-2012-012: Resolve AMI User Unauthorized Shell Access through
ExternalIVR The AMI Originate action can allow a remote user to
specify information that can be used to execute shell commands on
the system hosting Asterisk. This can result in an unwanted
escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to
perform actions that would typically require the "system" class
authorization. Previous attempts to prevent this permission
escalation (AST-2011-006, AST-2012-004) have sought to do so by
inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched
a predefined set of values, rejecting the command if the user
lacked the "system" class authorization. As noted by IBM X-Force
Research, the "ExternalIVR" application is not listed in the
predefined set of values. The solution for this particular
vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class
authorization. Unfortunately, the approach of inspecting fields
in the Originate action against known applications/functions has
a significant flaw. The predefined set of values can be bypassed
by creative use of the Originate action or by certain dialplan
configurations, which is beyond the ability of Asterisk to
analyze at run-time. Attempting to work around these scenarios
would result in severely restricting the applications or
functions and prevent their usage for legitimate means. As such,
any additional security vulnerabilities, where an
application/function that would normally require the "system"
class authorization can be executed by users with the "originate"
class authorization, will not be addressed. Instead, the
README-SERIOUSLY.bestpractices.txt file has been updated to
reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper
system configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
X-Force Research ........ Merged revisions 371998 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371999 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 372000 from
http://svn.asterisk.org/svn/asterisk/branches/11
* include/asterisk/bridging.h, include/asterisk/datastore.h,
main/file.c, include/asterisk/strings.h, include/asterisk/pbx.h,
channels/sip/include/srtp.h, main/audiohook.c,
include/asterisk/translate.h, main/cdr.c, main/channel.c,
include/asterisk/crypto.h, include/asterisk/config_options.h,
include/asterisk/bridging_technology.h,
include/asterisk/audiohook.h,
apps/confbridge/include/confbridge.h, include/asterisk/format.h,
include/asterisk/netsock2.h, include/asterisk/rtp_engine.h,
include/asterisk/ccss.h, main/pbx.c, include/asterisk/utils.h,
channels/sip/srtp.c, channels/chan_sip.c,
include/asterisk/format_pref.h, include/asterisk/astobj2.h,
include/asterisk/presencestate.h, channels/chan_agent.c,
include/asterisk/config.h, pbx/pbx_lua.c,
formats/format_ogg_vorbis.c, include/asterisk/channel.h,
main/named_acl.c, codecs/speex/speex_resampler.h,
include/asterisk/manager.h, include/asterisk/format_cap.h,
include/asterisk/framehook.h, include/asterisk/heap.h,
channels/sig_pri.h, Makefile, include/asterisk/message.h: Clean
up doxygen warnings This patch fixes numerous doxygen warnings
across Asterisk. It also updates the makefile to regenerate the
doxygen configuration on the local system before running doxygen
to help prevent warnings/errors on the local system. Much thanks
to Andrew for tackling one of the Asterisk janitor projects!
(issue ASTERISK-20259) Reported by: Andrew Latham Patches:
doxygen_partial.diff uploaded by Andrew Latham (license 5985)
make_progdocs.diff uploaded by Andrew Latham (license 5985)
* doc/CODING-GUIDELINES (added), /: Restore CODING-GUIDELINES to
doc folder In r294740, the CODING-GUIDELINES was removed from the
doc folder in favor of the content on the Asterisk wiki. Some
folks still look in the doc folder initially for coding guideline
suggestions; as such, this patch adds a CODING-GUIDELINES file
back into the doc folder. The content of the file merely points
to the correct page on the Asterisk wiki where the coding
guidelines currently live. (closes issue ASTERISK-20279) Reported
by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
Andrew Latham (license 5985) ........ Merged revisions 371961
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 371962 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371963 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-29 22:48 +0000 [r371951-371952] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/md5.h: Ensure alignment of in[] field in
MD5Context struct. The struct MD5Context character buffer is cast
to an int32_t* without making sure that said buffer is aligned.
