4780 Commits (04a30e28a3ac8f29ac7967c077be621ad02ae47b)

Author	SHA1	Message	Date
George Joseph	cc668bd522	modules: change module LOAD_FAILUREs to LOAD_DECLINES ... In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25	9 years ago
zuul	e528c2fc18	Merge "Revert "app_queue: Handle the caller being redirected out of a queue bridge"" into 14	9 years ago
zuul	db246cd87e	Merge "app_queue: Member stuck as pending after forwarding previous call from queue" into 14	9 years ago
Joshua Colp	d36a260fcf	Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references." into 14	9 years ago
Sean Bright	b7a2d271c1	Revert "app_queue: Handle the caller being redirected out of a queue bridge" ... This reverts commit 163e9e53dc. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b	9 years ago
Joshua Colp	117092efac	Merge "app_queue: Fix locking behavior in stasis message handlers" into 14	9 years ago
Robert Mordec	6ae6e16468	app_queue: Member stuck as pending after forwarding previous call from queue ... Queue member will get stuck in pending_members if queue calls a device that is different from the one observed for state changes. This patch removes members from pending_members as a result of channel stasis events such as blind or attended transfers and hangup. ASTERISK-26862 #close Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727	9 years ago
Sean Bright	e8764a93cb	app_queue: Fix locking behavior in stasis message handlers ... The queue_stasis_data structure contains various mutable fields that require appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and 'caller_uniqueid' fields need to be locked when read from or written to. Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088	9 years ago
Richard Mudgett	e7ed625d23	app_confbridge: Fix ConfbridgeTalking AMI event description. ... Thanks to Chris Howard for pointing this out on the wiki. Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705	9 years ago
Richard Mudgett	dcc2d8df0d	autochan/mixmonitor/chanspy: Fix unsafe channel locking and references. ... Dereferencing struct ast_autochan.chan without first calling ast_autochan_channel_lock() is unsafe because the pointer could change at any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() itself uses struct ast_autochan.chan unsafely and can result in a deadlock if the original channel happens to get destroyed after a masquerade in addition to the pointer getting changed. The problem is more likely to happen with v11 and earlier because masquerades are used to optimize out local channels on those versions. However, it could still happen on newer versions if the channel is executing a dialplan application when the channel is transferred or redirected. In this situation a masquerade still must be used. * Added a lock to struct ast_autochan to safely be able to use ast_autochan.chan while trying to get the channel lock in ast_autochan_channel_lock(). The locking order is the channel lock then the autochan lock. Locking in the other direction requires deadlock avoidance. * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. * Fix unsafe ast_autochan.chan usages in app_chanspy.c. * app_chanspy.c: Removed unused autochan parameter from next_channel(). ASTERISK-26867 Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592	9 years ago
Sean Bright	e0767d7a6d	app_queue: Handle the caller being redirected out of a queue bridge ... A caller can leave the Queue() application after being bridged with a member in a few ways: * Caller or member hangup * Caller is transferred somewhere else (blind or atx) * Caller is externally redirected elsewhere The first 2 scenarios are currently handled by subscribing to stasis messages, but the 3rd is not explicitly covered. If a caller is redirected away from the Queue() application, the member who was last bridged with that caller will remain in an "In use" state until the caller hangs up. This patch adds handling of the caller leaving the queue via redirection. We monitor the caller-member bridge, and if the caller is the one that leaves, we treat it the same as we would a caller hangup. ASTERISK-26400 #close Reported by: Etienne Lessard Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334	9 years ago
Daniel Journo	893628abdd	app_voicemail: Cannot set fromstring on a per-mailbox basis ... * apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe	9 years ago
