4791 Commits (6b7697ed486fc3a8e5e7a72344437e66bd4ae507)

Author	SHA1	Message	Date
Joshua Colp	b05d2fda0c	Merge "app_queue: Fix locking behavior in stasis message handlers"	8 years ago
Sean Bright	c13ea6080e	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	8 years ago
Richard Mudgett	57656e2b5b	app_confbridge: Fix ConfbridgeTalking AMI event description. ... Thanks to Chris Howard for pointing this out on the wiki. Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705	8 years ago
Sean Bright	982d6173c5	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	8 years ago
Daniel Journo	60998371e3	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	8 years ago
frahaase	5b1796f59d	Binaural synthesis (confbridge): DTMF conference management. ... DTMF configuration options for the binaural softmix bridge: toggle binaural rendering (per channel). ASTERISK-26292 Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8	8 years ago
frahaase	094c26aa68	Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix. ... Adds binaural synthesis to bridge_softmix (via convolution using libfftw3). Binaural synthesis is conducted at 48kHz. For a conference, only one spatial representation is rendered. The default rendering is applied for mono-capable channels. ASTERISK-26292 Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf	8 years ago
Sean Bright	ab04a018e4	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	8 years ago
Sean Bright	b18f1bfb13	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	8 years ago
zuul	0b5a17082b	Merge "app_voicemail: Allow 'Comedian Mail' branding to be overriden"	8 years ago
zuul	9498ad8361	Merge "app_voicemail: VoiceMailPlayMsg did not play database stored messages"	8 years ago
Sean Bright	275f469a4d	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	8 years ago
zuul	4c79dff775	Merge "app_record: Add option to prevent silence from being truncated"	8 years ago
zuul	182c737353	Merge "cli: Fix various CLI documentation and completion issues"	8 years ago
rrittgarn	2b245b12d9	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	8 years ago
Sean Bright	662c9e69fa	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	8 years ago
Joshua Colp	72845bd4b5	Merge "core: Cleanup some channel snapshot staging anomalies."	8 years ago
zuul	c68e52d213	Merge "app_queue: reset abandoned in sl for sl2 calculations"	8 years ago
Sebastian Gutierrez	9f394d074a	app_queue: reset abandoned in sl for sl2 calculations ... ASTERISK-26775 #close Change-Id: I86de4b1a699d6edc77fea9b70d839440e4088284	8 years ago
Sean Bright	3f94373778	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	8 years ago
Sean Bright	0910773077	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	8 years ago
Richard Mudgett	16fdb11bc3	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	8 years ago
zuul	ada0032305	Merge "app_queue: Fix queues randomly disappearing on reload"	8 years ago
zuul	3eabae43c4	Merge "tests: use datadir for sound files"	8 years ago
kkm	8270d2436d	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	8 years ago
zuul	1479e049cb	Merge "app_queue: add RINGCANCELED log event on caller hang up"	8 years ago
Tzafrir Cohen	dbb9c8141d	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	8 years ago
Martin Tomec	40b9766a31	app_queue: add RINGCANCELED log event on caller hang up ... QueueLog did not log ringnoanswer when the caller abandoned call before first timeout. It was impossible to get agent membername and ringing duration for this short calls. After some discusions it seems that the best way is to add new event RINGCANCELED, which is generated after caller hangup during ringing. ASTERISK-26665 Change-Id: Ic70f7b0f32fc95c9378e5bcf63865519014805d3	8 years ago
Joshua Colp	fdf481636a	Merge "app_queue: Add QueueUpdate application."	8 years ago
Sebastian Gutierrez	8cc1cd5df7	app_queue: Add QueueUpdate application. ... Add an application that allows tracking outbound calls using app_queue. ASTERISK-19862 Change-Id: Ia0ab64aed934c25b2a25022adcc7c0624224346e	8 years ago
Sebastian Gutierrez	740ca862e4	app_queue: add new Service Level calculation ... Adds a new formula for SL2 and documentation ASTERISK-26559 Change-Id: I0970c620460507cd9d45b0d43600779c8915e770	8 years ago
Martin Tomec	f461f65dea	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	8 years ago
David Kerr	ddc951060a	app_originate: Add option to execute gosub prior to dial ... Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 that requested ability to add callerid into app_originate. Comments in that issue suggested that it was better solved by adding an option to gosub prior to originating the call. The attached patch implements this much like app_dial with two options one to gosub on the originating channel and one to gosub on the newly created channel and behaves just like app_dial. I have tested this patch by adding callerid info to the new channel and also SIPAddHeader (to e.g. add header to force auto answer) and confirmed it works. Have also tested both 'exten' and 'app' versions of app_originate. Opened by: dkerr Patch by: dkerr Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57	9 years ago
Joshua Colp	d3dba74017	Merge "Implement internal abstraction for iostreams"	9 years ago
Timo Teräs	070a51bf7c	Implement internal abstraction for iostreams ... fopencookie/funclose is a non-standard API and should not be used in portable software. Additionally, the way FILE's fd is used in non-blocking mode is undefined behaviour and cannot be relied on. This introduces internal abstraction for io streams, that allows implementing the desired virtualization of read/write operations with necessary timeout handling. ASTERISK-24515 #close ASTERISK-24517 #close Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85	9 years ago
Matt Jordan	cc86329228	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	4e8ab6cda9	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	4de5454ef1	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
zuul	0ec5abe592	Merge "Remove ASTERISK_REGISTER_FILE."	9 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Joshua Colp	95062fe220	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	9 years ago
Joshua Colp	e29e3a4f11	Merge "Binaural synthesis (confbridge): On/off setting for binaural synthesis."	9 years ago
frahaase	dce31f90ba	Binaural synthesis (confbridge): On/off setting for binaural synthesis. ... Adds setting to confbridge.conf (binaural_active) that determines if binaural synthesis can be available in bridge_softmix. ASTERISK-26292 Change-Id: I59dfcb8e55fe1df4ef32045882fea5bb58fc71db	9 years ago
Leandro Dardini	973e57d5ce	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	9 years ago
zuul	6f49ca5927	Merge "Audit ast_json_pack() calls for needed UTF-8 checks."	9 years ago
zuul	7947ee2894	Merge "app_queue.c: Fix clearing of pause reason string."	9 years ago
Richard Mudgett	9c49b96374	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	9 years ago
Richard Mudgett	c3bf1632cd	app_minivm.c: Fix malformed ast_json_pack() call. ... Change-Id: I082b239022fac462666e52a14a44304748908dc0	9 years ago
Richard Mudgett	9c54964dc5	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	9 years ago
George Joseph	86e8716952	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	9 years ago