Commit Graph

28998 Commits

Author SHA1 Message Date
Jenkins2
cad74cdd8f Merge "core: Fix segfault when invoking 'data get' CLI command" into 13 2017-07-05 18:29:28 -05:00
Jenkins2
dc1fc28a4a Merge "pjproject_bundled: Allow passing configure options to bundled" into 13 2017-07-05 17:39:56 -05:00
George Joseph
642c597507 Merge "pjsip_distributor.c: Fix deadlock with TCP type transports." into 13 2017-07-05 16:08:18 -05:00
Jenkins2
cf6e0b8f8b Merge "chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support)." into 13 2017-07-05 16:06:44 -05:00
Jenkins2
aadece5664 Merge "pjsip_distributor.c: Fix unidentified_requests hash functions." into 13 2017-07-05 15:07:51 -05:00
Jenkins2
d55798fff0 Merge "bridge_native_rtp: Keep rtp instance refs on bridge_channel" into 13 2017-07-05 15:02:28 -05:00
George Joseph
40490768cc Merge "chan_pjsip: Fix ability to send UPDATE on COLP" into 13 2017-07-05 14:38:01 -05:00
Jenkins2
2bbe8cb3f5 Merge "channel: Clear channel flag in error branch." into 13 2017-07-05 08:56:46 -05:00
Sean Bright
6258de458b core: Fix segfault when invoking 'data get' CLI command
Invoking 'data get /asterisk/core/channeltypes' caused a crash because
of an assumption of a tech's capabilities to be non-NULL. The
'Surrogate' tech, however, does have a NULL capabilities member,
resulting in a crash.

ASTERISK-27108 #close

Change-Id: I2fbe7715681f43d5565d1e1599269468c26b0e0a
2017-07-05 08:42:07 -04:00
Alexander Traud
39d2ebbf56 chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).
When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was
added in any case, because of a local Boolean-negation error of the return value
of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was
still always added with tlsenable=yes, because the domains were not compared
just on the address but also on the port – and TLS is always on a different port
than UDP/TCP.

ASTERISK-27106

Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c
2017-07-03 11:01:38 -05:00
Alexander Traud
9f4b3b966e chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).
Because of a copy-and-paste error when the struct ast_sockaddr changed,
tlsbindaddr was not added, when sip.conf contained autodomain=yes; see
"show sip domains" on the command-line interface (CLI) of Asterisk.

ASTERISK-27106

Change-Id: I3d0957150017c223136968ef1266f275d0d6695e
2017-07-03 10:53:03 -05:00
Corey Farrell
73520e9f58 channel: Clear channel flag in error branch.
Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when
ast_read returns NULL.

ASTERISK-27100 #close

Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d
2017-07-01 00:57:31 -04:00
Jenkins2
067410a445 Merge "app_queue: Fix returning to dialplan when a queue is empty" into 13 2017-06-30 14:45:43 -05:00
Richard Mudgett
0d64cbde57 pjsip_distributor.c: Fix deadlock with TCP type transports.
When a SIP message comes in on a transport, pjproject obtains the lock on
the transport and pulls the data out of the socket.  Unlike UDP, the TCP
transport does not allow concurrent access.  Without concurrency the
transport lock is not released when the transport's message complete
callback is called.  The processing continues and eventually Asterisk
starts processing the SIP message.  The first thing Asterisk tries to do
is determine the associated dialog of the message to determine the
associated serializer.  To get the associated serializer safely requires
us to get the dialog lock.

To send a request or response message for a dialog, pjproject obtains the
dialog lock and then obtains the transport lock.  Deadlock can result
because of the opposite order the locks are obtained.

* Fix the deadlock by obtaining the serializer associated with the dialog
another way that doesn't involve obtaining the dialog lock.  In this case,
we use an ao2 container to hold the associated endpoint and serializer.
The new locks are held a brief time and won't overlap other existing lock
times.

ASTERISK-27090 #close

Change-Id: I9ed63f4da9649e9db6ed4be29c360968917a89bd
2017-06-30 12:02:24 -05:00
Richard Mudgett
905d18e8bf pjsip_distributor.c: Fix unidentified_requests hash functions.
The OBJ_SEARCH_xxx defines should not be used as if they were individual
bits.  They represent a multi-bit enumeration value field.

