asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	0e65855b9d	app_voicemail: Add option to prevent message deletion. Adds an option to VoiceMailMain that prevents the user from deleting messages during that application invocation. This can be useful for public or shared mailboxes, where some users should be able to listen to messages but not delete them. ASTERISK-30063 #close Change-Id: Icdfb8423ae8d1fce65a056b603eb84a672e80a26	3 years ago
Naveen Albert	c277fd02e0	res_parking: Add music on hold override option. An m option to Park and ParkAndAnnounce now allows specifying a music on hold class override. ASTERISK-30087 Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e	3 years ago
Shloime Rosenblum	1978732b8b	res_agi: Evaluate dialplan functions and variables in agi exec if enabled Agi commnad exec can now evaluate dialplan functions and variables if variable AGIEXECFULL is set to yes. this can be useful when executing Playback or Read from agi. ASTERISK-30058 #close Change-Id: I669991f540496e7bddd096fec82b52c083036832	3 years ago
Moritz Fain	aaa14d3c7d	ari: expose channel driver's unique id to ARI channel resource This change exposes the channel driver's unique id (i.e. the Call-ID for chan_sip/chan_pjsip based channels) to ARI channel resources as `protocol_id`. ASTERISK-30027 Reported by: Moritz Fain Tested by: Moritz Fain Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87	3 years ago
Naveen Albert	b0de067565	app_confbridge: Add function to retrieve channels. Adds the CONFBRIDGE_CHANNELS function which can be used to retrieve a comma-separated list of channels, filtered by a particular type of participant category. This output can then be used with functions like UNSHIFT, SHIFT, POP, etc. ASTERISK-30036 #close Change-Id: I1950aff932437476dc1abab6f47fb4ac90520b83	3 years ago
Asterisk Development Team	efca7f4e8d	Update CHANGES and UPGRADE.txt for 18.12.0	3 years ago
Michael Cargile	216a55408e	apps/confbridge: Added hear_own_join_sound option to control who hears sound_join Added the hear_own_join_sound option to the confbridge user profile to control who hears the sound_join audio file. When set to 'yes' the user entering the conference and the participants already in the conference will hear the sound_join audio file. When set to 'no' the user entering the conference will not hear the sound_join audio file, but the participants already in the conference will hear the sound_join audio file. ASTERISK-29931 Added by Michael Cargile Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2	3 years ago
Naveen Albert	cb53ad5671	chan_dahdi: Don't append cadences on dahdi restart. Currently, if any custom ring cadences are specified, they are appended to the array of cadences from wherever we left off last time. This works properly the first time, but on subsequent dahdi restarts, it means that the existing cadences are left alone and (most likely) the same cadences are then re-added afterwards. In short order, the cadence array gets maxed out and the user begins seeing warnings that the array is full and no more cadences may be added. This buggy behavior persists until Asterisk is completely restarted; however, if and when dahdi restart is run again, then the same problem is reintroduced. This fixes this behavior so that cadence parsing is more idempotent, that is so running dahdi restart multiple times starts adding cadences from the beginning, rather than from wherever the last cadence was added. As before, it is still not possible to revert to the default cadences by simply removing all cadences in this manner, nor is it possible to delete existing cadences. However, this does make it possible to update existing cadences, which was not possible before, and also ensures that the cadences remain unchanged if the config remains unchanged. ASTERISK-29990 #close Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d	3 years ago
Naveen Albert	ce00f8758d	func_db: Add function to return cardinality at prefix Adds the DB_KEYCOUNT function, which can be used to retrieve the number of keys at a given prefix in AstDB. ASTERISK-29968 #close Change-Id: Ib2393b77b7e962dbaae6192f8576bc3f6ba92d09	3 years ago
Naveen Albert	09e989f972	asterisk.c: Warn of incompatibilities with remote console. Some command line options to Asterisk only apply when Asterisk is started and cannot be used with remote console mode. If a user tries to use any of these, they are currently simply silently ignored. This prints out a warning if incompatible options are used, informing users that an option used cannot be used with remote console mode. Additionally, some clarifications are added to the help text and man page. ASTERISK-22246 ASTERISK-26582 Change-Id: I980a5380ef2c19e8ea348596396d5382893c4337	3 years ago
