29405 Commits (2c7ee034bcb8a4f8e18758aa6779b05428effd51)

 

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Author	SHA1	Message	Date
Jenkins2	2c7ee034bc	Merge "app_confbridge: Make sure name recordings are always removed from the filesystem" into 14	9 years ago
Joshua Colp	c861132918	Merge "chan_iax2: On reload make sure to check for existing MWI subscription" into 14	9 years ago
Jenkins2	ac444d853f	Merge "res/res_stasis_snoop: generate silence when audiohook returns null" into 14	9 years ago
Jenkins2	3c789a1096	Merge "Sounds: Update for core sounds 1.6 release" into 14	9 years ago
Sergej Kasumovic	86c585bf19	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	9 years ago
Sergej Kasumovic	c6e1712909	chan_iax2: On reload make sure to check for existing MWI subscription ... On every reload of chan_iax2 module, MWI subscription was added, which results in additional taskprocessors being accumulated over time. This commit fixes it by making sure we check for existing subscription first. This was verified with 'core show taskprocessors' CLI command. ASTERISK-27122 #close Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9	9 years ago
Rusty Newton	9e2ac88741	Sounds: Update for core sounds 1.6 release ... Added necessary lines to make the en_NZ language set selectable and to get core sounds 1.6 pulled down. ASTERISK-26807 #close ASTERISK-25816 #close ASTERISK-26274 #close Change-Id: I84e4dd4696568cc1ba318d12ac4b075461d6eed4	9 years ago
Jenkins2	70672ef112	Merge "res/res_pjsip_t38 ensure t38 requests get rejected quickly" into 14	9 years ago
Torrey Searle	c56bc632c2	res/res_stasis_snoop: generate silence when audiohook returns null ... Currently when rtp is paused, no packets are written to the recorded audio file, causing the silence to be skipped and recording not properly time aligned. The read handler as been adapted to return a silence frame of the correct size. ASTERISK-27128 #close Change-Id: I2d7f60650457860b9c70907b14426756b058a844	9 years ago
Torrey Searle	0a1efd0ce2	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	9 years ago
Sean Bright	c663d59acd	basic-pbx: Remove res_pjsip_multihomed from sample config ... ASTERISK-27127 #close Reported by: HZMI8gkCvPpom0tM Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789	9 years ago
Jenkins2	fd84b3d9f9	Merge "res_musiconhold: Add kill_escalation_delay, kill_method to class" into 14	9 years ago
Joshua Colp	def0cf7a74	Merge "Avoid setting maxfiles for a remote asterisk" into 14	9 years ago
George Joseph	9bd8812337	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	9 years ago
Tzafrir Cohen	625e3e2e7b	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	9 years ago
George Joseph	498768651e	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	9 years ago
Jenkins2	4279d75412	Merge "res_pjsip: Fix crash with from_user containing invalid characters." into 14	9 years ago
Jenkins2	6b1fc4842f	Merge "json.c: Add backtrace log to find 'Invalid UTF-8 string' errors" into 14	9 years ago
George Joseph	83633a9c3f	Merge "res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock." into 14	9 years ago
Jenkins2	f738225b47	Merge "bridge_native_rtp.c: Fix direct media video RTP instance ACL check." into 14	9 years ago
Benjamin Keith Ford	966f9b5b9f	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	9 years ago
Richard Mudgett	f675622707	json.c: Add backtrace log to find 'Invalid UTF-8 string' errors ... Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929	9 years ago
Joshua Colp	42248e1569	Merge "app_voicemail: Cleanup ODBC connection handling" into 14	9 years ago
Richard Mudgett	3c8725a8cd	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	9 years ago
George Joseph	725a4637cf	Fix alembic branches ... Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187	9 years ago
Joshua Colp	4ce81fb759	Merge "core: Fix segfault when invoking 'data get' CLI command" into 14	9 years ago
Jenkins2	3bef1a0df9	Merge "pjproject_bundled: Allow passing configure options to bundled" into 14	9 years ago
Jenkins2	29fff43c79	Merge "channel: Clear channel flag in error branch." into 14	9 years ago
Richard Mudgett	d21eb77d22	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	9 years ago
Jenkins2	ae56efb727	Merge "chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support)." into 14	9 years ago
Jenkins2	75622dfc7d	Merge "pjsip_distributor.c: Fix deadlock with TCP type transports." into 14	9 years ago
Jenkins2	5238f49b83	Merge "pjsip_distributor.c: Fix unidentified_requests hash functions." into 14	9 years ago
Jenkins2	1a56e0f222	Merge "bridge_native_rtp: Keep rtp instance refs on bridge_channel" into 14	9 years ago
George Joseph	f9288cbf38	Merge "chan_pjsip: Fix ability to send UPDATE on COLP" into 14	9 years ago
Sean Bright	ebacfc8618	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	9 years ago
Alexander Traud	43f636f6e5	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	9 years ago
Alexander Traud	213713bc4d	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	9 years ago
Sean Bright	da69d6ce0a	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	9 years ago
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
Richard Mudgett	b5cd0370ad	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	9 years ago
Richard Mudgett	655dada7db	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	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
George Joseph	f8a95ef5b6	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	9 years ago
George Joseph	be0feb2ce8	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	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