Change-Id: I32abc9a475396dab02402a7014357dd94284e17b
2017-06-30 12:00:21 -05:00
Jenkins2
35bc1ee28d Merge "res_rtp_asterisk: Fix issues with ICE renegotiation." into 13 2017-06-30 11:29:29 -05:00
George Joseph
bbe68f139d pjproject_bundled: Allow passing configure options to bundled
There wasn't any good way to pass options like --host or --build
down to the pjproject configure which makes cross-compiling difficult.

* Added a new PJPROJECT_CONFIGURE_OPTS environment variable which
  can be used to pass arbitrary options to pjproject configure.
* Automatically set the pjproject configure --host and --build
  options to match those supplied for the asterisk configure.

ASTERISK-27097 #close
Reported-by: Kinsey Moore

Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e
2017-06-30 07:39:07 -06:00
George Joseph
6bd7c0f37c chan_pjsip: Fix ability to send UPDATE on COLP
When connected_line_method is "invite", we're supposed to determine
if the client can support UPDATE and if it can, send UPDATE instead
of INVITE to avoid the SDP renegotiation.  Not only was pjproject
not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing
that invite_tsx wasn't NULL which isn't always the case.

* Updated chan_pjsip/update_connected_line_information to drop the
  requirement that invite_tsx isn't NULL.
* Submitted patch to pjproject sip_inv.c that sets the
  PJSIP_INV_SUPPORT_UPDATE flag correctly.
* Updated pjsip.conf.sample to clarify what happens when "invite"
  is specified.

ASTERISK-27095

Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560
2017-06-29 14:44:43 -06:00
Ivan Poddubny
2c43ca0ac5 app_queue: Fix returning to dialplan when a queue is empty
The fix for ASTERISK-25665 introduced a regression.
The return value of queue_exec used to be 0 in case of leavewhenempty
but it was changed to -1 (returned from wait_our_turn and passed
transparently by queue_exec), thus leading to hangup instead of returning
back to dialplan.

This commit resets the value back to 0 in this case, restoring
original behavior.

ASTERISK-27065 #close
Reported by: Marek Cervenka

Change-Id: Id9c83b75aeda463250155e88c5004be52bbca5ac
2017-06-29 09:51:50 -05:00
Jenkins2
997c11235e Merge "app_voicemail: IMAP connection control" into 13 2017-06-29 09:03:05 -05:00
Joshua Colp
0426b1d88a res_rtp_asterisk: Fix issues with ICE renegotiation.
When re-inviting to add more streams it is possible for
the role of existing ICE sessions to be changed to the
incorrect value. This results in subsequent refreshes
within the sessions getting a role conflict and the ICE
session breaking down. This change only sets the role to
be the new value if an ICE renegotiation is actually
going to happen, otherwise the existing role is preserved.

As well if we encounter a situation where a unidirectional
ICE negotiation happens and the other side does not send us
candidates we will not store any information for sending
traffic, even though we know where they are reachable. This
change fixes this by using the source of the ICE traffic
itself as the target if no candidates are known and we
receive some ICE traffic.

ASTERISK-27088

Change-Id: I71228181e358917fcefc3100fad21b2fc02a59a9
2017-06-28 14:13:54 +00:00
George Joseph
eb48e99bd4 bridge_native_rtp: Keep rtp instance refs on bridge_channel
There have been reports of deadlocks caused by an attempt to send a frame
to a channel's rtp instance after the channel has left the native bridge
and been destroyed.  This patch effectively causes the bridge channel to
keep a reference to the glue and both the audio and video rtp instances
so what gets started will get stopped.

ASTERISK-26978 #close
Reported-by: Ross Beer

Change-Id: I9e1ac49fa4af68d64826ccccd152593cf8cdb21a
2017-06-27 11:08:19 -05:00
Torrey Searle
1f59d08924 res/res_pjsip_t38: fix incorrect increment of media_count
The T38 sdp callback incorrectly has a side effect of incrementing
the media_count.  This can lead to core dumps.