Naveen Albert	79689d9df8	func_evalexten: Extension evaluation function. This adds the EVAL_EXTEN function, which may be used to retrieve the variable-substituted data at any extension. ASTERISK-29486 Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432	3 years ago
Mark Petersen	bb2102a991	chan_sip.c Session timers get removed on UPDATE If Asterisk receives a SIP REFER with Session-Timers UAC maintain Session-Timers when sending UPDATE" ASTERISK-29843 Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c	3 years ago
Naveen Albert	8bc6d42a27	chan_pjsip: Add ability to send flash events. PJSIP currently is capable of receiving flash events and converting them to FLASH control frames, but it currently lacks support for doing the reverse: taking a FLASH control frame and converting it into a flash event in the SIP domain. This adds the ability for PJSIP to process flash control frames by converting them into the appropriate SIP INFO message, which can then be sent to the peer. This allows, for example, flash events to be sent between Asterisk systems using PJSIP. ASTERISK-29941 #close Change-Id: I1590221a4d238597f79672fa5825dd4a920c94dd	3 years ago
Naveen Albert	3686a97d79	cli: Add command to evaluate dialplan functions. Adds the dialplan eval function commands to evaluate a dialplan function from the CLI. The return value and function result are printed out and can be used for testing or debugging. ASTERISK-29820 #close Change-Id: I833e97ea54c49336aca145330a2adeebfad05209	3 years ago
Mark Petersen	16e59db514	chan_pjsip: add allow_sending_180_after_183 option added new global config option "allow_sending_180_after_183" that if enabled will preserve 180 after a 183 ASTERISK-29842 Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18	3 years ago
Joshua C. Colp	09e8667fa5	res_pjsip: Always set async_operations to 1. The async_operations setting on a transport configures how many simultaneous incoming packets the transport can handle when multiple threads are polling and waiting on the transport. As we only use a single thread this was needlessly creating incoming packets when set to a non-default value, wasting memory. ASTERISK-30006 Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f	3 years ago
Asterisk Development Team	801317ae05	Update CHANGES and UPGRADE.txt for 18.11.2	3 years ago
Asterisk Development Team	f325cb3d13	Update CHANGES and UPGRADE.txt for 18.11.0	3 years ago
Joshua C. Colp	39cd09c246	func_odbc: Add SQL_ESC_BACKSLASHES dialplan function. Some databases depending on their configuration using backslashes for escaping. When combined with the use of ' this can result in a broken func_odbc query. This change adds a SQL_ESC_BACKSLASHES dialplan function which can be used to escape the backslashes. This is done as a dialplan function instead of being always done as some databases do not require this, and always doing it would result in incorrect data being put into the database. ASTERISK-29838 Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d	3 years ago
Naveen Albert	b7edc08e33	app_queue: Add music on hold option to Queue. Adds the m option to the Queue application, which allows a music on hold class to be specified at runtime which will override the class configured in queues.conf. This option functions like the m option to Dial. ASTERISK-29876 #close Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7	3 years ago
Kfir Itzhak	3959d20ba5	app_queue: Add QueueWithdrawCaller AMI action This adds a new AMI action called QueueWithdrawCaller. This AMI action makes it possible to withdraw a caller from a queue, in a safe and a generic manner. This can be useful for retrieving a specific call and dispatching it to a specific extension. It works by signaling the caller to exit the queue application whenever it can. Therefore, it is not guaranteed that the call will leave the queue. ASTERISK-29909 #close Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec	3 years ago
Alexei Gradinari	8666455bd8	res_pjsip_pubsub: update RLS to reflect the changes to the lists This patch makes the Resource List Subscriptions (RLS) dynamic. The asterisk updates the current subscriptions to reflect the changes to the list on the subscriptions refresh. If list items are added, removed, updated or do not exist anymore, the asterisk regenerates the resource list. ASTERISK-29906 #close Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3	3 years ago
Naveen Albert	59b25e9d59	ami: Allow events to be globally disabled. The disabledevents setting has been added to the general section in manager.conf, which allows users to specify events that should be globally disabled and not sent to any AMI listeners. This allows for processing of these AMI events to end sooner and, for frequent AMI events such as Newexten which users may not have any need for, allows them to not be processed. Additionally, it also cleans up core debug as previously when debug was 3 or higher, the debug was constantly spammed by "Analyzing AMI event" messages along with a complete dump of the event contents (often for Newexten). ASTERISK-29853 #close Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205	3 years ago
