28544 Commits (e910dbab90ef3d628955c49f441b2c9dda1f222c)

 

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Author	SHA1	Message	Date
Igor Goncharovskiy	3faca1d4ff	Fix closing rtp ports after call finished in chan_unistim. ... Fix ASTERISK-26565 by adding ast_rtp_instance_stop before rtp instance destroy for chan_unistim. Also several fixes for displayed text translation. Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc	9 years ago
Joshua Colp	10e64a5fb8	Merge "res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp." into 13	9 years ago
zuul	cf08676c00	Merge "build: Fix default values for some SANITIZER options" into 13	9 years ago
Richard Mudgett	412d43fa21	res_pjsip.c: Rework endpt_send_request() req_wrapper code. ... * Don't hold the req_wrapper lock too long in endpt_send_request(). We could block the PJSIP monitor thread if the timeout timer expires. sip_get_tpselector_from_endpoint() does a sorcery access that could take awhile accessing a database. pjsip_endpt_send_request() might take awhile if selecting a transport. * Shorten the time that the req_wrapper lock is held in the callback functions. * Simplify endpt_send_request() req_wrapper->timeout code. * Removed some redundant req_wrapper->timeout_timer->id assignments. Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9	9 years ago
Richard Mudgett	2e7fc56d3c	res_pjsip: Fix tdata leaks in off nominal paths. ... Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b	9 years ago
Richard Mudgett	da68b185b3	res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. ... Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94	9 years ago
George Joseph	ee73af1d88	Update for 13.12.2	9 years ago
Joshua Colp	b70eb07c53	res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp. ... When optimistic SRTP was on it was possible for us to still set up a call without an audio stream if an offer was received with required SRTP. This change makes it so this scenario will now fail with a 488 response. ASTERISK-26575 Change-Id: I7d14187037681f48879bd20319ac79d0877318f3	9 years ago
Joshua Colp	71dc333565	app_queue: Add mention of 'ABANDON' variable to CHANGES. ... ASTERISK-26558 Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e	9 years ago
George Joseph	ccc8a37584	Merge "Revert "autoservice: Use frame deferral API"" into 13	9 years ago
George Joseph	07e7ac5119	Merge "Revert "Add API for channel frame deferral."" into 13	9 years ago
George Joseph	06045fc29e	Merge "Revert "AGI: Only defer frames when in an interception routine."" into 13	9 years ago
George Joseph	9e13c3e2a1	Merge "Revert "channel: Use frame deferral API for safe sleep."" into 13	9 years ago
George Joseph	6b5a7ced13	Revert "Add API for channel frame deferral." ... This reverts commit 9231a56cf3. Multiple testsuite failures were detected after the fact. Change-Id: I3bac8d7c3ddb69a4ddf6c5d6de0ffa5ff7ff3af7	9 years ago
George Joseph	6be5d8de0d	Revert "AGI: Only defer frames when in an interception routine." ... This reverts commit 5c10091f3d. Multiple testsuite failures were detected after the fact. Change-Id: I397a841acc17ae230c512449cd6bed89d2ef3b73	9 years ago
George Joseph	1df434e2b4	Revert "autoservice: Use frame deferral API" ... This reverts commit 2e3a354575. Multiple testsuite failures were detected after the fact. Change-Id: Ia45fa4633fae74dca345b24bb6722737c63035de	9 years ago
George Joseph	58c88cfbaa	Revert "channel: Use frame deferral API for safe sleep." ... This reverts commit 44f7e25239. Multiple testsuite failures were detected after the fact. Change-Id: I56299087da22128a95f0c8f3955f740890d7ca65	9 years ago
Joshua Colp	44fd077739	Merge "app_queue: new variable set when abandoned" into 13	9 years ago
George Joseph	a562fbe618	build: Fix default values for some SANITIZER options ... 2 of the sanitizers didn't have default values so in systems that don't support sanitizers menuselect would spit out warnings. They were harmless but confusing. They've now been set to "0". Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58	9 years ago
George Joseph	dfcb2b6c24	Merge "res_pjsip_session: Do not call session supplements when it's too late." into 13	9 years ago
Sebastian Gutierrez	7fd5031c1c	app_queue: new variable set when abandoned ... sets the variable ABANDONED to TRUE if the call was not answered. ASTERISK-26558 Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3	9 years ago
