29016 Commits (3858d99b735dd45239984d6ff8eedaaa2beb45f6)

 

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Author	SHA1	Message	Date
Sergej Kasumovic	3858d99b73	app_confbridge: Make sure name recordings are always removed from the filesystem ... This commit fixes two possible scenarios: * When recording name and if during recording you hangup, file is never removed. This is due to the fact file location is nulled. * When recording name and if you hangup during thank-you prompt, file is never removed. ASTERISK-27123 #close Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625	8 years ago
Jenkins2	ec182b6d50	Merge "res_musiconhold: Add kill_escalation_delay, kill_method to class" into 13	8 years ago
Joshua Colp	3bf42a58c4	Merge "http.c: Reduce log spam" into 13	8 years ago
George Joseph	4e555437dc	res_musiconhold: Add kill_escalation_delay, kill_method to class ... By default, when res_musiconhold reloads or unloads, it sends a HUP signal to custom applications (and all descendants), waits 100ms, then sends a TERM signal, waits 100ms, then finally sends a KILL signal. An application which is interacting with an external device and/or spawns children of its own may not be able to exit cleanly in the default times, expecially if sent a KILL signal, or if it's children are getting signals directly from res_musiconhoild. * To allow extra time, the 'kill_escalation_delay' class option can be used to set the number of milliseconds res_musiconhold waits before escalating kill signals, with the default being the current 100ms. * To control to whom the signals are sent, the "kill_method" class option can be set to "process_group" (the default, existing behavior), which sends signals to the application and its descendants directly, or "process" which sends signals only to the application itself. Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b	8 years ago
Tzafrir Cohen	4f2f3bfebf	Avoid setting maxfiles for a remote asterisk ... Setting maxfiles (maximum number of open files) has no practical effect on a remote asterisk (rasterisk, rasterisk -x). It has an ill effect of printing an extra message, which may be annoying in case of -x. ASTERISK-27105 #close Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2	8 years ago
George Joseph	32b98ad956	http.c: Reduce log spam ... Messages like "fwrite() failed: Connection reset by peer" are no help whatsoever, especially since they can be caused simply by a client disconnecting. * Make those WARNINGs DEBUGs. * Check the return of the headers fprintf. Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b	8 years ago
Jenkins2	1498d66f69	Merge "res_pjsip: Fix crash with from_user containing invalid characters." into 13	8 years ago
Jenkins2	b5cd0f9ff3	Merge "json.c: Add backtrace log to find 'Invalid UTF-8 string' errors" into 13	8 years ago
Jenkins2	ccd1015bcb	Merge "res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock." into 13	8 years ago
Jenkins2	baa51a3bb5	Merge "bridge_native_rtp.c: Fix direct media video RTP instance ACL check." into 13	8 years ago
Benjamin Keith Ford	25e18bf514	res_pjsip: Fix crash with from_user containing invalid characters. ... If the from_user field contains certain characters (like @, {, ^, etc.), PJSIP will return a null value for the URI when attempting to parse it. This causes a crash when trying to dial out through a trunk that contains these invalid characters in its from_user field. This change checks the configuration and ensures that an endpoint will not be created if the from_user contains an invalid character. It also adds a null check to the PJSIP URI parsing as a backup. ASTERISK-27036 #close Reported by: Maxim Vasilev Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0	8 years ago
Richard Mudgett	8a803f75a0	json.c: Add backtrace log to find 'Invalid UTF-8 string' errors ... Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929	8 years ago
Jenkins2	66eb4f34cf	Merge "app_voicemail: Cleanup ODBC connection handling" into 13	8 years ago
