4652 Commits (f6f4cf459f43f072604927209b39646f84aaa2e2)

Author	SHA1	Message	Date
Jacek Konieczny	6a9c18fb59	app_echo: forward and generate VIDUPDATE frames ... When using app_echo via WebRTC with VP8 video the video would appear only after a few minutes, because there would be nothing to request a full reference frame. This fixes the problem in both ways: - echos any VIDUPDATE frames received on the channel - sends one such frame when first video frame is to be forwarded This makes the echo work with Firefox and Chrome WebRTC implementation. ASTERISK-25867 #close Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e	9 years ago
Gianluca Merlo	c6e4c48e67	config: fix flags in uint option handler ... The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e	9 years ago
Andrew Nagy	d2c09ed73b	app_stasis: Don't hang up if app is not registered ... This prevents pbx_core from hanging up the channel if the app isn't registered. ASTERISK-25846 #close Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce	9 years ago
Walter Doekes	336cae73cc	app_chanspy: Fix occasional deadlock with ChanSpy and Local channels. ... Channel masquerading had a conflict with autochannel locking. When locking autochannel->channel, the channel is fetched from the autochannel and then locked. During the fetch, the autochannel -- which has no locks itself -- can be modified by someone who owns the channel lock. That means that the value of autochan->channel cannot be trusted until you hold the lock. In practice, this caused problems with Local channels getting masqueraded away while the ChanSpy attempted to get info from that channel. The old channel which was about to get removed got locked, but the new (replaced) channel got unlocked (no-op). Because the replaced channel was now locked (and would never get unlocked), it couldn't get removed from the channel list in a timely manner, and would now cause deadlocks when iterating over the channel list. This change checks the autochannel after locking the channel for changes to the autochannel. If the channel had been changed, the lock is reobtained on the new channel. In theory it seems possible that after this fix, the lock attempt on the old (wrong) channel can be on an already destroyed lock, maybe causing a crash. But that hasn't been observed in the wild and is harder induce than the current deadlock. Thanks go to Filip Frank for suggesting a fix similar to this and especially to IRC user hexanol for pointing out why this deadlock was possible and testing this fix. And to Richard for catching my rookie while loop mistake ;) ASTERISK-25321 #close Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def	9 years ago
Rodrigo Ramírez Norambuena	79dc5e2f00	app_queue: fix Calculate talktime when is first call answered ... Fix calculate of average time for talktime is wrong when is completed the first call beacuse the time for talked would be that call. ASTERISK-25800 #close Change-Id: I94f79028935913cd9174b090b52bb300b91b9492	9 years ago
Richard Mudgett	6f978fbfe5	app_confbridge: Only use b_profile options from the conference. ... A user cannot set new bridge options after the conference is created by the first user. Attempting to do so is documented as undefined behavior. This patch ensures that the bridge profile options used are from the conference and not what a subsequent user may have tried to set. Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266	9 years ago
Joshua Colp	c9e1c203b5	Merge "app_confbridge.c: Replace inlined code with existing function." into 13	9 years ago
Joshua Colp	2177dbea50	Merge topic 'ASTERISK-20987' into 13 ... * changes: app_confbridge: Add ability to get the muted conference state. app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. app_confbridge: Make non-admin users join a muted conference muted.	9 years ago
George Joseph	8804d0973c	build_system: Fix some warnings highlighted by clang ... Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd	9 years ago
Richard Mudgett	ed3c9c1512	app_confbridge.c: Replace inlined code with existing function. ... Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51	9 years ago
Richard Mudgett	1d0abf86e7	app_confbridge: Add ability to get the muted conference state. ... * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. * Added Muted header to AMI ConfbridgeListRooms action response list events to indicate the muted conference state. * Added Muted column to CLI "confbridge list" output to indicate the muted conference state and made the locked column a yes/no value instead of a locked/unlocked value. ASTERISK-20987 Reported by: hristo Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1	9 years ago
Richard Mudgett	f0d40afa69	app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. ... Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7	9 years ago
Richard Mudgett	3e51e5c7fd	app_confbridge: Make non-admin users join a muted conference muted. ... ASTERISK-20987 #close Reported by: hristo Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38	9 years ago
Martin Tomec	338a8ffed6	app_queue: Add member flag "in_call" to prevent reading wrong lastcall time ... Member lastcall time is updated later than member status. There was chance to check wrapuptime for available member with wrong (old) lastcall time. New boolean flag "in_call" is set to true right before connecting call, and reset to false after update of lastcall time. Members with "in_call" set to true are treat as unavailable. ASTERISK-19820 #close Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500	9 years ago
Rodrigo Ramírez Norambuena	e13719bff1	app_queue: Added reason pause of member ... In app_queue added value Paused Reason on QueueMemberStatus when a member on queue is paused and the reason was set. ASTERISK-25480 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e	9 years ago
