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Author	SHA1	Message	Date
Jenkins2	0f45c979a3	Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."	8 years ago
Corey Farrell	78a50b0343	core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. ... This adds support for parsing timelen values from config files. This includes support for all flags which apply to PARSE_INT32. Support for this parser is added to ACO via the OPT_TIMELEN_T option type. Fixes an issue where extra characters provided to ast_app_parse_timelen were ignored, they now cause an error. Testing is included. ASTERISK-27117 #close Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554	8 years ago
Joshua Colp	065c3005ad	res_rtp_asterisk / res_pjsip: Add support for BUNDLE. ... BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11	8 years ago
Torrey Searle	8b535a406b	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	8 years ago
Torrey Searle	d42a9cc9dc	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
Corey Farrell	6b138046e7	core: Add digit filtering to ast_waitfordigit_full ... This adds a parameter to ast_waitfordigit_full which can be used to only stop waiting when certain expected digits are received. Any unexpected DTMF digits are simply ignored. This also creates a new dialplan application WaitDigit. ASTERISK-27129 #close Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9	8 years ago
Holger Hans Peter Freyther	b54eb167b4	app_playback.c: Use the timezonename parameter ... In say_date_generic the timezonename parameter is passed but never used. Fix it by passing it to the ast_localtime function. ASTERISK-27124 Change-Id: I63106b8db10426d417d7275f22554a616e92fae4	8 years ago
Sean Bright	e83b9d141a	basic-pbx: Remove res_pjsip_multihomed from sample config ... ASTERISK-27127 #close Reported by: HZMI8gkCvPpom0tM Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789	8 years ago
Joshua Colp	27aeca3594	Merge "app_stream_echo: misc bug fixes"	8 years ago
Joshua Colp	8082f6cf7e	Merge "res_rtp_asterisk: trigger source change control frame when dtls is established"	8 years ago
Joshua Colp	8b27bb100b	Merge "res_musiconhold: Add kill_escalation_delay, kill_method to class"	8 years ago
Joshua Colp	767a163fea	Merge "manager: Remove AMI "Queues" action."	8 years ago
Joshua Colp	95b35cb1cb	Merge "Avoid setting maxfiles for a remote asterisk"	8 years ago
Jenkins2	fbcfa6b4b2	Merge "http.c: Reduce log spam"	8 years ago
Joshua Colp	7f09fd2c2f	bridge/core_unreal: Fix SFU bugs with forwarding frames. ... This change fixes a few things uncovered during SFU testing. 1. Unreal channels incorrectly forwarded video frames when no video stream was present on them. This caused a crash when they were read as the core requires a stream to exist for the underlying media type. The Unreal channel will now ensure a stream exists for the media type before forwarding the frame and if no stream exists then the frame is dropped. 2. Mapping of frames during bridging from the stream number of the underlying channel to the stream number of the bridge was done in the wrong location. This resulted in the frame getting dropped. This mapping now occurs on reading of the frame from the channel. 3. Bridging was using the wrong ast_read function resulting in it living in a non-multistream world. 4. In bridge_softmix when adding new streams to existing channels the wrong stream topology was copied resulting in no streams being added. Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8	8 years ago
George Joseph	b7a875778a	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
Benjamin Keith Ford	5d86da61a6	manager: Remove AMI "Queues" action. ... When performing the "Queues" action via AMI, it outputs the same text that the Asterisk CLI outputs when running a "queue show" command, which does not conform with the AMI spec. "QueueStatus" already does what the "Queues" action should do, so instead of correcting the output, the "Queues" action will be removed and "QueueStatus" should be used instead. ASTERISK-27073 #close Reported by: Brian Change-Id: Id11743859758255b69cc3a557750d7a56c6d16f8	8 years ago
Tzafrir Cohen	d58ef31acd	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	303f935a50	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 from ast_iostream_printf of headers. Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b	8 years ago
Jenkins2	3e7cfe3a92	Merge "res_pjsip: Fix crash with from_user containing invalid characters."	8 years ago
Jenkins2	b0e184f0a7	Merge "json.c: Add backtrace log to find 'Invalid UTF-8 string' errors"	8 years ago
Jenkins2	f878ac2d07	Merge "res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock."	8 years ago
Jenkins2	789475336b	Merge "bridge_native_rtp.c: Fix direct media video RTP instance ACL check."	8 years ago
Benjamin Keith Ford	8f72128e66	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
George Joseph	17103ca898	Merge "app_queue: Add priority to AMI QueueStatus"	8 years ago
Richard Mudgett	03ae8b0105	json.c: Add backtrace log to find 'Invalid UTF-8 string' errors ... Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929	8 years ago
Joshua Colp	6f35428c87	Merge "app_voicemail: Cleanup ODBC connection handling"	8 years ago
Jenkins2	d6c08cc559	Merge "core: Remove 'Data Retrieval API'"	8 years ago
Richard Mudgett	9cd8a1df79	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
George Joseph	7a4f577eb7	Fix alembic branches ... Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187	8 years ago
Joshua Colp	b104e484b6	Merge "channel: Clear channel flag in error branch."	8 years ago
Jenkins2	33aa3907eb	Merge "pjproject_bundled: Allow passing configure options to bundled"	8 years ago
Richard Mudgett	1028f64be4	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	7a306468f4	Merge "bridge_native_rtp: Keep rtp instance refs on bridge_channel"	8 years ago
Jenkins2	75022f6b11	Merge "chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support)."	8 years ago
Jenkins2	2ec388680b	Merge "chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support)."	8 years ago
George Joseph	a10bc3e23f	Merge "pjsip_distributor.c: Fix deadlock with TCP type transports."	8 years ago
Jenkins2	16f0fa52c0	Merge "pjsip_distributor.c: Fix unidentified_requests hash functions."	8 years ago
Jenkins2	d2b32cd009	Merge "chan_pjsip: Fix ability to send UPDATE on COLP"	8 years ago
Sean Bright	325eeced6a	core: Remove 'Data Retrieval API' ... This API was not actively maintained, was not added to new modules (such as res_pjsip), and there exist better alternatives to acquire the same information, such as the ARI. Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83	8 years ago
Rodrigo Ramírez Norambuena	d556c67f9f	app_queue: Add change priority of call ... This patch include a feature to change the priority a caller in a queue by CLI and AMI. Change-Id: I55d520d71cc1cefe9a9b81fefaefc14679e96133	8 years ago
Alexander Traud	910c05455d	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	4398aa8fa4	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	950b39a4f5	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	50ddb56dad	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	b62a3f0a67	Merge "app_queue: Fix returning to dialplan when a queue is empty"	8 years ago
Richard Mudgett	b485f6c59c	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	65a5ac0168	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	e1c0e14fac	Merge "res_pjsip: Add DTMF INFO Failback mode"	8 years ago
Joshua Colp	16e43ef701	Merge "res_rtp_asterisk: Fix issues with ICE renegotiation."	8 years ago