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Author	SHA1	Message	Date
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	9 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	9 years ago
Jenkins2	cf60a0c5bd	Merge "format_mp3: Don't try to build format_mp3 if we don't have sources"	9 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
Kevin Harwell	51375686f7	core/conversions: Added string to unsigned integer and long conversions ... Added functions that convert a string to an unsigned integer or unsigned long. A couple of unit test were also created to test the routines. The reasons for adding these conversion utilities (and hopefully eventually more) are as follows: * Conversion routines are functionally contained with consistent and better error checking * The function names offer a better description of what is happening * It encourages code reuse for easier bug fixing at a single source * It's simpler to use * It's unit testable For instance, currently in a lot of places when converting to an integer or similar the "sscanf" function is used. When using "sscanf" it may not be immediately clear what's happening as it lacks semantic naming. Limited error checking is usually done as well. For example, most of the time a check is done to make sure the value converted, but does not check for overflows or negative valued conversions when converting unsigned numbers. Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the built in error handling that "strtoul" has. For instance "strtoul" contains overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly more complex in its use. And maybe a bit controversial, but it may be ("big if") potentially slower than "strtoul" in some cases. Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb	9 years ago
Jenkins2	e74c48a46f	Merge "res_pjsip_session.c: Process initial INVITE sooner. (key exists)"	9 years ago
Joshua Colp	aa4c800060	Merge "Fix spelling queues.conf.sample file"	9 years ago
Joshua Colp	5a7af00e80	asterisk: Audit locking of channel when manipulating flags. ... When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10	9 years ago
Richard Mudgett	30fbed65f1	res_pjsip_session.c: Process initial INVITE sooner. (key exists) ... Retransmissions of an initial INVITE could be queued in the serializer before we have processed the first INVITE message. If the first INVITE message doesn't get completely processed before the retransmissions are seen then we could try to setup the same call from the retransmissions. A symptom of this is seeing a (key exists) message associated with an INVITE. An earlier change attempted to address this kind of problem by calculating a distributor serializer to use for unassociated messages. Part of that change also made incoming calls keep using that distributor serializer. (ASTERISK-26088) However, some leftover code was still deferring the INVITE processing to the session's serializer even though we were already in that serializer. This not only is unnecessary but would cause the same call resetup problem. * Removed the code to defer processing the initial INVITE to the session's serializer because we are already running in that serializer. ASTERISK-26998 #close Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6	9 years ago
Joshua Colp	094093b31d	Merge "chan_sip: Change sip_get_codec() to return correct codec list"	9 years ago
Rodrigo Ramírez Norambuena	6e7b78414f	Fix spelling queues.conf.sample file ... Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee	9 years ago
George Joseph	ce4d8dac91	Merge changes from topic 'sdp_api_adjustments' ... * changes: SDP: Make process possible multiple fmtp attributes per rtpmap. SDP: Explicitly stop a RTP instance before destoying it. SDP: Rework merge_capabilities(). SDP: Update ast_get_topology_from_sdp() to keep RTP map.	9 years ago
George Joseph	28d4e6be9b	Merge "SDP: Remove sdp_state.remote_capabilities"	9 years ago
Jenkins2	f09e079294	Merge "SDP: Add interface_address to specify our address to use."	9 years ago