Commit Graph

29555 Commits

Author SHA1 Message Date
Torrey Searle a48d3e4d31 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 11:46:23 -05:00
Jenkins2 d59b0efabd Merge "res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact" 2017-06-22 16:01:52 -05:00
Richard Mudgett 975e271b01 res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer
Change-Id: I0bc97c6608de1d1a4228826b3b3be43f162f05f3
2017-06-21 18:25:17 -05:00
Alexei Gradinari 34db4c3993 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 18:24:31 -05:00
Jenkins2 01536546e2 Merge "bridge: stuck channel(s) after failed attended transfer" 2017-06-21 17:57:21 -05:00
Jenkins2 0cf331d7e9 Merge "core_local: local channel data not being properly unref'ed and unlocked" 2017-06-21 17:30:01 -05:00
Kevin Harwell 27dae55fb6 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 16:18:13 -05:00
Kevin Harwell 45a1f4e2ae 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 11:17:46 -05:00
Jenkins2 db5e269365 Merge "res_corosync: Change thread stack size" 2017-06-20 18:18:19 -05:00
Jenkins2 0c0d69d4f3 Merge "cdr: fix mistake spelling of a word for Unanswered." 2017-06-20 09:25:17 -05:00
Joshua Colp 0ecf504de9 Merge "res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact" 2017-06-20 05:47:46 -05:00
Joshua Colp 57bbba7d43 Merge "res_stasis: Plug reference leak on stolen channels" 2017-06-19 16:49:39 -05:00
Rodrigo Ramírez Norambuena a7488f8a70 cdr: fix mistake spelling of a word for Unanswered.
Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df
2017-06-19 12:28:18 -04:00
George Joseph 3f5bf287a2 Merge "SDP: Add get/set option calls for RTP sched context per type." 2017-06-19 09:27:43 -05:00
Jenkins2 317234bdc6 Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing"" 2017-06-19 09:09:58 -05:00
Jenkins2 d57378d522 Merge "app_voicemail: IMAP logout on reload/unload" 2017-06-19 08:52:12 -05:00
Alexei Gradinari d7b6e06abb 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 17:54:43 -05:00
Joshua Colp 58d50025a0 Merge "formats/format_g729: Fix typo in comment" 2017-06-16 16:37:18 -05:00
George Joseph 854a6de819 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 15:08:45 -05:00
Matthew Fredrickson e33bd96638 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 14:56:37 -05:00
Frederic LE FOLL 0ad95bc8a0 Core/PBX: Deadlock between dialplan execution and application unregistration.
Not easy to reproduce, but we have noticed deadlocks when unloading a module
while dialplan is handling a request.

The deadlock is between :
1) Dialplan execution: pbx_extension_helper() first taking conlock,
then pbx_findapp() [when called] asking for lock on apps list.
2) Application unregistration: ast_unregister_application() first taking lock
on apps list, then unreference_cached_app() [when called] asking for conlock.

As a protection, I suggest to modify ast_unregister_application(), so that it
anticipates the need of conlock, before taking the lock on apps list.
The side effect is a longer unavailability of conlock when unregistering an
application.

ASTERISK-27041

Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2
2017-06-16 13:26:22 -05:00
Joshua Colp 0405185357 Merge "SDP: Search for the ice-lite attribute in the right place." 2017-06-16 12:00:38 -05:00
Jenkins2 d81293a5dd Merge changes from topic 'sdp_api_adjustments'
* changes:
  SDP: Set the remote c= line in RTP instance.
  SDP: Add t= line in sdp_create_from_state()
  stream: Ignore declined streams for some topology calls.
2017-06-16 11:51:41 -05:00
Jenkins2 2f684eb6a5 Merge "stream: Add ast_stream_topology_del_stream() and unit test." 2017-06-16 11:50:32 -05:00
Alexei Gradinari 7a46309d3d 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 11:25:07 -05:00
Jenkins2 322a1c3222 Merge "res_ari: Add "module loaded" check to ari stubs" 2017-06-16 11:12:48 -05:00
Alexei Gradinari 53b7df82f4 app_voicemail: IMAP logout on reload/unload
Closing IMAP connection on module reload or unload.

