29720 Commits (15fbcc74d8fd1b3655f4c34a280fe76665bd0c5c)

 

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Author	SHA1	Message	Date
Joshua Colp	8d70c081a4	Merge "format_mp3: Re-work menuselect/build issues"	8 years ago
Jenkins2	48d047ad5a	Merge "res_pjsip: New endpoint option "refer_blind_progress""	8 years ago
Jenkins2	77f7c1b98f	Merge "Sqlite3: make busy_timeout configurable."	8 years ago
Jenkins2	f166074524	Merge "test_json: Fix test names with reserved words"	8 years ago
Sean Bright	001f4ddda4	pbx_builtin: Properly handle hangup during Background ... Before this patch, when a user hung up during a Background, we would stuff 0xff into a char and attempt a dialplan lookup of it. This caused problems for some realtime engines which interpreted the value as the beginning of an invalid UTF-8 sequence. ASTERISK-19291 #close Reported by: Andrew Nowrot Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358	8 years ago
Joshua Colp	f6eeaaafd5	channel / app_meetme: Fix parentheses. ... ASTERISK-27025 Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed	8 years ago
Sean Bright	9dce4a947b	stasis_recording: Correct ast_asprintf error checking ... ASTERISK-27021 #close Reported by: Tim Morgan Change-Id: I0ac061f040093e806c3b1f4e2340864f3ce4dd75	8 years ago
Sean Bright	5c27fe2187	format: Reintroduce smoother flags ... In review 4843 (ASTERISK-24858), we added a hack that forced a smoother creation when sending signed linear so that the byte order was adjusted during transmission. This was needed because smoother flags were lost during the new format work that was done in Asterisk 13. Rather than rolling that same hack into res_rtp_multicast, re-introduce smoother flags so that formats can dictate their own options. Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16	8 years ago
Mark Michelson	39d14834f8	Confbridge: Add "sfu" video mode to bridge profile options. ... A previous commit added plumbing to bridge_softmix to allow for an SFU experience with Asterisk. This commit adds an option to app_confbridge that allows for a confbridge to actually make use of the SFU video mode. SFU mode is implemented in a "set it and forget it" kind of way. That is, when the bridge is created, if SFU mode is enabled, then the video mode gets set to SFU and cannot be changed. Future improvements may allow for a hybrid experience (e.g. forward multiple video streams, specifically those of the most recent talkers), but for this addition, no such capability is present. Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020	8 years ago
Mark Michelson	2da869408a	Add primitive SFU support to bridge_softmix. ... This sets up the "plumbing" in bridge_softmix to be able to accommodate Asterisk asking as an SFU (selective forwarding unit) for conferences. The way this works is that whenever a channel enters or leaves a conference, all participants in the bridge get sent a stream topology change request. The topologies consist of the channels' original topology, along with video destination streams corresponding to each participants' source video streams. So for instance, if Alice, Bob, and Carol are in the conference, and each supplies one video stream, then the topologies for each would look like so: Alice: Audio, Source video(Alice), Destination Video(Bob), Destination video (Carol) Bob: Audio, Source video(Bob) Destination Video(Alice), Destination video (Carol) Carol: Audio, Source video(Carol) Destination Video(Alice), Destination video (Bob) This way, video that arrives from a source video stream can then be copied out to the destination video streams on the other participants' channels. Once the bridge gets told that a topology on a channel has changed, the bridge constructs a map in order to get the video frames routed to the proper destination streams. This is done using the bridge channel's stream_map. This change is bare-bones with regards to SFU support. Some key features are missing at this point: * Stream limits. This commit makes no effort to limit the number of streams on a specific channel. This means that if there were 50 video callers in a conference, bridge_softmix will happily send out topology change requests to every channel in the bridge, requesting 50+ streams. * Configuration. The plumbing has been added to bridge_softmix, but there has been nothing added as of yet to app_confbridge to enable SFU video mode. * Testing. Some functions included here have unit tests. However, the functionality as a whole has only been verified by hand-tracing the code. * Selectivenss. For a "selective" forwarding unit, this does not currently have any means of being selective. * Features. Presumably, someone might wish to only receive video from specific sources. There are no external-facing functions at the moment that allow for users to select who they receive video from. * Efficiency. The current scheme treats all video streams as being unidirectional. We could be re-using a source video stream as a desetnation, too. But to simplify things on this first round, I did it this way. Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d	8 years ago
Sean Bright	045d7b8cb7	format_mp3: Re-work menuselect/build issues ... Rather than removing format_mp3 from ALL_C_MODS (which caused format_mp3 to not show up in menuselect), use .PHONY targets when the necessary source files are not present. ASTERISK-23951 Reported by: Tzafrir Cohen Change-Id: I0a7512c51acc9e86043671795020b0de725bd9e8	8 years ago