Since the buffer follows two uint32_t's, the chance of 'in' being
(32 bits) unaligned is nil in practice. But adding code to ensure
that 'in' stays aligned costs nothing and removes all doubts
about the casts being safe. (closes issue ASTERISK-20241)
Reported by: Walter Doekes Patches: tmp.diff (license #5674)
patch uploaded by Walter Doekes
* /, apps/app_meetme.c: Fix compile errors. ........ Merged
revisions 371950 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-29 21:15 +0000 [r371922] Jonathan Rose <jrose@digium.com>
* /, apps/app_meetme.c: app_meetme: Adding test events for
following activity in MeetMe. ........ Merged revisions 371919
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 371920 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371921 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-29 19:57 +0000 [r371892-371894] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Fix theoretical compile error with HAVE_EPOLL.
Really shows how much epoll is used since it had not been
reported yet. ........ Merged revisions 371893 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/channel.c, /: Initialize file descriptors for dummy channels
to -1. Dummy channels usually aren't read from, but functions
like SHELL and CURL use autoservice on the channel. (closes issue
ASTERISK-20283) Reported by: Gareth Palmer Patches:
svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
(modified) ........ Merged revisions 371888 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371890 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371891 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-29 19:38 +0000 [r371889] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, UPGRADE.txt: chan_sip: Change manager event
to confirm SIPqualifypeer into an ack Matt Jordan informed me
that it was more appropriate to use an astman_send_ack here
instead of making an event response. I've also used this
opportunity to update UPGRADE.txt to mention this change in
behavior. (issue AST-969) Reported by: John Bigelow
2012-08-29 18:40 +0000 [r371863] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, /: Fix hangup cause passthrough regression. The
v1.8 -r369258 change to fix the F and F(x) action logic
introduced a regression in passing the hangup cause from the
called channel to the caller channel. (closes issue
ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by
Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged
revisions 371860 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371861 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371862 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-29 17:35 +0000 [r371823-371851] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
Doekes ........ Merged revisions 371824 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371825 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371845 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_sip.c: chan_sip: Send a manager event to confirm
SIPqualifypeer completes Prior to this patch, Issuing
SIPqualifypeer either resulted in an error or if it succeeded, a
few \r\ns. This patch adds a SIPqualifypeerComplete event issued
as a response when the command is successfully executed. (closes
issue AST-969) Reported by: John Bigelow
2012-08-27 21:51 +0000 [r371785-371791] Mark Michelson <mmichelson@digium.com>
* configs/agents.conf.sample, /: Fix misleading documentation in
agents.conf.sample regarding ackcall usage. The documentation
made it sound as if the DTMF acknowledgment was needed at the
time the agent logs in, rather than when the agent is called.
This is likely a relic from the days when there were multiple
ways of logging in agents. (closes issue AST-962) reported by
Steve Pitts ........ Merged revisions 371787 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371789 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371790 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/manager.c, /: Fix incorrect documentation of the
MailboxStatus manager command. The "Waiting" field was
misdocumented as reporting the number of messages waiting. In
reality, it simply indicated the presence or absence of waiting
messages. ........ Merged revisions 371782 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371783 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371784 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-27 18:16 +0000 [r371754] David M. Lee <dlee@digium.com>
* res/pjproject/pjlib-util/bin, res/pjproject/pjnath/build/output,
/, res/pjproject/pjlib/bin,
res/pjproject/pjlib-util/build/output, res/pjproject/pjnath/bin,
res/pjproject/pjlib/build/output: svn:ignore pjproject bin &
output for all platforms. ........ Merged revisions 371753 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-27 17:52 +0000 [r371751] Mark Michelson <mmichelson@digium.com>
* /, configs/queues.conf.sample: Fix incorrectly documented option
in queues.conf sharedlastcall defaults to "no" not "yes" (closes
issue AST-979) reported by Steve Pitts ........ Merged revisions
371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 371748 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371750 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-27 16:56 +0000 [r371721] David M. Lee <dlee@digium.com>
* main/lock.c, /: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
variants. The original implementations simply wrap pthread
functions, which take absolute time as an argument. The spinlock
version for systems without those functions treated the argument
as a delta. This patch fixes the spinlock version to be
consistent with the pthread version. (closes issue
ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
uploaded by Egor Gorlin (license 6416) ........ Merged revisions
371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 371720 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-27 14:13 +0000 [r371693] Kinsey Moore <kmoore@digium.com>
* /, main/utils.c: Implement workaround for BETTER_BACKTRACES crash
When compiling with BETTER_BACKTRACES enabled, Asterisk will
sometimes crash when "core show locks" is run. This happens
regularly in the testsuite since several tests run "core show
locks" to help with debugging. This seems to be a fault with
libraries on certain operating systems (notably CentOS 6.2/6.3)
running on virtual machines and utilizing gcc 4.4.6. (closes
issue ASTERISK-20090) ........ Merged revisions 371690 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371691 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371692 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-26 23:10 +0000 [r371665] Alec L Davis <sivad.a@paradise.net.nz>
* /, main/dsp.c: mf_detect: incorrectly used DTMF_GSIZE instead of
MF_GSIZE ........ Merged revisions 371662 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371663 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371664 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-23 04:12 +0000 [r371633] Mark Michelson <mmichelson@digium.com>
* tests/test_scoped_lock.c (added): I forgot to add the unit tests
for scoped locks earlier today.
