asterisk

Commit Graph

Author	SHA1	Message	Date
Josh Soref	5ad9ec2447	apps: Spelling fixes Correct typos of the following word families: simultaneously administrator directforward attachfmt dailplan automatically applicable nouns explicit outside sponsored attachment audio spied doesn't counting encoded implements recursively emailaddress arguments queuerules members priority output advanced silencethreshold brazilian debugging argument meadmin formatting integrated sneakiness ASTERISK-29714 Change-Id: Ie5ecaec91c00b26309da4e51cfc0991a5bb7d092	4 years ago
Naveen Albert	656880a4b1	app_voicemail: Fix phantom voicemail bug on rerecord If users are able to press # for options while leaving a message and then press 3 to rerecord the message, if the caller hangs up during the rerecord prompt but before Asterisk starts recording a message, then an "empty" voicemail gets processed whereby an email gets sent out notifying the user of a 0:00 duration message. The file doesn't actually exist, so playback will fail since there was no message to begin with. This adds a check after the streaming of the rerecord announcement to see if the caller has hung up. If so, we bail out early so that we can clean up properly. ASTERISK-29391 #close Change-Id: Id965d72759a2fd3b39afb76fec08aaebebe75c31	4 years ago
Rodrigo Ramírez Norambuena	d81d5ad832	app_queue: Add LoginTime field for member in a queue. Add a time_t logintime to storage a time when a member is added into a queue. Also, includes show this time (in seconds) using a 'queue show' command and the field LoginTime for response for AMI events. ASTERISK-18069 #close Change-Id: Ied6c3a300f78d78eebedeb3e16a1520fc3fff190	4 years ago
Shloime Rosenblum	154c592799	apps/app_playback.c: Add 'mix' option to app_playback I am adding a mix option that will play by filename and say.conf unlike say option that will only play with say.conf. It will look on the format of the name, if it is like say it play with say.conf if not it will play the file name. ASTERISK-29662 Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a	4 years ago
Naveen Albert	60bbfe4572	app_read: Fix null pointer crash If the terminator character is not explicitly specified and an indications tone is used for reading a digit, there is no null pointer check so Asterisk crashes. This prevents null usage from occuring. ASTERISK-29673 #close Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac	4 years ago
Naveen Albert	eff78c8549	app_queue: Fix hint updates for included contexts Previously, if custom hints were used with the hint: format in app_queue, when device state changes occured, app_queue would only do a literal string comparison of the context used for the hint in app_queue and the context of the hint which just changed state. This caused hints to not update and become stale if the context associated with the agent included the context which actually changes state, essentially completely breaking device state for any such agents defined in this manner. This fix adds an additional check to ensure that included contexts are also compared against the context which changed state, so that the behavior is correct no matter whether the context is specified to app_queue directly or indirectly. ASTERISK-29578 #close Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78	4 years ago
Naveen Albert	eb874f92db	logger: Add custom logging capabilities Adds the ability for users to log to custom log levels by providing custom log level names in logger.conf. Also adds a logger show levels CLI command. ASTERISK-29529 Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702	4 years ago
Sean Bright	245778a756	app_externalivr.c: Fix mixed leading whitespace in source code. No functional changes. Change-Id: I46514152c0af67f395526374aaa847ccd6a85378	4 years ago
Carlos Oliva	ad1f7fae70	app_mp3: Force output to 16 bits in mpg123 In new mpg123 versions (since 1.26) the default output is 32 bits Asterisk expects the output in 16 bits, so we force the output to be on 16 bits. It will work wit new and old versions of mpg123. Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key! ASTERISK-29635 #close Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816	4 years ago
Naveen Albert	203e73f5af	app_mf: Add channel agnostic MF sender Adds a SendMF application and PlayMF manager event to send arbitrary R1 MF tones on the current or specified channel. ASTERISK-29496 Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4	4 years ago
Naveen Albert	5fe3a745e4	app_stack: Include current location if branch fails Previously, the error emitted when app_stack tries to branch to a dialplan location that doesn't exist has included only the information about the attempted branch in the error log. This adds the current location as well so users can see where the branch failed in the logs. ASTERISK-29626 Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce	4 years ago
Sean Bright	f67b72093e	app_voicemail.c: Ability to silence instructions if greeting is present. There is an option to silence voicemail instructions but it does not take into consideration if a recorded greeting exists or not. Add a new 'S' option that does that. ASTERISK-29632 #close Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4	4 years ago
Sean Bright	16b0f460f6	config_options: Handle ACO arrays correctly in generated XML docs. There are 3 separate changes here but they are all closely related: * Only try to set matchfield attributes on 'field' nodes * We need to adjust how we treat the category pointer based on the value of the category_match, to avoid memory corruption. We now generate a regex-like string when match types other than ACO_WHITELIST and ACO_BLACKLIST are used. * Switch app_agent_pool from ACO_BLACKLIST_ARRAY to ACO_BLACKLIST_EXACT since we only have one category we need to ignore, not two. ASTERISK-29614 #close Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e	4 years ago