Mark Michelson	e043d1a55c	res_pjsip_session: Do not call session supplements when it's too late. ... res_pjsip_sesssion was hooking into transaction and invite state changes. One of the reasons for doing so was due to the PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the message sending process, and so we should call session supplements to alter the outgoing message. In reality, this event was meant to indicate that the message either a) had already been sent, or b) required a DNS lookup and would be sent when the DNS query completed. In case (a), this meant we were altering an already-sent request/response for no reason. In case (b), this potentially meant we could be trying to alter a request/response at the same time that the DNS resolution completed. In this case, it meant we might be stomping on memory being used by the thread actually sending the message. This caused potential crashes and memory corruption. This patch removes the calls to session supplements from the case where the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to alter the message at this point is too late, and it can cause nothing but harm to try to do it. Because there were no longer any calls to the handle_outgoing() function, it has been removed. Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92	9 years ago
Mark Michelson	44f7e25239	channel: Use frame deferral API for safe sleep. ... This is another case where manual frame deferral can be replaced with centralized routines instead. Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e	9 years ago
Mark Michelson	2e3a354575	autoservice: Use frame deferral API ... Rather than use manual frame deferral, just let the channel API do it for us. ASTERISK-26343 Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49	9 years ago
Mark Michelson	5c10091f3d	AGI: Only defer frames when in an interception routine. ... AGI recently was modified to defer important frames. This was because when AGI was used in a connected line interception routine, the resulting connected line frame would end up getting discarded by the AGI. However, this caused bad behavior in other cases. Specifically, during a transfer, if someone attempted to manually set the Caller ID on a channel in an AGI, the deferred connected line frame would end up overwriting what had been manually set in the AGI. Since the initial issue was specific to interception routines, this change removes the manual frame deferral from AGI and instead uses the new frame deferral API in interception routines. ASTERISK-26343 #close Reported by Morton Tryfoss Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208	9 years ago
zuul	3870e8d76d	Merge "automon: restore mixing of the both channels after recording stops" into 13	9 years ago
zuul	7477c95749	Merge "Add API for channel frame deferral." into 13	9 years ago
Joshua Colp	093eadf46a	Merge "chan_ooh323: reset rrq count on gk registration" into 13	9 years ago
Joshua Colp	0310845acd	Merge "chan_ooh323: Fixes to work right with Cisco devices" into 13	9 years ago
Joshua Colp	77e56bc2e0	Merge "stasis_recording/stored: remove calls to deprecated readdir_r function." into 13	9 years ago
Joshua Colp	dc04c1725f	Merge "res_stasis: Don't unsubscribe from a NULL bridge." into 13	9 years ago
Joshua Colp	58df0ccc0e	Merge "res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems" into 13	9 years ago
Joshua Colp	768bac97cc	Merge "res_stasis: Set a video source mode on Stasis created bridges" into 13	9 years ago
Joshua Colp	87c884965c	Merge "main/bridge: Add some verbose logging for video source changes" into 13	9 years ago
Joshua Colp	222cee2410	Merge "main/bridge_channel: Fix channel reference leak on video source" into 13	9 years ago
Joshua Colp	814d8a99d1	Merge "bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source" into 13	9 years ago
Mark Michelson	9231a56cf3	Add API for channel frame deferral. ... There are several places in Asterisk that have duplicated logic for deferring important frames until later. This commit adds a couple of API calls to facilitate this automatically. ast_channel_start_defer_frames(): Future reads of deferrable frames on this channel will be deferred until later. ast_channel_stop_defer_frames(): Any frames that have been deferred get requeued onto the channel. ASTERISK-26343 Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641	9 years ago
Kevin Harwell	a3614d75f6	Revert "chan_sip: Fix lastrtprx always updated" ... This reverts commit 93332cb1d0. Unfortunately, the aforementioned commit caused a regression (incoming calls would eventually disconnect). Thus it is being removed. ASTERISK-26523 #close ASTERISK-25270 Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d	9 years ago
zuul	64e71fcf68	Merge "pjproject_bundled: Fix issue with libasteriskpj needing libresample" into 13	9 years ago
Alexander Anikin	a9ac1f5de4	chan_ooh323: Fixes to work right with Cisco devices ... Changed output packets queue processing algo to one read-one write instead of all read-all send Remove h.245 tunneling parameter from ReleaseComplete packet ASTERISK-24400 #close Reported by: Dmitry Melekhov Tested by: Dmitry Melekhov Change-Id: I0b31933b062a21011dbac9a82b8bcfe345f406f6	9 years ago
Alexander Anikin	0ee249075a	chan_ooh323: reset rrq count on gk registration ... reset registration attempts count on success registration on gatekeeper Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336	9 years ago
zuul	f02647ebbd	Merge "chan_ooh323: Fix infinite loop on read second part of H.225 packet" into 13	9 years ago