2012-08-22 15:55 +0000 [r371620] Joshua Colp <jcolp@digium.com>
* /, channels/chan_motif.c: Add support for call-id logging to
chan_motif. Review: https://reviewboard.asterisk.org/r/2077/
........ Merged revisions 371619 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-21 21:01 +0000 [r371572-371593] Mark Michelson <mmichelson@digium.com>
* cdr/cdr_tds.c, main/xmldoc.c, apps/app_dial.c,
channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c,
main/file.c, main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
res/res_config_sqlite.c: Fix misuses of asprintf throughout the
code. This fixes three main issues * Change asprintf() uses to
ast_asprintf() so that it pairs properly with ast_free() and no
longer causes MALLOC_DEBUG to freak out. * When ast_asprintf()
fails, set the pointer NULL if it will be referenced later. * Fix
some memory leaks that were spotted while taking care of the
first two points. (Closes issue ASTERISK-20135) reported by
Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071
........ Merged revisions 371590 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371591 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371592 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/config.c, include/asterisk/lock.h: Add scoped locks to
Asterisk. With the SCOPED_LOCK macro, you can create a variable
that locks a specific lock and unlocks the lock when the variable
goes out of scope. This is useful for situations where many
breaks, continues, returns, or other interruptions would require
separate unlock statements. With a scoped lock, these aren't
necessary. There are specializations for mutexes, read locks,
write locks, ao2 locks, ao2 read locks, ao2 write locks, and
channel locks. Each of these is a SCOPED_LOCK at heart though.
Review: https://reviewboard.asterisk.org/r/2060
* /, res/res_rtp_asterisk.c: Use thread-local storage to store
pj_thread_descs. pj_thread_register() takes a parameter of type
pj_thread_desc. It was assumed that pj_thread_register either
used this item temporarily or made a copy of it. Unfortunately,
all it does is keep a pointer to the structure in thread-local
storage. This means that if our pj_thread_desc goes out of scope,
then pjlib will be referencing bogus data quite often, most
commonly on operations involving a pj_mutex_t. In our case, our
pj_thread_desc was on the stack and went out of scope very
shortly after registering our thread with pjlib. With this
change, the pj_thread_desc is stored in thread-local storage so
the pointer that pjlib keeps in thread-local storage will
reference legitimate memory. (closes issue ASTERISK-20237)
reported by Jeremy Pepper Patches: ASTERISK-20237.patch uploaded
by Mark Michelson (license #5049) Tested by Jeremy Pepper
........ Merged revisions 371571 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-20 15:39 +0000 [r371535-371547] Kinsey Moore <kmoore@digium.com>
* main/udptl.c, /: Ignore recovered zero-length secondary UDPTL
packets In some cases, recovering lost packets using the
secondary packet recovery mechanism with UDPTL/T.38 can result in
the recovery of zero-length packets. These must be ignored or the
frame generated from them can cause segfaults and allocation
failures. (closes issue ASTERISK-19762) (closes issue
ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
Gagnon (rgagnon) ........ Merged revisions 371544 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371545 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371546 from
http://svn.asterisk.org/svn/asterisk/branches/11
* main/utils.c: Fix for commit r371535
* main/utils.c: Apply work-around for BETTER_BACKTRACES crash When
compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in
the testsuite since several tests run "core show locks" to help
with debugging. This seems to be a fault with libraries on
certain operating systems (notably CentOS 6.2/6.3) running on
virtual machines and utilizing gcc 4.4.6. (issue ASTERISK-20090)
2012-08-18 02:09 +0000 [r371493-371521] Matthew Jordan <mjordan@digium.com>
* main/http.c, /: Remove old debug code from http configuration
loading (closes issue ASTERISK-20254) Reported by: Andrew Latham
Patches: http.diff uploaded by Andrew Latham (license #5985)
........ Merged revisions 371520 from
http://svn.asterisk.org/svn/asterisk/branches/11
* res/res_xmpp.c, /: Fix typo in JabberSend that looked for '2'
instead of '@' in recipient argument The summary says about all
there is to say. (closes issue ASTERISK-20239) Reported by:
Gregory Porras ........ Merged revisions 371518 from
http://svn.asterisk.org/svn/asterisk/branches/11
* funcs/func_hangupcause.c, /: Make the name of the
"HangupCauseClear" application consistent The name of the
"HangupCauseClear" application is "HangupCauseClear", not
