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Author	SHA1	Message	Date
Jenkins2	067410a445	Merge "app_queue: Fix returning to dialplan when a queue is empty" into 13	8 years ago
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	8 years ago
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	8 years ago
Jenkins2	35bc1ee28d	Merge "res_rtp_asterisk: Fix issues with ICE renegotiation." into 13	8 years ago
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	8 years ago
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	8 years ago
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	8 years ago
Jenkins2	997c11235e	Merge "app_voicemail: IMAP connection control" into 13	8 years ago
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	8 years ago
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	8 years ago
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	8 years ago
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	8 years ago
Torrey Searle	154d2914fa	res/res_pjsip_t38 ensure t38 requests get rejected quickly ... arm the t38 webhook always, so we can correctly reject a T38 negotiation request when t38 is disabled on a channel Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d	8 years ago
Richard Mudgett	764d04fa87	res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer ... Change-Id: I0bc97c6608de1d1a4228826b3b3be43f162f05f3	8 years ago
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	8 years ago
Jenkins2	bd9d72793d	Merge "core_local: local channel data not being properly unref'ed and unlocked" into 13	8 years ago
Joshua Colp	7b325d55e1	Merge "bridge: stuck channel(s) after failed attended transfer" into 13	8 years ago
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	8 years ago
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	8 years ago
Jenkins2	9554166182	Merge "res_corosync: Change thread stack size" into 13	8 years ago
Jenkins2	2619c04f1e	Merge "cdr: fix mistake spelling of a word for Unanswered." into 13	8 years ago
Jenkins2	34b3665612	Merge "res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact" into 13	8 years ago
Rodrigo Ramírez Norambuena	cecf6540dc	cdr: fix mistake spelling of a word for Unanswered. ... Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df	8 years ago
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	8 years ago
Jenkins2	507ce0aa95	Merge "res_stasis: Plug reference leak on stolen channels" into 13	8 years ago
George Joseph	470bccd769	Merge "app_voicemail: IMAP logout on reload/unload" into 13	8 years ago
Jenkins2	707e0e62e6	Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing"" into 13	8 years ago
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	8 years ago
Jenkins2	2c87fced4f	Merge "formats/format_g729: Fix typo in comment" into 13	8 years ago
Jenkins2	47b9651658	Merge "Core/PBX: Deadlock between dialplan execution and application unregistration." into 13	8 years ago
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	8 years ago
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	8 years ago
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	8 years ago
Jenkins2	0b7a0681a3	Merge "res_ari: Add "module loaded" check to ari stubs" into 13	8 years ago
Jenkins2	d88344c3d4	Merge "chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read." into 13	8 years ago
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	8 years ago
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	8 years ago
Jenkins2	37bc343b58	Merge "channel: Fix reference counting in ast_channel_suppress." into 13	8 years ago
Alexei Gradinari	3b6c327c51	app_voicemail: IMAP logout on reload/unload ... Closing IMAP connection on module reload or unload. ASTERISK-24052 #close Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525	8 years ago
Jenkins2	ae6c38db98	Merge "res_pjsip_pubsub: Fix reference to released endpoint" into 13	8 years ago
Jenkins2	5fa52f0b5a	Merge "bridge: Add a deferred queue." into 13	8 years ago
Richard Mudgett	b9a4ab8c8c	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	8 years ago
Jenkins2	8b12ad2f61	Merge "app_voicemail: IMAP logout on MWI unsubscribe" into 13	8 years ago
Jenkins2	a3b142dbe0	Merge "app_voicemail.c: Fix compile error when IMAP enabled." into 13	8 years ago
Jenkins2	320fb81580	Merge "res_pjsip_refer/session: Calls dropped during transfer" into 13	8 years ago
Joshua Colp	4910a3bf40	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	8 years ago
George Joseph	8a858275eb	Merge "res_rtp_asterisk: Fix ssrc change for rtcp srtp" into 13	8 years ago
Jenkins2	ffedf68136	Merge "res_pjsip_session: Correct inverted test in session_outgoing_nat_hook" into 13	8 years ago
Jenkins2	4ade9085d5	Merge "res_pjsip_transport_websocket: Add NULL check in get_write_timeout" into 13	8 years ago
George Joseph	3065bcd021	Merge "pjproject_bundled: Use the asterisk github mirror for download" into 13	8 years ago