28978 Commits (2c43ca0ac50764ab17d691844a84158bbf590b0e)

 

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Author	SHA1	Message	Date
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
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
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
Joshua Colp	4ece39f476	Merge "BuildSystem: Add patches to allow building with recent LibreSSL" into 13	8 years ago
Richard Mudgett	f1a209d5ac	app_voicemail.c: Fix compile error when IMAP enabled. ... Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1	8 years ago
Frederic LE FOLL	dc307af7f2	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	8 years ago
George Joseph	c2eea791e4	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	8 years ago
George Joseph	2dee95cc7a	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	8 years ago
Jenkins2	e26d15fabb	Merge "CFLAGS for BIND8 support" into 13	8 years ago
Jørgen H	e16a669c70	res_pjsip_transport_websocket: Add NULL check in get_write_timeout ... Added check for NULL return value when calling ast_sorcery_retrieve_by_id in function get_write_timeout ASTERISK-27046 Change-Id: I9357717278da631c3a1cb502c412693929b0cb41	8 years ago
Jenkins2	44d4b55697	Merge "res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled" into 13	8 years ago
George Joseph	7dafe82751	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	8 years ago