ASTERISK-24052 #close

Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525
2017-06-16 10:45:06 -05:00
Jan Friesse 9aeab4aced 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 07:53:22 -05:00
Jenkins2 b1f46c2a36 Merge "chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read." 2017-06-16 07:51:17 -05:00
George Joseph 1ac0096512 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 19:34:03 -05:00
Joshua Colp 41bd01c861 Merge "channel: Fix reference counting in ast_channel_suppress." 2017-06-15 16:24:55 -05:00
Jenkins2 2adc0aef19 Merge "res_pjsip_pubsub: Fix reference to released endpoint" 2017-06-15 15:24:25 -05:00
Joshua Colp 1f2ab6e72a Merge "bridge: Add a deferred queue." 2017-06-15 15:02:26 -05:00
Richard Mudgett 11ec2945c7 chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read.
The construction of the returned string assumed incorrectly that the
supplied buffer would always be initialized as an empty string.  If it is
not an empty string we could overrun the supplied buffer by the length of
the non-empty buffer string plus one.  It is also theoreticaly possible
for the supplied buffer to be overrun by a string terminator during a read
operation even if the supplied buffer is an empty string.

* Fix the assumption that the supplied buffer would already be an empty
string.  The buffer is not guaranteed to contain an empty string by all
possible callers.

* Fix string terminator buffer overrun potential.

Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9
2017-06-15 12:35:53 -05:00
Richard Mudgett e563a1920e SDP: Add get/set option calls for RTP sched context per type.
Change-Id: I82dc75c63c48904e9e5a49e2205dcc06e88487e4
2017-06-15 09:42:15 -05:00
Richard Mudgett 716abaf33d SDP: Search for the ice-lite attribute in the right place.
* Pulled finding the rtcp-mux attribute flag out of the ICE candidate for
loop.  Also ordered the RTCP ICE candidate skip test to fail earlier.

Change-Id: I8905d9c68563027a46cd3ae14dbcc27e9c814809
2017-06-15 09:42:15 -05:00
Richard Mudgett a95584d079 SDP: Set the remote c= line in RTP instance.
Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c
2017-06-15 09:42:15 -05:00
Richard Mudgett 06265b8c8a stream: Add ast_stream_topology_del_stream() and unit test.
Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9
2017-06-15 09:42:15 -05:00
Richard Mudgett 0fdb99c268 SDP: Add t= line in sdp_create_from_state()
Change-Id: I4060391328a893101ed87d0d9bacbbab4fd8b141
2017-06-15 09:42:15 -05:00
Richard Mudgett 4797a8bb81 stream: Ignore declined streams for some topology calls.
* Made ast_format_cap_from_stream_topology() not include any formats from
declined streams.

* Made ast_stream_topology_get_first_stream_by_type() ignore declined
streams to return the first active stream of the type.

* Updated unit tests to check these changes have the expected effect.

Change-Id: Iabbc6a3e8edf263a25fd3056c3c614407c7897df
2017-06-15 09:42:15 -05:00
George Joseph 15c04db3bd Merge "app_voicemail.c: Fix compile error when IMAP enabled." 2017-06-15 09:06:23 -05:00
George Joseph f743e2e84c Merge "app_voicemail: IMAP logout on MWI unsubscribe" 2017-06-15 09:05:57 -05:00
Jenkins2 2c3c862cee Merge "res_pjsip_refer/session: Calls dropped during transfer" 2017-06-15 08:12:43 -05:00
Joshua Colp bd16c3c524 channel: Fix reference counting in ast_channel_suppress.
The ast_channel_suppress function wrongly decremented the
reference count of the underlying structure used to keep
track of what should be suppressed on a channel if the
function was called multiple times on the same channel.

This change cleans up the reference counting a bit so
this no longer occurs.

ASTERISK-27016

Change-Id: I2eed4077cb4916e6626f9f120b63b963acc5c136
2017-06-15 07:36:59 -05:00
George Joseph 54a08a2e43 Merge "res_rtp_asterisk: Fix ssrc change for rtcp srtp" 2017-06-14 16:05:37 -05:00
Jenkins2 4681b9baef Merge "res_pjsip_session: Correct inverted test in session_outgoing_nat_hook" 2017-06-14 15:54:22 -05:00
Jenkins2 433e876317 Merge "res_pjsip_transport_websocket: Add NULL check in get_write_timeout" 2017-06-14 15:24:32 -05:00
George Joseph c27acb4c99 Merge "pjproject_bundled: Use the asterisk github mirror for download" 2017-06-14 14:42:32 -05:00
Richard Mudgett b8b0b61a24 app_voicemail.c: Fix compile error when IMAP enabled.
Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1
2017-06-14 12:35:54 -05:00
Alexei Gradinari 023eede265 app_voicemail: IMAP logout on MWI unsubscribe
Closing IMAP connection on MWI unsubscribe.

ASTERISK-24052 #close

Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd
2017-06-14 11:30:34 -05:00