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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	9 years ago
George Joseph	80e11bd79b	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	9 years ago
Ivan Poddubny	7827755570	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	9 years ago
Jenkins2	d59b0efabd	Merge "res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact"	9 years ago
Alexei Gradinari	0cef7b9d4e	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	9 years ago
Richard Mudgett	975e271b01	res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer ... Change-Id: I0bc97c6608de1d1a4228826b3b3be43f162f05f3	9 years ago
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	9 years ago
Jenkins2	01536546e2	Merge "bridge: stuck channel(s) after failed attended transfer"	9 years ago
Jenkins2	0cf331d7e9	Merge "core_local: local channel data not being properly unref'ed and unlocked"	9 years ago
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	9 years ago
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	9 years ago
Jenkins2	db5e269365	Merge "res_corosync: Change thread stack size"	9 years ago
Jenkins2	0c0d69d4f3	Merge "cdr: fix mistake spelling of a word for Unanswered."	9 years ago
Joshua Colp	0ecf504de9	Merge "res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact"	9 years ago
Joshua Colp	57bbba7d43	Merge "res_stasis: Plug reference leak on stolen channels"	9 years ago
Rodrigo Ramírez Norambuena	a7488f8a70	cdr: fix mistake spelling of a word for Unanswered. ... Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df	9 years ago
George Joseph	3f5bf287a2	Merge "SDP: Add get/set option calls for RTP sched context per type."	9 years ago
Jenkins2	317234bdc6	Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing""	9 years ago
Jenkins2	d57378d522	Merge "app_voicemail: IMAP logout on reload/unload"	9 years ago
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	9 years ago
Joshua Colp	58d50025a0	Merge "formats/format_g729: Fix typo in comment"	9 years ago
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	9 years ago
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	9 years ago
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	9 years ago
Joshua Colp	0405185357	Merge "SDP: Search for the ice-lite attribute in the right place."	9 years ago
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.	9 years ago
Jenkins2	2f684eb6a5	Merge "stream: Add ast_stream_topology_del_stream() and unit test."	9 years ago
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	9 years ago
Jenkins2	322a1c3222	Merge "res_ari: Add "module loaded" check to ari stubs"	9 years ago
Alexei Gradinari	53b7df82f4	app_voicemail: IMAP logout on reload/unload ... Closing IMAP connection on module reload or unload. ASTERISK-24052 #close Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525	9 years ago
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	9 years ago
Jenkins2	b1f46c2a36	Merge "chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read."	9 years ago
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	9 years ago
Joshua Colp	41bd01c861	Merge "channel: Fix reference counting in ast_channel_suppress."	9 years ago
Jenkins2	2adc0aef19	Merge "res_pjsip_pubsub: Fix reference to released endpoint"	9 years ago
Joshua Colp	1f2ab6e72a	Merge "bridge: Add a deferred queue."	9 years ago
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	9 years ago
Richard Mudgett	e563a1920e	SDP: Add get/set option calls for RTP sched context per type. ... Change-Id: I82dc75c63c48904e9e5a49e2205dcc06e88487e4	9 years ago
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	9 years ago
Richard Mudgett	a95584d079	SDP: Set the remote c= line in RTP instance. ... Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c	9 years ago
Richard Mudgett	06265b8c8a	stream: Add ast_stream_topology_del_stream() and unit test. ... Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9	9 years ago
Richard Mudgett	0fdb99c268	SDP: Add t= line in sdp_create_from_state() ... Change-Id: I4060391328a893101ed87d0d9bacbbab4fd8b141	9 years ago
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	9 years ago
George Joseph	15c04db3bd	Merge "app_voicemail.c: Fix compile error when IMAP enabled."	9 years ago
George Joseph	f743e2e84c	Merge "app_voicemail: IMAP logout on MWI unsubscribe"	9 years ago
Jenkins2	2c3c862cee	Merge "res_pjsip_refer/session: Calls dropped during transfer"	9 years ago
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	9 years ago
George Joseph	54a08a2e43	Merge "res_rtp_asterisk: Fix ssrc change for rtcp srtp"	9 years ago
Jenkins2	4681b9baef	Merge "res_pjsip_session: Correct inverted test in session_outgoing_nat_hook"	9 years ago
Jenkins2	433e876317	Merge "res_pjsip_transport_websocket: Add NULL check in get_write_timeout"	9 years ago