29332 Commits (178d2bc7ef17c506b9fb9c119e494642adc4b38c)

 

All Branches   

Search

Author	SHA1	Message	Date
Jenkins2	da691efba8	Merge "format_mp3: Re-work menuselect/build issues" into 14	9 years ago
Joshua Colp	d3ee60522d	Merge "app_confbridge: Race between removing and playing name recording while leaving" into 14	9 years ago
Jenkins2	665cfd99dc	Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 14	9 years ago
Jenkins2	e0fbf43f30	Merge "test_json: Fix test names with reserved words" into 14	9 years ago
Sean Bright	7bdaed63fd	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	9 years ago
Joshua Colp	b86c8803fe	channel / app_meetme: Fix parentheses. ... ASTERISK-27025 Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed	9 years ago
Sean Bright	4e2b8471e8	stasis_recording: Correct ast_asprintf error checking ... ASTERISK-27021 #close Reported by: Tim Morgan Change-Id: I0ac061f040093e806c3b1f4e2340864f3ce4dd75	9 years ago
Sean Bright	ca2b3584b2	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	9 years ago
Sean Bright	c01f105e35	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	247b23c816	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
Joshua Colp	e5cd02bcb4	Merge "format_mp3: Don't try to build format_mp3 if we don't have sources" into 14	9 years ago
Joshua Colp	61e4fc5f2d	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	009e2cdb2e	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	4e757a4ca0	Merge "asterisk: Audit locking of channel when manipulating flags." into 14	9 years ago
George Joseph	93fd4cf412	Merge "res_agi: Fix malformed AGI usage response" into 14	9 years ago
Jenkins2	7c5529bcd1	Merge "res_agi: Prevent crash when SET VARIABLE called without arguments" into 14	9 years ago
George Joseph	d5805abf51	Merge "res_agi: Allow configuration of audio format of EAGI pipe" into 14	9 years ago
Jenkins2	3bcb9ffa3f	Merge "unittests: Add a unit test that causes a SEGV and..." into 14	9 years ago
Sean Bright	a7953f3089	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
Jenkins2	cb5e8fc3dc	Merge "res_agi: Clarify 'RECORD FILE' documentation" into 14	9 years ago
George Joseph	61b9b3fd11	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	e300431077	Merge "res_format_attr_h26x: Trim blanks in fmtp attributes" into 14	9 years ago
Jenkins2	d35c1b3add	Merge "app_queue: Fix members showing as being in call when not." into 14	9 years ago
Sean Bright	aaad4daf6d	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	9d93d4cbfa	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	143b758071	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	e08136b0dd	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	9b2eaa0b52	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	a4689c0576	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
Jenkins2	77dfb63f03	Merge "res_pjsip_session : fixed wrong From Header number On Re-invite" into 14	9 years ago
Jenkins2	491cbf8581	Merge "chan_sip: Better ICE handling for RTCP-MUX" into 14	9 years ago
Robert Mordec	5565b73a0f	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	9db41320b8	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	f487eb1c17	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
Jenkins2	2684e9c2ea	Merge "app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON" into 14	9 years ago
Yasin CANER	89826974ae	res_pjsip_session : fixed wrong From Header number On Re-invite ... ASTERISK-26964 #close Change-Id: I55a9caa7dc90e6c4c219cb09b5c2ec08af84a302	9 years ago
Joshua Colp	d219ecf808	Merge "res_hep_rtcp: Add support level to module info" into 14	9 years ago
George Joseph	703d6a4b1b	Merge "AST-2017-004: chan_skinny: Add EOF check in skinny_session" into 14	9 years ago
Jenkins2	17b23c4447	Merge "AST-2017-003: Handle zero-length body parts correctly." into 14	9 years ago
Mark Michelson	04c7bd61ce	AST-2017-003: Handle zero-length body parts correctly. ... ASTERISK-26939 #close Change-Id: I7ea235ab39833a187db4e078f0788bd0af0a24fd	9 years ago
George Joseph	86333e8079	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	83f6a6a2d6	AST-2017-002: Ensure transaction key buffer is large enough. ... ASTERISK-26938 #close Change-Id: I266490792fd8896a23be7cb92f316b7e69356413	9 years ago
Sean Bright	0a5265fe10	res_hep_rtcp: Add support level to module info ... Change-Id: I5661478f9cf12d431f730e42be79323b62831e92	9 years ago
Ivan Poddubny	933c3f028d	app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON ... There are 2 places in app_queue.c that log EXITEMPTY event: one in wait_our_turn, and another one in queue_exec in the loop trying to call an agent after wait_our_turn. In most cases it leads to logging EXITEMPTY twice. ABANDON is also logged on two places, and in the rare case when an agent and caller hang up simultaneously it's also possible to get duplicates in queue_log. This commit changes wait_our_turn to return -1 ("the caller should exit the queue") instead of 0 ("the caller's turn has arrived") in case of leaving when empty, so queue_exec skips the agent calling loop. Also, leave_queue is now executed only once in this case, because 2nd time is just a noop when the queue entry has already been removed. Also, it sets qe->handled to -1 to indicate that the call was not answered by an agent, but the necessary handling has already been done in order to avoid logging an extra ABANDON entry. ASTERISK-25665 #close Reported by: Ove Aursand Change-Id: I4578dd383bf2ac41589cf167865e8aaebcd4c11e	9 years ago
Jenkins2	17a7013221	Merge "res_pjsip_session.c: Process initial INVITE sooner. (key exists)" into 14	9 years ago
Rodrigo Ramírez Norambuena	fca75547e5	Fix spelling queues.conf.sample file ... Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee	9 years ago
Joshua Colp	ede77acaaf	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	d7461f22d9	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
Vitezslav Novy	bf19c0e596	chan_sip: Change sip_get_codec() to return correct codec list ... Return cahnnel nativeformats to fix bridge technology selection process. Same approach as in pjsip module. ASTERISK-26143 Reported-by: Henning Holtschneider Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48	9 years ago
Joshua Colp	85127e8f8d	Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages." into 14	9 years ago