Joshua Colp	937ba51659	Merge "voicemail: Move app_voicemail / res_mwi_external conflict to runtime" into 13	9 years ago
George Joseph	4ec85a9f07	voicemail: Move app_voicemail / res_mwi_external conflict to runtime ... The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247	9 years ago
Richard Mudgett	80a8b2a4cd	app_dial: Immediately exit dial if the caller is already hung up. ... If a caller hangs up before dial is executed within an AGI then the AGI has likely eaten all queued frames before executing the dial in DeadAGI mode. With the caller hung up and no pending frames from the caller's read queue, dial would not know that the call has hung up until a called channel answers. It is rather annoying to whoever just answered the non-existent call. Dial should not continue execution in DeadAGI mode, hangup handlers, or the h exten. * Added a check early in dial to abort dialing if the caller has hungup. ASTERISK-25307 #close Reported by: David Cunningham Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418	9 years ago
Dade Brandon	1d3d20dd68	app_amd: Correct documentation to reflect functionality ... Update documentation to reflect that maximum_number_of_words has functionality inconsistent with the variable name (and inconsistent with prior documentation.) Update documentation for silence_threshold, which previously implied that it was measuring time, rather than noise averages in the sample. Update the comments in amd.conf.sample. ASTERISK-25639 #close Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093	9 years ago
Carlos Oliva	ae428d8460	app_queue: update RT members when the 1st call joins a queue with no agents ... If a call enters on a queue and the members on that queue are updated in realtime (ex: using mysql inserting a new agent) the queue members are never refreshed and the call will stay in the queue until other event occurs. This happens only if this is the first call of the queue and there is no agents servicing. This patch prevent this issue, ensuring realtime members are updated if there is one call in the queue and no available agents ASTERISK-25442 #close Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682	9 years ago
Corey Farrell	ddf4dddf4f	app_meetme: Set default value for audio_buffers. ... The default value was never set for audio_buffers, causing bad audio quality. This ensures the default is always set. ASTERISK-25569 #close Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44	10 years ago
Joshua Colp	0843e6043e	Merge "Confbridge: Add a user timeout option" into 13	10 years ago
Mark Michelson	fdd2afcd16	Confbridge: Add a user timeout option ... This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98	10 years ago
Alec Davis	7debb986a5	app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! ... commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525) refer ASTERISK-24958 above commit removed ast_channel_lock(qe->chan); but failed to remove corresponding ast_channel_unlock(qe->chan); ASTERISK-25561 #close Reported Alec Davis Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a	10 years ago
Joshua Colp	3e0f161761	Merge "ast_format_cap_get_names: To display all formats, the buffer was increased." into 13	10 years ago
Alexander Traud	1bff400df7	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Walter Doekes	8410336681	docs: Fix a few typo's in app docs (more then, resourse). ... Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7	10 years ago
Matt Jordan	f6062ac616	Merge "app_queue.c: Force COLP update if outgoing channel name changed." into 13	10 years ago
Matt Jordan	9425eaa96d	Merge "app_queue.c: Factor out a connected line update routine." into 13	10 years ago
Matt Jordan	549d726065	Merge "app_dial.c: Make 'A' option pass COLP updates." into 13	10 years ago
Matt Jordan	5efb96520b	Merge "app_dial.c: Force COLP update if outgoing channel name changed." into 13	10 years ago
Joshua Colp	bc856bda64	Merge "app_dial.c: Factor out a connected line update routine." into 13	10 years ago
Richard Mudgett	6b1e7583c1	app_queue.c: Force COLP update if outgoing channel name changed. ... * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 #close Reported by: John Hardin Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172	10 years ago
Richard Mudgett	6bf304bf25	app_queue.c: Factor out a connected line update routine. ... Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3	10 years ago
Richard Mudgett	e36b5f1e8e	app_dial.c: Make 'A' option pass COLP updates. ... While the 'A' option is playing the announcement file allow the caller and peer to exchange COLP update frames. ASTERISK-25423 Reported by: John Hardin Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9	10 years ago
Richard Mudgett	747bfac895	app_dial.c: Force COLP update if outgoing channel name changed. ... * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 Reported by: John Hardin Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c	10 years ago
Richard Mudgett	14481d9aa0	app_dial.c: Factor out a connected line update routine. ... Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091	10 years ago
Richard Mudgett	bbeda190c3	app_dial.c: Remove some no-op code. ... Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54	10 years ago
Richard Mudgett	5f15cd93f0	app_page.c: Fix crash when forwarding with a predial handler. ... Page uses the async method of dialing with the dial API. When a call gets forwarded there is no calling channel available. If the predial handler was set then the calling channel could not be put into auto-service for the forwarded call because it doesn't exist. A crash is the result. * Moved the callee predial parameter string processing to before the string is passed to the dial API rather than having the dial API do it. There are a few benefits do doing this. The first is the predial parameter string processing doesn't need to be done for each channel called by the dial API. The second is in async mode and the forwarded channel is to have the predial handler executed on it then the non-existent calling channel does not need to be present to process the predial parameter string. * Don't start auto-service on a non-existent calling channel to execute the predial handler when the dial API is in async mode and forwarding a call. ASTERISK-25384 #close Reported by: Chet Stevens Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981	10 years ago