George Joseph	80206cdc65	test_json: Fix test names with reserved words ... Some of the test names were actually reserved words (true, false, int, null, string, bool). When the jenkins test results analyzer does its thing it tries to create a map using the test names as keys and fails because they're reserved words. Added "type_" to those test names. Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b	8 years ago
Jenkins2	cf60a0c5bd	Merge "format_mp3: Don't try to build format_mp3 if we don't have sources"	8 years ago
Joshua Colp	9c4f63263c	manager: Clear the flag on the other channel. ... During the channel flag audit an incorrect change was done. The flag should be cleared on the second channel. ASTERISK-26469 Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8	9 years ago
Sean Bright	1f136fe885	res_srtp: Add support for libsrtp2 ... ASTERISK-25294 #close Reported by: Tzafrir Cohen ASTERISK-26976 #close Reported by: Alex Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40	9 years ago
Jenkins2	56b6a71548	Merge "asterisk: Audit locking of channel when manipulating flags."	9 years ago
George Joseph	6bb3cedb1b	Merge "res_agi: Prevent crash when SET VARIABLE called without arguments"	9 years ago
George Joseph	366b10a500	Merge "res_agi: Allow configuration of audio format of EAGI pipe"	9 years ago
Jenkins2	915acf1e5d	Merge "res_agi: Fix malformed AGI usage response"	9 years ago
Jenkins2	24318aac45	Merge "unittests: Add a unit test that causes a SEGV and..."	9 years ago
Sean Bright	59348aa182	format_mp3: Don't try to build format_mp3 if we don't have sources ... ASTERISK-23951 #close Reported by: Tzafrir Cohen Change-Id: Iebf181d44bb735787fde4b5be863c4d7e2478a30	9 years ago
Martin Tomec	44c5a144ce	Sqlite3: make busy_timeout configurable. ... Enables runtime configuration of busy_timeout for sqlite databases. Default timeout remains 1000ms. ASTERISK-27014 #close Change-Id: I8921a3aac3c335843be4cb17d2dd0a5c157a36da	9 years ago
Jenkins2	d6992daaef	Merge "res_agi: Clarify 'RECORD FILE' documentation"	9 years ago
George Joseph	08edd54c1b	unittests: Add a unit test that causes a SEGV and... ... ...that can only be run by explicitly calling it with 'test execute category /DO_NOT_RUN/ name RAISE_SEGV' This allows us to more easily test CI and debugging tools that should do certain things when asterisk coredumps. To allow this a new member was added to the ast_test_info structure named 'explicit_only'. If set by a test, the test will be skipped during a 'test execute all' or 'test execute category ...'. Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed	9 years ago
Joshua Colp	2e7866ebb7	Merge "app_queue: Add QUEUE_RAISE_PENALTY feature"	9 years ago
Jenkins2	c247625012	Merge "chan_sip: Better ICE handling for RTCP-MUX"	9 years ago
Jenkins2	cd0e6a2324	Merge "res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm"	9 years ago
Jenkins2	2200009099	Merge "res_format_attr_h26x: Trim blanks in fmtp attributes"	9 years ago
Sean Bright	d847fe6585	res_agi: Allow configuration of audio format of EAGI pipe ... This change allows the format of the EAGI audio pipe to be changed by setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of the loaded formats. ASTERISK-26124 #close Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd	9 years ago
Sean Bright	e2e6baa8d8	res_agi: Clarify 'RECORD FILE' documentation ... Documented the 'beep' option in both the parameters list and the command description. ASTERISK-23839 #close Change-Id: I4970395c922dbdce3f7cf0f56d5b065ec9aa53ea	9 years ago
Sean Bright	3dcb3c88aa	res_agi: Prevent crash when SET VARIABLE called without arguments ... Explicitly check that the appropriate number of arguments were passed to SET VARIABLE before attempting to reference them. Also initialize the arguments array to zeroes before populating it. ASTERISK-22432 #close Change-Id: I5143607d80a2724f749c1674f3126b04ed32ea97	9 years ago
Sean Bright	e490aa3176	res_agi: Fix malformed AGI usage response ... If the generated XML documentation for a command does not end with a \n, the postamble of the usage message does not appear on its own line. ASTERISK-25662 #close Change-Id: If190f1e9e37fe215fed95897d78d4a6e142b0020	9 years ago
Sean Bright	8ae0227cf3	res_format_attr_h26x: Trim blanks in fmtp attributes ... Some devices separate format attributes with a semicolon followed by a space, so trim blanks before trying to match them. ASTERISK-27008 #close Change-Id: Ia44cb2e4fef5c73dc541a29da79cb0e19c22d9cc	9 years ago
Joshua Colp	faab058014	app_queue: Fix members showing as being in call when not. ... A change was done which added an 'in_call' flag to queue members that was set to true while talking to an agent. Unfortunately in practice this does not accurately reflect whether they are talking to an agent or not. If a Local channel is involved and a transfer is performed then the app_queue application would incorrectly think the agent was still in a call with the caller. This was done to fix a race condition between an agent becoming available by device state and the checking of the last call information for the wrapup time. There was a small window where the last call information would be the previous value instead of the new one. This change goes about fixing the original issue in a different way by considering the call completed if device state is received which would make the agent available and if they are currently in a call. If this occurs the last call information is updated before the agent becomes available ensuring that old information is not present when checking if the member should be called. This also improves the transfer situation by actually updating and enforcing the wrapup time. ASTERISK-26399 ASTERISK-26400 ASTERISK-26715 ASTERISK-26975 Change-Id: Ife1cb686e3173b3a6d368601adef9aff69d4beea	9 years ago