Sean Bright	bca1462d90	realtime: Fix ast_load_realtime_multientry handling ... ast_load_realtime_multientry() returns an ast_config structure whose ast_categorys are keyed with the empty strings. Several modules were giving semantic meaning to the category names causing problems at runtime. * app_directory: Treated the category name as the mailbox name, and would fail to direct calls to the appropriate extension after an entry was chosen. * app_queue: Queues, queue members, and queue rules were all affected and needed to be updated. * pbx_realtime: Pattern matching would never succeed because the extension entered by the user was always compared to the empty string. Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7	9 years ago
Sean Bright	d1196fd39e	app_voicemail: vm_authenticate accesses uninitialized memory ... vm_authenticate doesn't always set the passed ast_vm_user argument, so we initialize to 0 before passing it in. ASTERISK-25893 #close Reported by: Filip Jenicek Change-Id: Ia3cc0128f93d352ed9add8d5c2f0f7232c2cbe4a	9 years ago
zuul	6db4ac4438	Merge "app_voicemail: Allow 'Comedian Mail' branding to be overriden" into 14	9 years ago
zuul	a65a73a1ef	Merge "app_voicemail: VoiceMailPlayMsg did not play database stored messages" into 14	9 years ago
Sean Bright	23ee13dcd1	app_voicemail: Allow 'Comedian Mail' branding to be overriden ... Original patch by John Covert, slight modifications by me. ASTERISK-17428 #close Reported by: John Covert Patches: app_voicemail.c.patch (license #5512) patch uploaded by John Covert Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6	9 years ago
zuul	40dbbaebe3	Merge "app_record: Add option to prevent silence from being truncated" into 14	9 years ago
zuul	18ec2919cf	Merge "cli: Fix various CLI documentation and completion issues" into 14	9 years ago
rrittgarn	9296a728a9	app_voicemail: VoiceMailPlayMsg did not play database stored messages ... When attempting to use VoiceMailPlayMsg with a realtime data backend the message is located, but never retrieved. This patch adds the required RETRIEVE and DISPOSE calls that will fetch the message from the database (and IMAP storage as well for that matter). Also, removed extraneous make_file call. ASTERISK-26723 #close Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c	9 years ago
Sean Bright	fa8a1eff2f	app_record: Add option to prevent silence from being truncated ... When using Record() with the silence detection feature, the stream is written out to the given file. However, if only 'silence' is detected, this file is then truncated to the first second of the recording. This patch adds the 'u' option to Record() to override that behavior. ASTERISK-18286 #close Reported by: var Patches: app_record-1.8.7.1.diff (license #6184) patch uploaded by var Change-Id: Ia1cd163483235efe2db05e52f39054288553b957	9 years ago
zuul	4ce60b8419	Merge "core: Cleanup some channel snapshot staging anomalies." into 14	9 years ago
Sean Bright	dc8dd56684	cli: Fix various CLI documentation and completion issues ... * app_minivm: Use built-in completion facilities to complete optional arguments. * app_voicemail: Use built-in completion facilities to complete optional arguments. * app_confbridge: Add missing colons after 'Usage' text. * chan_alsa: Use built-in completion facilities to complete optional arguments. * chan_sip: Use built-in completion facilities to complete optional arguments. Add completions for 'load' for 'sip show user', 'sip show peer', and 'sip qualify peer.' * chan_skinny: Correct and extend completions for 'skinny reset' and 'skinny show line.' * func_odbc: Correct completions for 'odbc read' and 'odbc write' * main/astmm: Use built-in completion facilities to complete arguments for 'memory' commands. * main/bridge: Correct completions for 'bridge kick.' * main/ccss: Use built-in completion facilities to complete arguments for 'cc cancel' command. * main/cli: Add 'all' completion for 'channel request hangup.' Correct completions for 'core set debug channel.' Correct completions for 'core show calls.' * main/pbx_app: Remove redundant completions for 'core show applications.' * main/pbx_hangup_handler: Remove unused completions for 'core show hanguphandlers all.' * res_sorcery_memory_cache: Add completion for 'reload' argument of 'sorcery memory cache stale' and properly implement. Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca	9 years ago
Sean Bright	c74af2951b	manager: Restore Originate failure behavior from Asterisk 11 ... In Asterisk 11, if the 'Originate' AMI command failed to connect the provided Channel while in extension mode, a 'failed' extension would be looked up and run. This was, I believe, unintentionally removed in 51b6c49. This patch restores that behavior. This also adds an enum for the various 'synchronous' modes in an attempt to make them meaningful. ASTERISK-26115 #close Reported by: Nasir Iqbal Change-Id: I8afbd06725e99610e02adb529137d4800c05345d	9 years ago