Change-Id: I7bb2f4987de4046ec52cfc34e5ea0662dae32af8
2017-06-27 17:46:43 +02:00
Torrey Searle
9fbc34d2bd res_pjsip: Add DTMF INFO Failback mode
The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated.  This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.

ASTERISK-27066 #close

Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
2017-06-23 09:15:24 +02:00
Richard Mudgett
764d04fa87 res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer
Change-Id: I0bc97c6608de1d1a4228826b3b3be43f162f05f3
2017-06-21 18:21:57 -05:00
Alexei Gradinari
0f6a9617eb res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact
Do not need to unsubscribe/subscribe on creating the ednpoint's contact.
The modified function create_mwi_subscriptions_for_endpoint adds
the subscription only if it does not exist.

The subscriptions aren't added for active contacts
which are retrieved on startup from realtime
if mwi_disable_initial_unsolicited=yes.
Because the mwi_contact_added is not called.
So the subscriptions also should be created on updating contact.

ASTERISK-26230 #close

Change-Id: I47e265af9296ca09aa42a316fdacac104148cee4
2017-06-21 18:21:57 -05:00
Jenkins2
bd9d72793d Merge "core_local: local channel data not being properly unref'ed and unlocked" into 13 2017-06-21 18:06:44 -05:00
Joshua Colp
7b325d55e1 Merge "bridge: stuck channel(s) after failed attended transfer" into 13 2017-06-21 17:36:54 -05:00
Kevin Harwell
1f9913f272 core_local: local channel data not being properly unref'ed and unlocked
In an earlier version of Asterisk a local channel [un]lock all functions were
added in order to keep a crash from occurring when a channel hung up too early
during an attended transfer. Unfortunately, when a transfer failure occurs and
depending on the timing, the local channels sometime do not get properly
unlocked and deref'ed after being locked and ref'ed. This happens because the
underlying local channel structure gets NULLed out before unlocking.

This patch reworks those [un]lock functions and makes sure the values that get
locked and ref'ed later get unlocked and deref'ed.

ASTERISK-27074 #close

Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09
2017-06-21 16:17:02 -05:00
Kevin Harwell
67664fbf95 bridge: stuck channel(s) after failed attended transfer
If an attended transfer failed it was possible for some of the channels
involved to get "stuck" because Asterisk was not hanging up the transfer target.

This patch ensures Asterisk hangs up the transfer target when an attended
transfer failure occurs.

ASTERISK-27075 #close

Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9
2017-06-21 11:16:47 -05:00
Jenkins2
9554166182 Merge "res_corosync: Change thread stack size" into 13 2017-06-20 18:12:14 -05:00
Jenkins2
2619c04f1e Merge "cdr: fix mistake spelling of a word for Unanswered." into 13 2017-06-20 09:15:58 -05:00
Jenkins2
34b3665612 Merge "res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact" into 13 2017-06-20 05:45:51 -05:00
Rodrigo Ramírez Norambuena
cecf6540dc cdr: fix mistake spelling of a word for Unanswered.
Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df
2017-06-20 04:59:59 -05:00
Alexei Gradinari
8f356192d1 app_voicemail: IMAP connection control
A new global option "imap_poll_logout" was added to specify whether need to
disconnect from the IMAP server after polling of mailboxes.

ASTERISK-27068 #close

Closing IMAP connection after loading mailbox from voicemail.conf

ASTERISK-24052 #close

Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a
2017-06-19 18:21:29 -04:00
Jenkins2
507ce0aa95 Merge "res_stasis: Plug reference leak on stolen channels" into 13 2017-06-19 11:38:02 -05:00
George Joseph
470bccd769 Merge "app_voicemail: IMAP logout on reload/unload" into 13 2017-06-19 09:19:39 -05:00
Jenkins2
707e0e62e6 Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing"" into 13 2017-06-19 08:48:09 -05:00
Alexei Gradinari
8e749c8f51 res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact
If the endpoint's last contact is deleted unsolicited MWI has to be
unsubscribed.