Joshua Colp	dece2eb892	Merge "res_pjsip_session : fixed wrong From Header number On Re-invite"	9 years ago
Robert Mordec	36e90952ec	app_confbridge: Race between removing and playing name recording while leaving ... When user leaves a conference, its channel calls async_play_sound_file() in order to play the name announcement and then unlinks the sound file. The async_play_sound_file() function adds a task to conference playback queue, which then runs playback_common() function in a different thread. It leads to a race condition when, in some cases, channel thread may unlink the sound file before playback_common() had a chance to open it. This patch creates a file deletion task, that is queued after playback. ASTERISK-27012 #close Change-Id: I412f7922d412004b80917d4e892546c15bd70dd3	9 years ago
Kevin Harwell	440ff38c08	res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm ... When using rtcp mux if an rtcp payload came in it would still use the srtp unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp data was being passed to the rtp unprotect method this would result in an error. This patch ensures that the correct unprotect method is chosen by making sure the passed in rtcp flag is appropriately set when rtcp mux is enabled and an rtcp payload is received. ASTERISK-26979 #close Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241	9 years ago
Sean Bright	0f487978a9	chan_sip: Better ICE handling for RTCP-MUX ... If we are offered or are offering RTCP-MUX, don't consider RTCP ICE candidates. This confuses certain browsers (current Firefox for example) and causes intial audio setup delays. ASTERISK-26982 #close Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91	9 years ago
Steve Davies	be4beff3e4	app_queue: Add QUEUE_RAISE_PENALTY feature ... Additional variable to work alongside QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY, including an extra parameter in queuerules.conf. This value causes lower Agent penalty values to "raise up" so that they can join higher penalty agents and be treated equally after a period of time. ASTERISK-26995 #close Change-Id: If1c6421a983667a5ac4c359f6dac25b212b4c459	9 years ago
Joshua Colp	95c6b98acf	Merge "app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON"	9 years ago
Joshua Colp	8a83b473f1	Merge "app_stream_echo: Added a multi-stream echo application"	9 years ago
Jenkins2	5fffb96736	Merge "core/conversions: Added string to unsigned integer and long conversions"	9 years ago
Jenkins2	b5570de7b4	Merge "res_hep_rtcp: Add support level to module info"	9 years ago
Jenkins2	7631988c3d	Merge "AST-2017-004: chan_skinny: Add EOF check in skinny_session"	9 years ago
Jenkins2	79c7067c5e	Merge "AST-2017-003: Handle zero-length body parts correctly."	9 years ago
Mark Michelson	7c0466092c	AST-2017-003: Handle zero-length body parts correctly. ... ASTERISK-26939 #close Change-Id: I7ea235ab39833a187db4e078f0788bd0af0a24fd	9 years ago
George Joseph	949e9147bf	AST-2017-004: chan_skinny: Add EOF check in skinny_session ... The while(1) loop in skinny_session wasn't checking for EOF so a packet that was longer than a header but still truncated would spin the while loop infinitely. Not only does this permanently tie up a thread and drive a core to 100% utilization, the call of ast_log() in such a tight loop eats all available process memory. Added poll with timeout to top of read loop ASTERISK-26940 #close Reported-by: Sandro Gauci Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898	9 years ago
Mark Michelson	2bb98d8fac	AST-2017-002: Ensure transaction key buffer is large enough. ... ASTERISK-26938 #close Change-Id: I266490792fd8896a23be7cb92f316b7e69356413	9 years ago
Sean Bright	4141748e85	res_hep_rtcp: Add support level to module info ... Change-Id: I5661478f9cf12d431f730e42be79323b62831e92	9 years ago
Kevin Harwell	a60d1f3974	app_stream_echo: Added a multi-stream echo application ... If the channel does not have multi-stream support then this application acts just like app_echo. If it does have multi-stream support then each stream is echoed back to itself (one-to-one). If a "num" is specified, then a new topology is made that contains clones (from the channel's topology) of all media types that are not equal to the given "type". If the media type differs then the first stream matching the "type" is cloned into the new topology and then up to "num" - 1 of the same stream are also cloned into it. Any additional streams from the original topology matching the "type" are subsequently ignored (i.e. not added to the new topology). For this same case when a frame is read from a stream that frame is still echoed back like before, but now that frame is also echoed out to the additional streams that matched on the specified "type". ASTERISK-26997 #close Change-Id: I254144486734178e196c7f590a26ffc13543ff2c	9 years ago