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Author	SHA1	Message	Date
George Joseph	3f5bf287a2	Merge "SDP: Add get/set option calls for RTP sched context per type."	8 years ago
Jenkins2	317234bdc6	Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing""	8 years ago
Jenkins2	d57378d522	Merge "app_voicemail: IMAP logout on reload/unload"	8 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
George Joseph	c27acb4c99	Merge "pjproject_bundled: Use the asterisk github mirror for download"	9 years ago
Richard Mudgett	b8b0b61a24	app_voicemail.c: Fix compile error when IMAP enabled. ... Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1	9 years ago
Alexei Gradinari	023eede265	app_voicemail: IMAP logout on MWI unsubscribe ... Closing IMAP connection on MWI unsubscribe. ASTERISK-24052 #close Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd	9 years ago
George Joseph	65ed2ea311	res_pjsip_pubsub: Fix reference to released endpoint ... destroy_subscription was attempting to get the id of the subscription tree's endpoint after we'd already called ao2_cleanup on it causing a segfault. Moved the cleanup until after the debug statement and since endpoint could also be NULL at this point, check for that as well. ASTERISK-27057 #close Reported-by: Ryan Smith Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678	9 years ago
George Joseph	ea3f8c6889	res_pjsip_session: Correct inverted test in session_outgoing_nat_hook ... There was a typo introduced in commit 776ffd77 which was preventing the transport's external media address from being used. ASTERISK-27024 #close Reported-by: Christopher van de Sande patches: patch.diff submitted by Florian Floimair (license 6892) Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27	9 years ago
Joshua Colp	c33168c147	Merge "CFLAGS for BIND8 support"	9 years ago
George Joseph	88f18faf2a	res_rtp_asterisk: Fix ssrc change for rtcp srtp ... It looks like there was a copy/paste error in ast_rtp_change_source where if there was a rtcp srtp instance, instead of updating its ssrc we were updating the srtp instance ssrc twice. ASTERISK-27022 #close Reported-by: Michael Walton Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095	9 years ago
Joshua Colp	d6386a8f0c	bridge: Add a deferred queue. ... This change adds a deferred queue to bridging. If a bridge technology determines that a frame can not be written and should be deferred it can indicate back to bridging to do so. Bridging will then requeue any deferred frames upon a new channel joining the bridge. This change has been leveraged for T.38 request negotiate control frames. Without the deferred queue there is a race condition between the bridge receiving the T.38 request negotiate and the second channel joining and being in the bridge. If the channel is not yet in the bridge then the T.38 negotiation fails. A unit test has also been added that confirms that a T.38 request negotiate control frame is deferred when no other channel is in the bridge and that it is requeued when a new channel joins the bridge. ASTERISK-26923 Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415	9 years ago
Kevin Harwell	9e53c30610	res_pjsip_refer/session: Calls dropped during transfer ... When doing an attended transfer it's possible for the transferer, after receiving an accepted response from Asterisk, to send a BYE to Asterisk, which can then be processed before Asterisk has time to start and/or complete the transfer process. This of course causes the transfer to not complete successfully, thus dropping the call. This patch makes it so any BYEs received from the transferer, after the REFER, that initiate a session end are deferred until the transfer is complete. This allows the channel that would have otherwise been hung up by Asterisk to remain available throughout the transfer process. ASTERISK-27053 #close Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a	9 years ago
George Joseph	b2fd7e5069	pjproject_bundled: Use the asterisk github mirror for download ... We now mirror the pjproject tarball and md5 at https://github.com/asterisk/third-party/tree/master/pjproject To improve download reliability, we now get the tarball from our mirror instead of from pjsip.org. ASTERISK-27052 #close Reported-by: 'alex' Change-Id: I60236587a8935bfa71fcc391f4e2ecb31918c08a	9 years ago
Alexei Gradinari	42f738e052	res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled ... If sending unsolicited mwi to all endpoints on startup is disabled (mwi_disable_initial_unsolicited=yes) do not need to create subscriptions. If there are many (thousands) realtime endpoints configured with unsolicited mwi and Vociemail Storage configured as ODBC or IMAP there will be huge number of DB/IMAP requests on startup. ASTERISK-26230 #close Change-Id: I50ae909639e3ee298b931a54def4b2b9e0fb86c5	9 years ago
Jenkins2	abe1dd3039	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing."	9 years ago
Jenkins2	5d3420a2de	Merge "BuildSystem: Add patches to allow building with recent LibreSSL"	9 years ago
Jenkins2	b17e05b53a	Merge "BuildSystem: Fix build on FreeBSD due to missing crypt.h"	9 years ago
Jenkins2	44a0edcf71	Merge "codecs.conf.sample: Fix max_bandwidth speling error"	9 years ago
Joshua Colp	ac5f40f797	Merge "eventfd: Disable during cross compilation"	9 years ago