Kevin Harwell	aeddee39fb	app_record: RECORDED_FILE variable not being populated ... The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. This patch makes it so the variable is always set to the filename. ASTERISK-25410 #close Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653	10 years ago
Matt Jordan	efbd0f5a97	Merge "app_queue: AgentComplete event has wrong reason" into 13	10 years ago
Kevin Harwell	4fb95bbc4e	app_queue: AgentComplete event has wrong reason ... When a queued caller transfers an agent to another extension sometimes the raised AgentComplete event has a reason of "caller" and sometimes "transfer". Since a transfer has taken place this should always be transfer. This occurs because sometimes the stasis hangup event arrives before the transfer event thus writing a different reason out. With this patch, when a hangup event is received during a transfer it will check to see if the channel that is hanging up is part of a transfer. If so it will return and let the subsequently received transfer event handler take care of the cleanup. ASTERISK-25399 #close Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d	10 years ago
Kevin Harwell	6409e7b11a	app_queue: Crash when transferring ... During some transfer scenarios involving queues Asterisk would sometimes crash when trying to obtain a channel snapshot (could happen on caller or member channels). This occurred because the underlying channel had already disappeared when trying to obtain the latest snapshot. This patch adds a reference to both the member and caller channels that extends to the lifetime of the queue'd call, thus making sure the channels will always exist when retrieving the latest snapshots. ASTERISK-25185 #close Reported by: Etienne Lessard Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128	10 years ago
Richard Mudgett	bd867cd078	app_queue.c: Extract some functions for simpler code. ... * Extract set_queue_member_pause() from set_member_paused() for simpler and more consistent code. * Extract set_queue_member_ringinuse() from set_member_ringinuse_help_members() for simpler code. Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306	10 years ago
Richard Mudgett	ded51e3d77	app_queue.c: Fix error checking in QUEUE_MEMBER() read. ... Change-Id: I7294e13d27875851c2f4ef6818adba507509d224	10 years ago
Richard Mudgett	e5f5b9f384	app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'. ... Setting the 'paused' and 'ringinuse' options on a queue member using the dialplan function QUEUE_MEMBER did not behave the same way as the equivalent dialplan applications or AMI actions. * Made queue_function_mem_write() call the set_member_paused() and set_member_value() for the 'paused' and 'ringinuse' options respectively. A beneficial side effect is that the queue name is now optional and sets the value in all queues the interface is a member. * Update QUEUE_MEMBER XML documentation. * Fix error checking in QUEUE_MEMBER() write. ASTERISK-25215 #close Reported by: Lorne Gaetz Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb	10 years ago
Matt Jordan	0a1a550593	apps/app_dictate: Fix typo in attribution ... Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106	10 years ago
Matt Jordan	49f81ddb85	Makefile: Remove coverage files on 'make clean' ... This patch updates a variety of Makefiles in Asterisk's build system to remove .gcda and .gcno files when 'make clean' is executed. These files are generated when '--enable-coverage' is passed to the Asterisk configure script. Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602	10 years ago
Joshua Colp	e99e654d75	app_dial: Hold reference to calling channel formats when dialing outbound. ... Currently when requesting a channel the native formats of the calling channel are provided to the core for usage when dialing the outbound channel. This occurs without holding the channel lock or keeping a reference to the formats. This is problematic as the channel driver may end up changing the formats during this time. In the case of chan_sip this happens when an SDP negotiation completes. This change makes it so app_dial keeps a reference to the native formats of the calling channel which guarantees that they will remain valid for the period of time needed. ASTERISK-25172 #close Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db	10 years ago
Richard Mudgett	a2f4d03c87	app_directory: Fix crash when using the alias option 'a'. ... The voicemail.conf mailbox key/value pair is defined as: <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]] Where all fields in the value including the field values are optional. Since the parsing code for the mailbox key/value pair is sloppy, this patch tightens the parsing for the directory information. * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' respectively in search_directory_sub(). Those names make more sense. * Made sure that search_directory_sub() is dealing with the voicemail.conf mailbox options field if it even exists when looking for the 'hidefromdir' and 'alias' options. * Fix crash if a voicemail.conf mailbox is just <mailbox>=<password>,<name> when the 'a' option is used. If there were no fields after the name then the 'options' pointer was not checked for NULL. * Fix users.conf alias processing if the 'a' option is used. The wrong variable was used. ASTERISK-25087 #close Reported by: Chet Stevens Change-Id: I86052ea77307beddddba5279824d39dc0d593374	10 years ago