Richard Mudgett	f88b598ca2	core: Cleanup some channel snapshot staging anomalies. ... We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	9 years ago
zuul	bb1e9e1b58	Merge "app_queue: Fix queues randomly disappearing on reload" into 14	9 years ago
kkm	4ae2ff5f21	app_queue: Fix queues randomly disappearing on reload ... With 500+ queues and a reload every minute, a random queue disappears upon reload. The cause is mususe of the 'dead' flag. Namely, all queues were marked dead up front, and then "resurrected" by dropping this flag for those found in the configuration. But a queue marked dead can be removed also when control leaves the app entry point on a PBX thread. With this change, the queue is marked only not found, and at the end of reload only the queues that are still not found are actually marked as dead, so the dead flag is never reset, and set only on positively dead queues. ASTERISK-26755 Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf	9 years ago
Tzafrir Cohen	c06d9d7717	tests: use datadir for sound files ... Some (voicemail-related) tests API symlinks beep.gsm and other files from ast_config_AST_VAR_DIR. It should use ast_config_AST_DATA_DIR. ASTERISK-26740 #close Change-Id: Id49c56fb9e16df64b1a2b829693ca7601252df89	9 years ago
Martin Tomec	2840731d81	app_queue: Ensure member is removed from pending when hanging up. ... In some cases member is added to pending_members, and the channel is hung up before any extension state change. So the member would stay in pending_members forever. So when we call do_hang, we should also remove member from pending. ASTERISK-26621 #close Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54	9 years ago
Matt Jordan	46bedcbbad	apps/app_echo: Only relay a single video source change frame ... In 9785e8d0, app_echo was updated to relay video source updates to the channel for the purposes of displaying video in WebRTC tests. Unfortunately, this can cause a Kafkaesque nightmare if two or more Local channels are in a bridge together where their ends are in app_echo. When this situation occurs, a video update sent into app_echo will cause the video update to be relayed to the other Local channels, causing another round of video updates, etc. In not much time at all, the channel length queues will be overwhelmed, channel alert pipes will fail, and all hell will break loose as Asterisk merrily continues to throw more video update requests onto the channels. This patch updates app_echo to *only* relay a single video update. Once a video update has been made, all further video updates are dropped. This meets the intended purpose of the original patch: if we get a video update and we're in app_echo, go ahead and ask the sender to update themselves. However, once we've got that video stream sync'd up, don't keep spamming the world. Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74	9 years ago
Sebastian Gutierrez	abd41590d7	app_queue: new variable set when abandoned ... sets the variable ABANDONED to TRUE if the call was not answered. ASTERISK-26558 Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3	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
Joshua Colp	ebc293e609	app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. ... When executing the MailboxExists dialplan application and MAILBOX_EXISTS dialplan function the passed in temporary voice mailbox was not cleared, causing it to try to free garbage. ASTERISK-26503 #close Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3	10 years ago
Leandro Dardini	ef8c54238c	app_queue: Added initialization for "context" parameter ... When using Asterisk Realtime Architecture, empty fields are skipped and the default values are used. If the "context" parameter in queue was set and then cleared from the database, the old value remains in memory and it continues to be used. This change initialize the "context" parameter with an empty value, allowing clearing the parameter. ASTERISK-26462 #close Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905	10 years ago
Joshua Colp	839766f5e1	Merge "Audit ast_json_pack() calls for needed UTF-8 checks." into 14	10 years ago
zuul	22094b267d	Merge "app_queue.c: Fix clearing of pause reason string." into 14	10 years ago