"HangupcauseClear". The incorrect case of 'cause' caused the XML
documentation to not register properly. As an aside, this commit
message felt very awkward, but I'm not sure how else to note that
"X", which has to be "X", was referred to as "x". (closes issue
ASTERISK-20253) Reported by: Andrew Latham Patches:
hangupcause.diff uploaded by Andrew Latham (license #5985)
........ Merged revisions 371516 from
http://svn.asterisk.org/svn/asterisk/branches/11
* sounds/sounds.xml, res/res_curl.c, build_tools/cflags.xml,
utils/utils.xml, /, res/res_fax.c: Update module support level on
a variety of modules and compiler options Some core support
modules and compiler options were no longer tagged with a module
support level. This patch adds 'core' back to those options. Note
that this patch modifies a few of the patches provided by Andrew
Latham slightly. res_curl and res_fax are both 'core' supported
modules. (closes issue ASTERISK-20215) Reported by: Andrew Latham
Tested by: mjordan Patches: astcanary.diff (license #5985)
uploaded by Andrew Latham cflagsxml.diff (license #5985) uploaded
by Andrew Latham curl_fax.diff (license #5985) uploaded by Andrew
Latham soundsxml.diff (license #5985) uploaded by Andrew Latham
........ Merged revisions 371507 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, main/xmldoc.c: Fix memory leak in XML documentation When
formatting documentation fields, the XML documentation parser
calls xmldoc_get_formatted. This function allocates a string
buffer at the beginning of its routine. Unfortunately, on certain
code paths, it also calls xmldoc_string_cleanup, which assumes
that it will create the string buffer. The previously allocated
string buffer is then leaked by the xmldoc_string_cleanup
routine. Now: we don't do that. (closes issue AST-932) Reported
by: Alexander Homig ........ Merged revisions 371469 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371491 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371492 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-17 19:50 +0000 [r371483] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: When a peer registers using WebSocket do
not resolve the Contact provided. (closes issue ASTERISK-20238)
Reported by: james.mortensen ........ Merged revisions 371482
from http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-17 16:01 +0000 [r371439] Kinsey Moore <kmoore@digium.com>
* main/loader.c, /: Add instrumentation to subsystem reloads When
Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
generate TestEvent AMI events on subsystem reloads such as cdr,
dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions
371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 371437 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371438 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-17 12:42 +0000 [r371428] Russell Bryant <russell@russellbryant.com>
* res/res_rtp_asterisk.c, /: rtp: Ensure defaults are set without
rtp.conf. While building up a new install to test chan_motif, I
ran into a failure due to icesupport being disabled. This was due
to me not having an rtp.conf. It was intended in the code for it
to be enabled by default, but it was only applied if rtp.conf
existed. This patch updates res_rtp_asterisk to be consistent in
how it handles defaults. A few options didn't have their default
values set globally, including icesupport. They are now set and
icesupport is enabled by default, even if you do not have an
rtp.conf. ........ Merged revisions 371425 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-17 12:25 +0000 [r371427] Joshua Colp <jcolp@digium.com>
* res/res_format_attr_h264.c, /: Add some additional H.264
attributes, "max-smbps" and "max-fps", for passthrough. (closes
issue ASTERISK-20206) Reported by: ddkprog Patches:
res_format_attr_h264.c.diff uploaded by ddkprog (license 6008)
........ Merged revisions 371426 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-16 23:08 +0000 [r371400] Terry Wilson <twilson@digium.com>
* /, main/config.c: Handle integer over/under-flow in
ast_parse_args The strtol family of functions will return
*_MIN/*_MAX on overflow. To detect when an overflow has happened,
errno must be set to 0 before calling the function, then checked
afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged
revisions 371392 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371398 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371399 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-16 22:45 +0000 [r371396] Kinsey Moore <kmoore@digium.com>
* /, main/loader.c: Add module reload instrumentation for
TEST_FRAMEWORK This adds AMI events for module reloads when
Asterisk is built with TEST_FRAMEWORK enabled and corrects
generation of the module load AMI event. (issue PQ-1126) ........