Naveen Albert	0e4a1c5079	app_read: Allow reading # as a digit Allows for the digit # to be read as a digit, just like any other DTMF digit, as opposed to forcing it to be used as an end of input indicator. The default behavior remains unchanged. ASTERISK-18454 #close Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b	4 years ago
Naveen Albert	9e947b0463	app_queue: Don't reset queue stats on reload Prevents reloads of app_queue from also resetting queue statistics. Also preserves individual queue agent statistics if we're just reloading members. ASTERISK-28701 Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1	4 years ago
Naveen Albert	b5044586f7	app_morsecode: Add American Morse code Previously, the Morsecode application only supported international Morse code. This adds support for American Morse code and adds an option to configure the frequency used in off intervals. Additionally, the application checks for hangup between tones to prevent application execution from continuing after hangup. ASTERISK-29541 Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4	4 years ago
Naveen Albert	3f9ef427b5	app_milliwatt: Timing fix The Milliwatt application uses incorrect tone timings that cause it to play the 1004 Hz tone constantly. This adds an option to enable the correct timing behavior, so that the Milliwatt application can be used for milliwatt test lines. The default behavior remains unchanged for compatability reasons, even though it is incorrect. ASTERISK-29575 #close Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c	4 years ago
Naveen Albert	2394757e55	app_originate: Add ability to set codecs A list of codecs to use for dialplan-originated calls can now be specified in Originate, similar to the ability in call files and the manager action. Additionally, we now default to just using the slin codec for originated calls, rather than all the slin* codecs up through slin192, which has been known to cause issues and inconsistencies from AMI and call file behavior. ASTERISK-29543 Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883	4 years ago
Joshua C. Colp	f18107f191	app_dahdiras: Remove deprecated module. ASTERISK-29591 Change-Id: I021d37b729631d40f84e35bb21e2893777be1858	4 years ago
Joshua C. Colp	b1e5b1874c	app_nbscat: Remove deprecated module. ASTERISK-29590 Change-Id: I87cf0f536b77d222c8eda003376ac47fae86ed43	4 years ago
Joshua C. Colp	7ee6fb0372	app_image: Remove deprecated module. ASTERISK-29589 Change-Id: I8057eb2ca1ca4c3b27ed2fe04bea10e9cb551cdd	4 years ago
Joshua C. Colp	0b3a149001	app_url: Remove deprecated module. ASTERISK-29588 Change-Id: If846d40b37c5b646bcd7326111db280529a5971b	4 years ago
Joshua C. Colp	41afcb9422	app_fax: Remove deprecated module. ASTERISK-29587 Change-Id: I038237bbb56b1161d7d5e20cda11ed32e13d3ca2	4 years ago
Joshua C. Colp	83cad340fc	app_ices: Remove deprecated module. ASTERISK-29586 Change-Id: I1e0a4535135b00938b609fe0ccba9bbddbac93ad	4 years ago
Joshua C. Colp	141dc519b0	policy: Deprecate modules and add versions to others. app_meetme is deprecated in 19, to be removed in 21. app_osplookup is deprecated in 19, to be removed in 21. chan_alsa is deprecated in 19, to be removed in 21. chan_mgcp is deprecated in 19, to be removed in 21. chan_skinny is deprecated in 19, to be removed in 21. res_pktccops is deprecated in 19, to be removed in 21. app_macro was deprecated in 16, to be removed in 21. chan_sip was deprecated in 17, to be removed in 21. res_monitor was deprecated in 16, to be removed in 21. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29558 ASTERISK-29567 ASTERISK-29572 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Naveen Albert	0975cff6c0	app_queue: Allow streaming multiple announcement files Allows multiple files comprising an agent announcement to be played by separating on the ampersand, similar to the multi-file support in other Asterisk applications. ASTERISK-29528 Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a	4 years ago
Naveen Albert	6645cf8d45	app_dtmfstore: New application to store digits Adds application to asynchronously collect digits dialed on a channel in the TX or RX direction using a framehook and stores them in a specified variable, up to a configurable number of digits. ASTERISK-29477 Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f	4 years ago
Joshua C. Colp	90c9c90b11	docs: Remove embedded macro in WaitForCond XML documentation. Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c	4 years ago
Naveen Albert	244491f9b2	app_reload: New Reload application Adds an application to reload modules from within the dialplan. ASTERISK-29454 Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774	4 years ago
Naveen Albert	c01b4e0d4b	app_waitforcond: New application While several applications exist to wait for a certain event to occur, none allow waiting for any generic expression to become true. This application allows for waiting for a condition to become true, with configurable timeout and checking interval. ASTERISK-29444 Change-Id: I08adf2824b8bc63405778cf355963b5005612f41	4 years ago
Naveen Albert	1e5a2cfe30	app_dial: Expanded A option to add caller announcement Hitherto, the A option has made it possible to play audio upon answer to the called party only. This option is expanded to allow for playback of an audio file to the caller instead of or in addition to the audio played to the answerer. ASTERISK-29442 Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e	4 years ago
Naveen Albert	b742514553	app_originate: Allow setting Caller ID and variables Caller ID can now be set on the called channel and Variables can now be set on the destination using the Originate application, just as they can be currently using call files or the Manager Action. ASTERISK-29450 Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66	4 years ago
