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Author	SHA1	Message	Date
Joshua Colp	c2b3f6eb10	Merge changes from topic 'ASTERISK-26343' into 14 ... * changes: channel: Use frame deferral API for safe sleep. autoservice: Use frame deferral API AGI: Only defer frames when in an interception routine. Add API for channel frame deferral.	9 years ago
George Joseph	ea13d9c017	Merge "automon: restore mixing of the both channels after recording stops" into 14	9 years ago
Mark Michelson	1db9e7886c	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 (cherry picked from commit d149c4b9e0)	9 years ago
Mark Michelson	0288fba2f0	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 (cherry picked from commit 8ba3e2fc27)	9 years ago
Mark Michelson	8d8323b142	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
Mark Michelson	4a8b1940b8	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
Joshua Colp	5ad463581f	Merge "res_stasis: Don't unsubscribe from a NULL bridge." into 14	9 years ago
Joshua Colp	5873438150	Merge "chan_ooh323: reset rrq count on gk registration" into 14	9 years ago
Joshua Colp	4c8b3babf7	Merge "chan_ooh323: Fixes to work right with Cisco devices" into 14	9 years ago
Joshua Colp	1be6c9d26a	Merge "stasis_recording/stored: remove calls to deprecated readdir_r function." into 14	9 years ago
Joshua Colp	d7175c58b8	Merge "res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems" into 14	9 years ago
Joshua Colp	b19d7e79e9	Merge "res_stasis: Set a video source mode on Stasis created bridges" into 14	9 years ago
Joshua Colp	a4d145e465	Merge "main/bridge_channel: Fix channel reference leak on video source" into 14	9 years ago
Joshua Colp	51ff8e3899	Merge "main/bridge: Add some verbose logging for video source changes" into 14	9 years ago
Joshua Colp	a12c64cbba	Merge "bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source" into 14	9 years ago
Joshua Colp	7d966c5271	Merge "pjproject_bundled: Fix issue with libasteriskpj needing libresample" into 14	9 years ago
Joshua Colp	89286bf030	Merge "chan_ooh323: Fix infinite loop on read second part of H.225 packet" into 14	9 years ago
Joshua Colp	cfede6a1fc	res_stasis: Don't unsubscribe from a NULL bridge. ... A NULL bridge has special meaning in res_stasis for unsubscribing. It means that a subscription to ALL bridges should be removed. This should not be done as part of the normal subscription management in the res_stasis channel loop. ASTERISK-26468 Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0	9 years ago
Alexander Anikin	2849e596b8	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	e57f994237	chan_ooh323: reset rrq count on gk registration ... reset registration attempts count on success registration on gatekeeper Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336	9 years ago
Joshua Colp	ec5b3972ff	Merge "rtp_engine: Allow more than 32 dynamic payload types." into 14	9 years ago
Michael Kuron	0f627421b9	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	59d25b2081	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	d889fb375b	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	62d60c1f58	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	be254aa8e3	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	ff96e37509	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	25895897f8	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
Kevin Harwell	ac82b40bff	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
Alexander Anikin	e6b414a66a	chan_ooh323: Fix infinite loop on read second part of H.225 packet ... Fix logic on read second part of H.225 packet. There was infinite loop on wrong connections due to read before poll. Change-Id: I42b4bf75c46e4a5c5df5c5ca1f0bd74b8944e7ff	9 years ago
George Joseph	aa04c453b8	pjproject_bundled: Fix issue with libasteriskpj needing libresample ... libresample is only needed by pjproject if we're building pjsua, which we only do if TEST_FRAMEWORK is selected. It's required by pjsua to process audio which is needed by some testsuite tests. Unfortunately, pjproject relies on a newer version of libresample than the version that ships by most distros so we need to compile the version that's bundled with pjproject. Since we only need it for pjsua, we DON'T want it's symbols exposed when we actually build asterisk. There was a problem however... TEST_FRAMEWORK is only known AFTER we've already run ./configure on both asterisk and pjproject but pjproject's ./configure needs to test it to know whether to set up to build libresample or not. The previous way of figuring this out was to always tell ./configure "yes" but not actually build the library. This caused an issue where building libasteriskpj was being told to include libresample but it wasn't actually there. The solution is to still do a default pjproject configure during an asterisk ./configure but if makeopts or menuselect.makeopts changes subsequently, we now reconfigure pjproject, taking into account the current state of TEST_FRAMEWORK. Previously, if makeopts or menuselect.makeopts changed, only a recompile of pjproject was done. Change-Id: I9b5d84c61384a3ae07fe30e85c49698378cc4685	9 years ago