Merged revisions 371393 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371394 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371395 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-16 19:52 +0000 [r371356-371383] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable
to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING
flag was used instead, which will frequently flip during
reinvites. (closes issue AST-897) Reported by: Thomas Arimont
........ Merged revisions 371357 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371358 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371382 from
http://svn.asterisk.org/svn/asterisk/branches/11
* /, channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP
answer is included in the SIP ACK Under certain conditions, a SIP
transaction involving directmedia wouldn't trigger a re-invite
because the SDP answer was included in an ACK instead of in a
message that we would have triggered the invite with. This patch
just queues a source change control frame if the dialog is using
directmedia when we find sdp for an ACK. (closes issue AST-913)
Reported by: Thomas Arimont ........ Merged revisions 371337 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371338 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371355 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-15 23:35 +0000 [r371325] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Fix bug where final queue member would not
be removed from memory. If a static queue had realtime members,
then there could be a potential for those realtime members not to
be properly deleted from memory. If the queue's members were
loaded from realtime and then all the members were deleted from
the backend, then the queue would still think these members
existed. The reason was that there was a short- circuit in code
such that if there were no members found in the backend, then the
queue would not be updated to reflect this. Note that this only
affected static queues with realtime members. Realtime queues
with realtime members were unaffected by this issue. (closes
issue ASTERISK-19793) reported by Marcus Haas ........ Merged
revisions 371306 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371313 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371324 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-15 20:43 +0000 [r371296] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix Segfault When Registering SIP Over
WebSockets The helper function, get_address_family_filter, in
chan_sip for dns resolution by address family was not recognizing
the websockets transport and resulting in a null pointer being
sent to functions in netsock2, in an attempt to determine if we
are bound to ANY address ([::]) or not. This patch fixes this
issue by handling the transport types SIP_TRANSPORT_WS and
SIP_TRANSPORT_WSS which results in a sock address being set
properly for use in determining the address family. (closes issue
ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven
Beisiegel, James Mortensen Patches:
asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young
(license 5026) ........ Merged revisions 371295 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-15 20:18 +0000 [r371259-371277] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
relatedpeer on SIP dialog destruction The other instance of this
bug was fixed by jcolp/file in r121496. If we are destroying a
dialog only set the MWI dialog pointer on the related peer to
NULL if it is the dialog currently being destroyed. (closes issue
ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged
revisions 371270 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371271 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371272 from
http://svn.asterisk.org/svn/asterisk/branches/11
* channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c,
channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/chan_sip.c: Add HANGUPCAUSE information to callee
channels This adds HANGUPCAUSE information to called channels so
that hangup handlers can, in conjunction with predial dialplan
execution, access the hangupcause information when the dialed
channel hangs up on a one-to-one basis instead of a many-to-one
basis as with HANGUPCAUSE usage on the caller channel. Review:
https://reviewboard.asterisk.org/r/2069/ (closes issue
ASTERISK-20198) ........ Merged revisions 371258 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-13 20:36 +0000 [r371228] Kinsey Moore <kmoore@digium.com>
* main/loader.c, /, apps/app_meetme.c: Add test instrumentation
This adds test instrumentation for loading and unloading of
modules and for certain actions in MeetMe to be used in the
testsuite or any other consumer of AMI events. These will only be
generated when Asterisk is built with TEST_FRAMEWORK enabled.
(issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 371203 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371227 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-13 20:02 +0000 [r371202] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix problem where incorrect pointer was
checked for nullity. ........ Merged revisions 371198 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371199 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371200 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-11 19:13 +0000 [r371170] Matthew Jordan <mjordan@digium.com>
* UPGRADE-11.txt (added), UPGRADE.txt: Add UPGRADE-11.txt file;
update UPGRADE.txt to reflect Asterisk 12
2012-08-10 22:04 +0000 [r371147] Richard Mudgett <rmudgett@digium.com>
* /, CHANGES: Update CHANGES for private party ID. ........ Merged
revisions 371146 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-10 21:35 +0000 [r371144] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Fix a couple of documentation problems in
app_queue.c * The RemoveQueueMember app made mention of options
that could be passed in, but no options are supported. I have
removed the listing of options from the documentation. * The
RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
that could be set. (closes issue AST-949) reported by Steve Pitts
(closes issue AST-954) reported by Steve Pitts ........ Merged
revisions 371141 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371142 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
revisions 371143 from
http://svn.asterisk.org/svn/asterisk/branches/11
2012-08-10 21:09 +0000 [r371134] Matthew Jordan <mjordan@digium.com>
* /: Remove 10 properties, add 11 properties
2012-08-10 19:54 +0000 [r371120] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, channels/sig_pri.c,
funcs/func_callerid.c, main/cli.c, main/channel.c,
channels/chan_misdn.c, channels/chan_sip.c,
main/channel_internal_api.c, main/features.c: Add private
representation of caller, connected and redirecting party ids.
This patch adds the feature "Private representation of caller,
connected and redirecting party ids", as previously discussed
with us (DATUS) and Digium. 1. Feature motivation Until now it is
quite difficult to modify a party number or name which can only
be seen by exactly one particular instantiated technology channel
subscriber. One example where a modified party number or name on
one channel is spread over several channels are supplementary
services like call transfer or pickup. To implement these
features Asterisk internally copies caller and connected ids from
one channel to another. Another example are extension
subscriptions. The monitoring entities (watchers) are notified of
state changes and - if desired - of party numbers or names which
represent the involving call parties. One major feature where a
private representation of party names is essentially needed, i.e.
where a party name shall be exclusively signaled to only one
particular user, is a private user-specific name resolution for
party numbers. A lookup in a private destination-dependent
telephone book shall provide party names which cannot be seen by
any other user at any time. 2. Feature Description This feature
comes along with the implementation of additional private party
id elements for caller id, connected id and redirecting ids
inside Asterisk channels. The private party id elements can be
read or set by the user using Asterisk dialplan functions. When a
technology channel is initiating a call, receives an internal
connected-line update event, or receives an internal redirecting
update event, it merges the corresponding public id with the
private id to create an effective party id. The effective party
id is then used for protocol signaling. The channel technologies
which initially support the private id representation with this
patch are SIP (chan_sip), mISDN (chan_misdn) and PRI
(chan_dahdi). Once a private name or number on a channel is set
and (implicitly) made valid, it is generally used for any further
protocol signaling until it is rewritten or invalidated. To
simplify the invalidation of private ids all internally generated
connected/redirecting update events and also all
connected/redirecting update events which are generated by
technology channels -- receiving regarding protocol information -
automatically trigger the invalidation of private ids. If not
using the private party id representation feature at all, i.e. if
using only the 'regular' caller-id, connected and redirecting
related functions, the current characteristic of Asterisk is not
affected by the new extended functionality. 3. User interface
Description To grant access to the private name and number
representation from the Asterisk dialplan, the CALLERID,
CONNECTEDLINE and REDIRECTING dialplan functions are extended by
the following data types. The formats of these data types are
equal to the corresponding regular 'non-private' already existing
data types: CALLERID: priv-all priv-name priv-name-valid
priv-name-charset priv-name-pres priv-num priv-num-valid
priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid
priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE:
priv-name priv-name-valid priv-name-pres priv-name-charset
priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr
priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag
REDIRECTING: priv-orig-name priv-orig-name-valid
priv-orig-name-pres priv-orig-name-charset priv-orig-num
priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type
priv-orig-subaddr-odd priv-orig-tag priv-from-name
priv-from-name-valid priv-from-name-pres priv-from-name-charset
priv-from-num priv-from-num-valid priv-from-num-pres
priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid
priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag
priv-to-name priv-to-name-valid priv-to-name-pres
priv-to-name-charset priv-to-num priv-to-num-valid
priv-to-num-pres priv-to-num-plan priv-to-subaddr
priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
priv-to-tag Reported by: Thomas Arimont Review:
https://reviewboard.asterisk.org/r/2030/
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