Naveen Albert	35437879e5	app_confbridge: New ConfKick() application Adds a new ConfKick() application, which may be used to kick a specific channel, all channels, or all non-admin channels from a specified conference bridge, similar to existing CLI and AMI commands. ASTERISK-29446 Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b	4 years ago
Naveen Albert	5f8cabc232	app_confbridge: New option to prevent answer supervision A new user option, answer_channel, adds the capability to prevent answering the channel if it hasn't already been answered yet. ASTERISK-29440 Change-Id: I26642729d0345f178c7b8045506605c8402de54b	4 years ago
Naveen Albert	567ea5abf8	app_voicemail: Configurable voicemail beep Hitherto, VoiceMail() played a non-customizable beep tone to indicate the caller could leave a message. In some cases, the beep may not be desired, or a different tone may be desired. To increase flexibility, a new option allows customization of the tone. If the t option is specified, the default beep will be overridden. Supplying an argument will cause it to use the specified file for the tone, and omitting it will cause it to skip the beep altogether. If the option is not used, the default behavior persists. ASTERISK-29349 Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280	4 years ago
Sean Bright	aac442eecd	app_queue.c: Remove dead 'updatecdr' code. Also removed the sample documentation, and some oddly-placed documentation about the timeout argument to the Queue() application itself. There is a large section on the timeout behavior below. ASTERISK-26614 #close Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217	4 years ago
Sean Bright	8d3d7bdb82	app_queue.c: Don't crash when realtime queue name is empty. ASTERISK-27542 #close Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a	4 years ago
Joshua C. Colp	a8a08bcd1e	app_queue: Only send QueueMemberStatus if status changes. If a queue member was updated with the same status multiple times each time a QueueMemberStatus event would be sent which would be a duplicate of the previous. This change makes it so that the QueueMemberStatus event is only sent if the status actually changes. ASTERISK-29355 Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116	4 years ago
Joshua C. Colp	149e5e5b86	xml: Embed module information into core XML documentation. This change embeds the MODULEINFO block of modules into the core XML documentation. This provides a shared mechanism for use by both menuselect and Asterisk for information and a definitive source of truth. ASTERISK-29335 Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90	4 years ago
Joshua C. Colp	7438586d8e	documentation: Fix non-matching module support levels. Some modules have a different support level documented in their MODULEINFO XML and Asterisk module definition. This change brings the two in sync for the modules which were not matching. ASTERISK-29336 Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35	4 years ago
Sean Bright	8987de270f	app_dial.c: Only send DTMF on first progress event. ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	4 years ago
Sean Bright	932eae69ab	app_page.c: Don't fail to Page if beep sound file is missing ASTERISK-16799 #close Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4	4 years ago
Ivan Poddubnyi	4d8fc97e4a	app_queue: Fix conversion of complex extension states into device states Queue members using dialplan hints as a state interface must handle INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE. ASTERISK-28369 Change-Id: I127e06943d4b4f1afc518f9e396de77449992b9f	4 years ago
Sebastien Duthil	6e695c867f	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	4 years ago
Sean Bright	4a71b08091	app_read: Release tone zone reference on early return. Change-Id: I350939f2220f9e5d44ddf4c8d9a4c99fde4d169a	4 years ago
Dan Cropp	55891227e8	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	4 years ago
Kevin Harwell	3bcf483373	app_mixmonitor: cleanup datastore when monitor thread fails to launch launch_monitor_thread is responsible for creating and initializing the mixmonitor, and dependent data structures. There was one off nominal path after the datastore gets created that triggers when the channel being monitored is hung up prior to monitor starting itself. If this happened the monitor thread would not "launch", and the mixmonitor object and associated objects are freed, including the underlying datastore data object. However, the datastore itself was not removed from the channel, so when the channel eventually gets destroyed it tries to access the previously freed datastore data and crashes. This patch removes and frees datastore object itself from the channel before freeing the mixmonitor object thus ensuring the channel does not call it when destroyed. ASTERISK-28947 #close Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8	4 years ago
Sean Bright	44d68bd56b	app_voicemail: Prevent deadlocks when out of ODBC database connections ASTERISK-28992 #close Change-Id: Ia7d608924036139ee2520b840d077762d02668d0	4 years ago
Sean Bright	357510cec3	app_chanspy: Spyee information missing in ChanSpyStop AMI Event The documentation in the wiki says there should be spyee-channel information elements in the ChanSpyStop AMI event. https://wiki.asterisk.org/wiki/x/Xc5uAg However, this is not the case in Asterisk <= 16.10.0 Version. We're using these Spyee* arguments since Asterisk 11.x, so these arguments vanished in Asterisk 12 or higher. For maximum compatibility, we still send the ChanSpyStop event even if we are not able to find any 'Spyee' information. ASTERISK-28883 #close Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f	4 years ago
Joshua C. Colp	eda3679c1c	voicemail: add option 'e' to play greetings as early media When using this option, answering the channel is deferred until all prompts/greetings have been played and the caller is about to leave their message. ASTERISK-29118 #close Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb	5 years ago

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5153 Commits (acf74178b15d92ec27efd88c908ad0366d71a40d)