zuul	3a4baca35a	Merge "rtp_engine: Allow more than 32 dynamic payload types." into 13	9 years ago
Michael Kuron	59c23e1768	automon: restore mixing of the both channels after recording stops ... This is a regression over Asterisk 11, introduced by 2dc8a06006. Previously, recordings started via the automon DTMF code would automatically be mixed together using sox because app_monitor would be called with the m option. This commit restores this behavior. Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759	9 years ago
Matt Jordan	e79acaeb75	res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems ... Not surprisingly, using Respoke (and possibly other systems) it is possible to blow past the 16k limit for a WebSocket packet size. This patch bumps it up to 32k, which, at least for Respoke, is sufficient. For now. Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that matter), this patch adds a LOW_MEMORY directive that sets the buffer to 8k for systems who have asked for their reduced memory availability to be considered. Change-Id: Id235902537091b58608196844dc4b045e383cd2e	9 years ago
Matt Jordan	7a83196985	res_stasis: Set a video source mode on Stasis created bridges ... When a bridge is created via ARI (through res_stasis), no video source mode is set by default. As a result, any endpoint sending video media won't ever see any video reflected back to it. This patch defaults a bridge to a 'follow the talker' video mode. Further work can be done to add routes that allow for the video mode to be controlled through the /bridges resource. Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866	9 years ago
Matt Jordan	e7dc536b7a	main/bridge_channel: Fix channel reference leak on video source ... When a channel is made the video source, the bridge holds a reference to it. Whenever the video source changes, that reference is released. However, a ref leak does occur if the channel leaves the bridge (such as being hung up) while it is the video source, as the bridge never releases the ref in such a case. This patch adds a line to the bridge_channel_internal_join routine such that, when a channel finishes its time in the bridge, it notifies the bridge via ast_bridge_remove_video_src that if it is a video source its reference should be released. ASTERISK-26555 #close Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a	9 years ago
Matt Jordan	7c824b955d	main/bridge: Add some verbose logging for video source changes ... It's actually quite useful to see the source of a video stream change. This doesn't happen terribly often, even with talk detection - but when it does, it's nice to know which channel is now providing your video stream. As a verbose 5 level message, it shouldn't be terribly spammy or costly to have, and is 'lower level' then most other verbose messages that the bridge system emits. ASTERISK-26555 Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c	9 years ago
Matt Jordan	fd6af2dee8	bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source ... WebRTC clients really, really want to know the SSRC of the media they're getting. Changing the SSRC is generally not a good thing. bridge_softmix, starting in Asterisk 12, started changing the SSRC of parties as they joined or left the bridge. With most phones, this isn't a problem: phones just play back the stream they're getting. With WebRTC clients, however, the SSRC is tied to a media stream that may be negotiated. When a new SSRC just shows up, the media can be dropped. As it turns out, the SSRC change shouldn't even be necessary. From the perspective of the client, it's still talking to Asterisk with the same media stream: why indicate that the far party has suddenly changed to a different source of media? This patch opts to just remove the SSRC changes. With this patch, video clients that join/leave a softmix bridge actually get the video stream instead of freaking out. ASTERISK-26555 Change-Id: I27fec098b32e7c8718b4b65f3fd5fa73527968bf	9 years ago
Kevin Harwell	bd4d7d8ad0	stasis_recording/stored: remove calls to deprecated readdir_r function. ... The readdir_r function has been deprecated and should no longer be used. This patch removes the readdir_r dependency (replaced it with readdir) and also moves the directory search code to a more centralized spot (file.c) Also removed a strict dependency on the dirent structure's d_type field as it is not portable. The code now checks to see if the value is available. If so, it tries to use it, but defaults back to using the stats function if necessary. Lastly, for most implementations of readdir it *should* be thread-safe to make concurrent calls to it as long as different directory streams are specified. glibc falls into this category. However, since it is possible that there exist some implementations that are not safe, locking has been added for those other than glibc. ASTERISK-26412 ASTERISK-26509 #close Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba	9 years ago