Joshua Colp	7d8b18d3fb	Merge "chan_sip: add missing account code" into 14	9 years ago
zuul	c573ace445	Merge "app_dial: Fix incorrect device state when channel is picked up." into 14	9 years ago
Sebastian Gutierrez	1012c28437	chan_sip: add missing account code ... Added missing account to AMI event of sip show peers ASTERISK-26176 #close Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482	9 years ago
Joshua Colp	abfb7b5d4a	Merge "res_pjsip_sdp_rtp: Limit number of formats to defined maximum." into 14	9 years ago
Joshua Colp	b2a078efc9	app_dial: Fix incorrect device state when channel is picked up. ... Given the scenario where multiple channels are dialed using Dial() but the caller is picked up using PickupChan() all outgoing channels except the channel specified to PickupChan() would be marked as ringing until the call had been hung up. When using the PickupChan application the channel executing the application is swapped into place of another channel. As part of this process the channel is answered. The Dial application has explicit logic which checks if the channel is answered, cancels all other outgoing channels, and bridges. This logic is different than the normal logic that is executed when an outgoing channel is answered. This different logic failed to publish dial events stating that the other outgoing channels had been canceled. As a result references to the outgoing channels were held onto by the dial masquerade process until the call had been ended and the channels had gone away. This would result in the channels appearing in the "core show channels" list despite not being present anymore and would also result in incorrect device state. This change makes it so that this logic also publishes dial events stating that the other outgoing channels have been canceled. ASTERISK-26549 Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f	9 years ago
Alexander Traud	6445f21caa	rtp_engine: Allow more than 32 dynamic payload types. ... Since adding all remaining rates of Signed Linear (ASTERISK-24274) and SILK (Gerrit 3136), only one RTP Payload Type is left in the dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964	9 years ago
Joshua Colp	946f55406f	Merge "define PATH_MAX for HURD" into 14	9 years ago
Joshua Colp	eaf04cb6d0	Merge "netsock.c: fix includes for HURD" into 14	9 years ago
Joshua Colp	96aaa6cbea	Merge "chan_sip: Incorrect display option Outbound reg. retry 403" into 14	9 years ago
zuul	afed2e75ae	Merge "res_pjsip_outbound_publish: Fix crash when publishing device state." into 14	9 years ago
zuul	5417fc9f65	Merge "bundled pjproject: Fix DNS write to freed memory." into 14	9 years ago
Joshua Colp	3b909277d6	Merge "pjproject_bundled: Fix compile of pjsua so it handles audio" into 14	9 years ago
zuul	0e55ec619e	Merge "codecs.conf.sample: Add sample and option descriptions for codec_opus" into 14	9 years ago
Joshua Colp	a7e22cf192	Merge "res/stasis: Add CLI commands for displaying/debugging ARI apps" into 14	9 years ago
Richard Mudgett	4fda9e7b0b	bundled pjproject: Fix DNS write to freed memory. ... PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS patch. The patch below fixes a write to freed memory under cartain DNS lookup conditions. 0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch ASTERISK-26516 Reported by: Richard Mudgett Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5	9 years ago
mkrokosz	d3e0d5d40b	res_pjsip_outbound_publish: Fix crash when publishing device state. ... While publishing device state between multiple instances of Asterisk, a crash will sporadically occur under high CPS which looks to be a race condition operating on the publisher queue. ASTERISK-26506 Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6	9 years ago
Tzafrir Cohen	2897fc9ab0	netsock.c: fix includes for HURD ... ASTERISK-25070 Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814	9 years ago
Tzafrir Cohen	3748e336ac	define PATH_MAX for HURD ... PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD define it to a constant. It is indeed not safe to assume there won't be longer paths and Asterisk generally does err safely on such cases. So even for HURD we'll just pretend PATH_MAX is 4096. ASTERISK-25070 #close Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3	9 years ago
Kevin Harwell	35f9d472ba	codecs.conf.sample: Add sample and option descriptions for codec_opus ... codecs.conf.sample was missing codec opus's configuration options, descriptions, and examples. This patch adds the configuration options and examples to codecs.conf.sample that can be used with codec_opus. ASTERISK-26538 #close Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b	9 years ago