4679 Commits (aadc58a1e7b333ba400e5d38513677d4dee10cf2)

Author	SHA1	Message	Date
Rodrigo Ramírez Norambuena	15aeb78c66	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	1bc54aee80	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	0aea293450	Merge "app_confbridge.c: Replace inlined code with existing function."	9 years ago
Joshua Colp	e541d9cf34	Merge topic 'ASTERISK-20987' ... * 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
Richard Mudgett	af6b15976d	app_confbridge.c: Replace inlined code with existing function. ... Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51	9 years ago
Richard Mudgett	7932336a3d	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	894045e7cf	app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. ... Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7	9 years ago
Richard Mudgett	12c93e8f81	app_confbridge: Make non-admin users join a muted conference muted. ... ASTERISK-20987 #close Reported by: hristo Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38	9 years ago
Rodrigo Ramírez Norambuena	f299dc0d76	app_queue: Add Lastpause field of queue member ... Add time when started a the last pause for a queue member for QueueMemberStatus ami event. Also show accumulate time in seconds when started a pause for a queue member to CLI command 'queue show'. ASTERISK-16394 #close Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c	9 years ago
Joshua Colp	ef293354dc	Merge "app_queue: fix some tab format"	9 years ago
Joshua Colp	fe059b3534	Merge "app_queue: Fix preserved reason of pause when Asterisk is restared"	9 years ago
Joshua Colp	527057ae24	Merge "app_queue.c: remove include for core_unreal.h not used in code."	9 years ago
George Joseph	40da6434c1	build_system: Fix some warnings highlighted by clang ... Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd	9 years ago
Rodrigo Ramírez Norambuena	8c664da0ff	app_queue: fix some tab format ... Change-Id: I2734392b131f1fb0949515d538f83f30fbc15d8c	9 years ago
Rodrigo Ramírez Norambuena	d3969d09ae	app_queue.c: remove include for core_unreal.h not used in code. ... Change-Id: Idc2ae8a6bd869a66544916906744a5678622262d	9 years ago
Rodrigo Ramírez Norambuena	378fed4900	app_queue: Fix preserved reason of pause when Asterisk is restared ... When the Asterisk is restared is not preseved reason paused of members. This patch fixed this cases, retain data on astdb and set when Asterisk is started. ASTERISK-25732 #close Report by: Rodrigo Ramírez Norambuena Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5	9 years ago
Martin Tomec	90b06d1a3c	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	10 years ago
Joshua Colp	0c62e0a55c	Merge "voicemail: Move app_voicemail / res_mwi_external conflict to runtime"	10 years ago
George Joseph	6d18fe151c	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	10 years ago
Richard Mudgett	f88b952093	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	10 years ago
Dade Brandon	ca394161cf	app_amd: Correct maximum_number_of_words functionality & documentation ... - The maximum_number_of_words was previously documented as being the number of words that when exceeded, would result in the AMD application returning that the audio represents a machine. This was inconsistent with its actual functionality - it was a number of words that when REACHED, would result in determination as a machine. This update corrects the functionality to match the previously documented functionality. This is a backwards incompatible change in configuration file, and has been added to UPGRADE.txt as a result. The sample configuration file and application defaults have been updated so that the default value is now 2, which reflects the same default functionality as previous versions. - 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	10 years ago
Carlos Oliva	3e7522533c	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	10 years ago
Corey Farrell	fcaebb0e43	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
Rodrigo Ramírez Norambuena	f2a84b500d	app_queue: Show reason of pause on CLI ... Add value of pause reason when is paused on CLI command "queue show" ASTERISK-25581 #close Report by: Rodrigo Ramírez Norambuena Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864	10 years ago
Joshua Colp	b52b494004	Merge "StatsD: Add sample rate compatibility"	10 years ago
Matt Jordan	65102d19bc	Merge "app_bridgeaddchan: ability to barge into existing call"	10 years ago
Alec Davis	8c14b91651	app_bridgeaddchan: ability to barge into existing call ... To be able to barge into a call by dialling a prefix+extension that maps to the extensions device. Senario is that DECT headset users may be away from their desks and need to transfer the call, the goal is that from any phone they dial a prefix then their extension and are added to the bridge that they are in, from there they can drop the headset call, as it's also on the handset, and transfer the caller. The dialplan would look like, where prefix=73, extension = 8512; exten => _738512,1,BridgeAdd(SIP/cisco0001) ASTERISK-25551 #close Reported By: Alec Davis Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540	10 years ago
tcambron	05addf3d8f	StatsD: Add sample rate compatibility ... Implemented support for the StatsD sample rate parameter, which is a parameter for determining when to send computed statistics to a client. Valid sample rate values are: Less than or equal to 0.0 will never be sent. Between 0.0 and 1.0 will randomly be sent. Greater than or equal to 1.0 will always be sent. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52	10 years ago
Alec Davis	4013f9d577	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
Mark Michelson	ed13732188	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
Joshua Colp	0be147f713	Merge "ast_format_cap_get_names: To display all formats, the buffer was increased."	10 years ago
Alexander Traud	cf79b62778	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Walter Doekes	39daf9f066	docs: Fix a few typo's in app docs (more then, resourse). ... Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7	10 years ago
Joshua Colp	29810aa620	Merge "StatsD: Send stuff to the StatsD server and test"	10 years ago
tcambron	c5093b21ad	StatsD: Send stuff to the StatsD server and test ... Added code to allow the StatsD dialplan application to send data to the server specified in statsd.conf. ASTERISK-25419 Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922	10 years ago
Matt Jordan	2ddd9ff2d9	Merge "app_queue: Added reason pause of member"	10 years ago
tcambron	fdfd0fb488	StatsD: Add user input validation to the application ... Added code to accept user input and validate it before allowing it to be sent to the StatsD server. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3	10 years ago
Rodrigo Ramírez Norambuena	92fa8d1e0e	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	10 years ago
Matt Jordan	da54d3ddd1	Merge "StatsD: Write skeleton Asterisk application"	10 years ago
tcambron	2fe9f09705	StatsD: Write skeleton Asterisk application ... Wrote the skeleton framework for the Asterisk StatsD dialplan application. This includes a load function, unload function, a callback for execution, and XML documentation. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363	10 years ago
Matt Jordan	9db74be3c0	Merge "app_queue.c: Force COLP update if outgoing channel name changed."	10 years ago
Matt Jordan	8bb8f99252	Merge "app_queue.c: Factor out a connected line update routine."	10 years ago
Matt Jordan	e0d8b6a65d	Merge "app_dial.c: Make 'A' option pass COLP updates."	10 years ago
Matt Jordan	360d076dfc	Merge "app_dial.c: Force COLP update if outgoing channel name changed."	10 years ago
Joshua Colp	afabf9da7f	Merge "app_dial.c: Factor out a connected line update routine."	10 years ago
Richard Mudgett	7c7a7ddd27	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	145608bd81	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	1d394774b2	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	680b76eb25	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	fdf0bcb04a	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