29362 Commits (584f969020b47dfbd3128b3f97dcb2fd29f307ca)

 

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Author	SHA1	Message	Date
Corey Farrell	584f969020	channel: Clear channel flag in error branch. ... Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when ast_read returns NULL. ASTERISK-27100 #close Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d	9 years ago
Jenkins2	6e48c4c8e1	Merge "app_queue: Fix returning to dialplan when a queue is empty" into 14	9 years ago
Jenkins2	e0fb4eb3b1	Merge "res_pjsip: Add DTMF INFO Failback mode" into 14	9 years ago
Jenkins2	c6d78b526d	Merge "res_rtp_asterisk: Fix issues with ICE renegotiation." into 14	9 years ago
Ivan Poddubny	5563613abd	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	2c3d589128	Merge "app_voicemail: IMAP connection control" into 14	9 years ago
Joshua Colp	7e196bc8db	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	9 years ago
Torrey Searle	3f1e34bd7c	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
Torrey Searle	ca7ce834d1	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	9 years ago
Jenkins2	7af4b869a9	Merge "res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact" into 14	9 years ago
Alexei Gradinari	227f3fe93f	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	497a870ead	res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer ... Change-Id: I0bc97c6608de1d1a4228826b3b3be43f162f05f3	9 years ago
Alexei Gradinari	758c8d4638	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	d1344a9caf	Merge "bridge: stuck channel(s) after failed attended transfer" into 14	9 years ago
Kevin Harwell	e733df06cb	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	6a2fe083ce	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	9d42be930b	Merge "res_corosync: Change thread stack size" into 14	9 years ago
Jenkins2	9cfc6d6910	Merge "cdr: fix mistake spelling of a word for Unanswered." into 14	9 years ago
Jenkins2	99a1bd4781	Merge "res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact" into 14	9 years ago
Rodrigo Ramírez Norambuena	8fac4b77db	cdr: fix mistake spelling of a word for Unanswered. ... Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df	9 years ago
Jenkins2	82c37b5b51	Merge "res_stasis: Plug reference leak on stolen channels" into 14	9 years ago
Jenkins2	8cc019ebd7	Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing"" into 14	9 years ago
Jenkins2	b802d28b7e	Merge "app_voicemail: IMAP logout on reload/unload" into 14	9 years ago
Alexei Gradinari	29ca13a7e5	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
Jenkins2	93c10e09f2	Merge "formats/format_g729: Fix typo in comment" into 14	9 years ago
George Joseph	4a66babfe0	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	178d2bc7ef	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	053d92b0ba	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
Alexei Gradinari	3f61d56ddf	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	88e8fc1279	Merge "res_ari: Add "module loaded" check to ari stubs" into 14	9 years ago
Alexei Gradinari	d5404e8bf5	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	0b6a72888c	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	dd4cfa0bce	Merge "chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read." into 14	9 years ago
George Joseph	dd1fe3436e	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
Jenkins2	f2e0160a71	Merge "channel: Fix reference counting in ast_channel_suppress." into 14	9 years ago
Joshua Colp	cb824891cb	Merge "res_pjsip_pubsub: Fix reference to released endpoint" into 14	9 years ago
Jenkins2	b0d3cc079f	Merge "bridge: Add a deferred queue." into 14	9 years ago
Richard Mudgett	83b6eda33b	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
Jenkins2	7db1abbf87	Merge "app_voicemail.c: Fix compile error when IMAP enabled." into 14	9 years ago
Jenkins2	cb20642b53	Merge "app_voicemail: IMAP logout on MWI unsubscribe" into 14	9 years ago
Jenkins2	6df551353b	Merge "res_pjsip_refer/session: Calls dropped during transfer" into 14	9 years ago
Joshua Colp	8485d2fc36	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
Jenkins2	b3b53c7042	Merge "res_rtp_asterisk: Fix ssrc change for rtcp srtp" into 14	9 years ago
Jenkins2	c60cc53c74	Merge "res_pjsip_session: Correct inverted test in session_outgoing_nat_hook" into 14	9 years ago
Jenkins2	a4da8ef721	Merge "res_pjsip_transport_websocket: Add NULL check in get_write_timeout" into 14	9 years ago
Jenkins2	43cdda17fc	Merge "pjproject_bundled: Use the asterisk github mirror for download" into 14	9 years ago
Joshua Colp	bbc2d6f214	Merge "BuildSystem: Add patches to allow building with recent LibreSSL" into 14	9 years ago
Richard Mudgett	10120638c2	app_voicemail.c: Fix compile error when IMAP enabled. ... Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1	9 years ago
Alexei Gradinari	09426d8d5f	app_voicemail: IMAP logout on MWI unsubscribe ... Closing IMAP connection on MWI unsubscribe. ASTERISK-24052 #close Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd	9 years ago
George Joseph	382ae913d7	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