Richard Mudgett	aa514f420b	res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock. ... When a message is received on the TURN socket, the code processing the message needs to call into the ICE/STUN session for further processing. This code path locks the TURN group lock then the ICE/STUN group lock. In another thread an ICE/STUN timer can fire off to send a keep alive message over the TURN socket. In this code path, the ICE/STUN group lock is obtained then the TURN group lock is obtained to send the packet. A classic deadlock case if the group locks are not the same. * Made TURN get created using the ICE/STUN session's group lock. NOTE: I was originally concerned that the ICE/STUN session can get recreated by ice_reset_session() for an event like RTCP multiplexing causing a change during SDP negotiation. In this case the TURN group lock would become different. However, TURN is also recreated as part of the ICE/STUN recreation in ice_create() when all known ICE candidates are added to the new ICE session. While the ICE/STUN and TURN sessions are being recreated there is a period where the group locks could be different. ASTERISK-27023 #close Patches: res_rtp_asterisk-turn-deadlock-fix.patch (license #6502) patch uploaded by Michael Walton (modified) Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9	8 years ago
Joshua Colp	5a894ff27e	Merge "Fix alembic branches" into 13	8 years ago
George Joseph	379fe65831	Fix alembic branches ... Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187	8 years ago
Jenkins2	cad74cdd8f	Merge "core: Fix segfault when invoking 'data get' CLI command" into 13	8 years ago
Jenkins2	dc1fc28a4a	Merge "pjproject_bundled: Allow passing configure options to bundled" into 13	8 years ago
Richard Mudgett	22c4c1a0ba	bridge_native_rtp.c: Fix direct media video RTP instance ACL check. ... The video stream was using the audio stream RTP instance addresses to check if the video RTP gets directed to an allowed direct media Access Control List (ACL) address. There is no guarantee that the video RTP instance uses the same addresses as the audio RTP instance. This looks like it has been a bug since v11 when direct media ACL was first added to chan_sip and then faithfully reproduced through a couple code refactorings into the new bridging architecture. Change-Id: I8ddd56320e0eea769f3ceed3fa5b6bdfb51d681a	8 years ago
George Joseph	642c597507	Merge "pjsip_distributor.c: Fix deadlock with TCP type transports." into 13	8 years ago
Jenkins2	cf6e0b8f8b	Merge "chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support)." into 13	8 years ago
Jenkins2	aadece5664	Merge "pjsip_distributor.c: Fix unidentified_requests hash functions." into 13	8 years ago
Jenkins2	d55798fff0	Merge "bridge_native_rtp: Keep rtp instance refs on bridge_channel" into 13	8 years ago
George Joseph	40490768cc	Merge "chan_pjsip: Fix ability to send UPDATE on COLP" into 13	8 years ago
Jenkins2	2bbe8cb3f5	Merge "channel: Clear channel flag in error branch." into 13	8 years ago
Sean Bright	6258de458b	core: Fix segfault when invoking 'data get' CLI command ... Invoking 'data get /asterisk/core/channeltypes' caused a crash because of an assumption of a tech's capabilities to be non-NULL. The 'Surrogate' tech, however, does have a NULL capabilities member, resulting in a crash. ASTERISK-27108 #close Change-Id: I2fbe7715681f43d5565d1e1599269468c26b0e0a	8 years ago
Alexander Traud	39d2ebbf56	chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support). ... When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was added in any case, because of a local Boolean-negation error of the return value of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was still always added with tlsenable=yes, because the domains were not compared just on the address but also on the port – and TLS is always on a different port than UDP/TCP. ASTERISK-27106 Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c	8 years ago
Alexander Traud	9f4b3b966e	chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support). ... Because of a copy-and-paste error when the struct ast_sockaddr changed, tlsbindaddr was not added, when sip.conf contained autodomain=yes; see "show sip domains" on the command-line interface (CLI) of Asterisk. ASTERISK-27106 Change-Id: I3d0957150017c223136968ef1266f275d0d6695e	8 years ago
Sean Bright	194625c1de	app_voicemail: Cleanup ODBC connection handling ... The primary focus of this patch is adding a missing call to ast_odbc_release_obj(), but is also a general cleanup of the ODBC related code in app_voicemail. ASTERISK-27093 #close Change-Id: I8e285142eaeb3146b4287a928276b70db76c902b	8 years ago
Corey Farrell	73520e9f58	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	8 years ago
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
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