ASTERISK-27051 #close

Change-Id: I33e174e0b9dba0998927d16d6d100fda5c7254e0
2017-06-16 18:45:51 -04:00
Jenkins2
2c87fced4f Merge "formats/format_g729: Fix typo in comment" into 13 2017-06-16 16:16:18 -05:00
Jenkins2
47b9651658 Merge "Core/PBX: Deadlock between dialplan execution and application unregistration." into 13 2017-06-16 16:05:39 -05:00
George Joseph
edfdb4dff5 res_stasis: Plug reference leak on stolen channels
When a stasis channel is stolen by another app, the control
structure is unreffed but never unlinked from the app_controls
container.  This causes the channel reference to leak.

Added OBJ_UNLINK to the callback in channel_stolen_cb.

Also added some additional channel lifecycle debug messages to
channel.c.

ASTERISK-27059 #close
Repoorted-by: George Joseph

Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14
2017-06-16 15:06:56 -05:00
Matthew Fredrickson
0a40073750 formats/format_g729: Fix typo in comment
There was a typo in a comment.  This commit is to fix the typo.

ASTERISK-27060 #close

Change-Id: Ic2699f8dbeaacd58ccb6ec3203e853e1babe3235
2017-06-16 15:00:54 -05:00
Alexei Gradinari
a6e4899612 res_pjsip: New endpoint option "notify_early_inuse_ringing"
This option was added to control whether to notify dialog-info state
'early' or 'confirmed' on Ringing when already INUSE.
The value "yes" is useful for some SIP phones (Cisco SPA)
to be able to indicate and pick up ringing devices.

ASTERISK-26919 #close

Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711
2017-06-16 12:08:27 -04:00
Jenkins2
0b7a0681a3 Merge "res_ari: Add "module loaded" check to ari stubs" into 13 2017-06-16 10:57:43 -05:00
Jenkins2
d88344c3d4 Merge "chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read." into 13 2017-06-16 07:48:28 -05:00
Jan Friesse
005a4afa6b res_corosync: Change thread stack size
In Corosync 2.x libraries were changed to use LibQB IPC.
Sadly LibQB IPC doesn't support copy-free access to received buffer, so
Corosync libraries were rewritten to use stack as buffer. Mostly the
needed stack size is quite small, but for all *_dispatch functions, 1MiB
is needed.

Asterisk function ast_pthread_create_background set stack size for new
thread to much smaller AST_BACKGROUND_STACKSIZE (~500KiB).

This results in Asterisk crash when running with Corosync 2.x.

Patch solves this issue by creating it's own version of
ast_pthread_create_background which sets stack size to much higher value
(actually it's AST_BACKGROUND_STACKSIZE + 3MiB).

Another problem may appear when "corosync show members" netconsole
command is executed. It is also executed in thread and also has only
500KiB stack size. Sadly it calls corosync_cfg_get_node_addrs which
again needs at least 1MiB stack.

Solution is to use HAVE_COROSYNC_CFG_STATE_TRACK as a discriminator
between Corosync 1.x and 2.x. If 1.x is found, nothing changes. If 2.x
is found, NodeID is displayed instead of IP address.

ASTERISK-25370 #close
Reported by: mdu113

Change-Id: Id95b0d21ab6e708e7d74ad8786c587211676fa08
2017-06-16 14:44:34 +02:00
George Joseph
7901b9853e res_ari: Add "module loaded" check to ari stubs
The recent change to make the use of LOAD_DECLINE more consistent
caused res_ari to unload itself before declining if the ari.conf
file wasn't found.  The ari stubs though still tried to use the
configuration resulting in segfaults.

This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests
to see if res_ari is actually loaded and causes the stubs to also
decline if it isn't.  The macro was then added to the mustache
template's "load_module" function.

ASTERISK-27026 #close
Reported-by: Ronald Raikes

Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d
2017-06-15 18:31:53 -06:00
Jenkins2
37bc343b58 Merge "channel: Fix reference counting in ast_channel_suppress." into 13 2017-06-15 16:18:26 -05:00
Alexei Gradinari
3b6c327c51 app_voicemail: IMAP logout on reload/unload
Closing IMAP connection on module reload or unload.

ASTERISK-24052 #close

Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525
2017-06-15 16:52:34 -04:00