Richard Mudgett	aba27b5a60	Audit ast_json_pack() calls for needed UTF-8 checks. ... Added needed UTF-8 checks before constructing json objects in various files for strings obtained outside the system. In this case string values from a channel driver's peer and not from the user setting channel variables. * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json object construction. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096	10 years ago
Richard Mudgett	857a9936f4	app_minivm.c: Fix malformed ast_json_pack() call. ... Change-Id: I082b239022fac462666e52a14a44304748908dc0	10 years ago
Richard Mudgett	030852e1b3	app_queue.c: Fix clearing of pause reason string. ... The pause reason is not always cleared when it should be cleared. * Made set_queue_member_pause() always clear pause reason if not pausing with a reason string. Change-Id: I993dad19626ec017478a230e980989438b778c53	10 years ago
George Joseph	90f8ba8800	app_dial: Add the "Q" option to set the cause on unanswered channels ... The "Q" option will set the cause on the unanswered channels when another channel answers. It overrides the default of ANSWERED_ELSEWHERE. NOTE: chan_sip does not support setting the cause on a CANCEL to anything other than ANSWERED_ELSEWHERE. ASTERISK-26446 #close Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47	10 years ago
Richard Mudgett	d910a51b33	app_queue: Fix CLI "queue show" and AMI Queues action output truncation. ... The output of CLI "queue show" and AMI Queues action is truncated and "failed to extend from 240 to 327" messages are generated if the queue member and interface names are lengthy. * Increase the string buffer size from 240 to 512 in order to accommodate for more information fields added to the output since v1.8. ASTERISK-26360 #close Reported by: Richard Mudgett Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d	10 years ago
zuul	c396565672	Merge "ConfBridge: Make some announcements asynchronous." into 14	10 years ago
zuul	71e36845bd	Merge "apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option" into 14	10 years ago
zuul	46d5691aa4	Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5" into 14	10 years ago
Mark Michelson	999b34301b	ConfBridge: Make some announcements asynchronous. ... Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a	10 years ago
zuul	f33ccbdf86	Merge "app_mp3: Use correct buffer size and the same sample rate as the channel" into 14	10 years ago
Matt Jordan	3e054a3826	apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option ... In any scenario in which the callee is not connected to the caller, the current code in app_dial will crash due to raising a Dial End Stasis Message after the callee channel has been hung up. This patch corrects the error by simply moving the explicit hangup of the callee (peer) channel until after the dial end message. ASTERISK-25691 #close Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d	10 years ago
Matt Jordan	15105c7494	apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 ... If the callee selects option '5' using the Dial application's privacy (P) option, the DIALSTATUS is erroneously set to ANSWER. This option reflects the callee sending the caller to VoiceMail one time; the call is definitely *not* ANSWERed in such a scenario. With this patch, the DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that is set when the 'send to VoiceMail every time' option is set. ASTERISK-25691 Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358	10 years ago
Michael Kuron	dd408708c9	app_mp3: Use correct buffer size and the same sample rate as the channel ... Previously, the buffer used for MP3 streamed from HTTP servers had a size of 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 minute. Only when the buffer is full does audio start to play. For MP3 files streamed from a server, that is usually not a big deal as long as the connection to the server is fast enough to supply that much data within a second or two. For MP3 live streams however, it takes 1 minute to download 1 minute of audio, so without this change, app_mp3 wasn't really usable for MP3 live streams. This commit changes the buffer size so that it covers 6 seconds of an MP3 file streamed from a server and 0.5 seconds of an MP3 live stream. The latter is identified by the use of a .m3u file extension. app_mp3 so far only supported 8 kHz audio. Now it always runs at the sample rate of the channel. ASTERISK-26085 #close Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0	10 years ago
zuul	7402b7ceb4	Merge "followme: initialize all config items on reload" into 14	10 years ago