Naveen Albert	42525b0fe2	func_channel: Add lastcontext and lastexten. Adds the lastcontext and lastexten channel fields to allow users to access previous dialplan execution locations. ASTERISK-29840 #close Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307	3 years ago
Naveen Albert	3499aea882	documentation: Add since tag to xmldocs DTD Adds the since tag to the documentation DTD so that individual applications, functions, etc. can now specify when they were added to Asterisk. This tag is added at the individual application, function, etc. level as opposed to at the module level because modules can expand over time as new functionality is added, and granularity only to the module level would generally not be useful. This enables the ability to more easily determine when new functionality was added to Asterisk, down to minor version as opposed to just by major version. This makes it easier for users to write more portable dialplan if desired to not use functionality that may not be widely available yet. ASTERISK-29896 #close Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5	3 years ago
Alexei Gradinari	e2423c6f49	res_pjsip_pubsub: provide a display name for RLS subscriptions Whereas BLFs allow to show a display name for each RLS entry, the asterisk provides only the extension now. This is not end user friendly. This commit adds a new resource_list option, resource_display_name, to indicate whether display name of resource or the resource name being provided for RLS entries. If this option is enabled, the Display Name will be provided. This option is disabled by default to remain the previous behavior. If the 'event' set to 'presence' or 'dialog' the non-empty HINT name will be set as the Display Name. The 'message-summary' is not supported yet. ASTERISK-29891 #close Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681	3 years ago
Naveen Albert	332eed3aa7	app_mf: Add max digits option to ReceiveMF. Adds an option to the ReceiveMF application to allow specifying a maximum number of digits. Originally, this capability was not added to ReceiveMF as it was with ReceiveSF because typically a ST digit is used to denote that sending of digits is complete. However, there are certain signaling protocols which simply transmit a digit (such as Expanded In-Band Signaling) and for these, it's necessary to be able to read a certain number of digits, as opposed to until receiving a ST digit. This capability is added as an option, as opposed to as a parameter, to remain compatible with existing usage (and not shift the parameters). ASTERISK-29877 #close Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b	3 years ago
Alexei Gradinari	fcdeb3e5b7	app_queue: load queues and members from Realtime when needed There are a lot of Queue AMI actions and Queue applications which do not load queue and queue members from Realtime. AMI actions QueuePause - if queue not in memory - response "Interface not found". QueueStatus/QueueSummary - if queue not in memory - empty response. Applications: PauseQueueMember - if queue not in memory Attempt to pause interface %s, not found UnpauseQueueMember - if queue not in memory Attempt to unpause interface xxxxx, not found This patch adds a new function load_realtime_queues which loads queue and queue members for desired queue or all queues and all members if param 'queuename' is NULL or empty. Calls the function load_realtime_queues when needed. Also this patch fixes leak of ast_config in function set_member_value. Also this patch fixes incorrect LOG_WARNING when pausing/unpausing already paused/unpaused member. The function ast_update_realtime returns 0 when no record modified. So 0 is not an error to warn about. ASTERISK-29873 #close ASTERISK-18416 #close ASTERISK-27597 #close Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c	3 years ago
Sean Bright	118e034287	manager.c: Generate valid XML if attribute names have leading digits. The XML Manager Event Interface (amxml) now generates attribute names that are compliant with the XML 1.1 specification. Previously, an attribute name that started with a digit would be rendered as-is, even though attribute names must not begin with a digit. We now prefix attribute names that start with a digit with an underscore ('_') to prevent XML validation failures. This is not backwards compatible but my assumption is that compliant XML parsers would already have been complaining about this. ASTERISK-29886 #close Change-Id: Icfaa56a131a082d803e9b7db5093806d455a0523	3 years ago
Asterisk Development Team	aa6a50630f	Update CHANGES and UPGRADE.txt for 18.10.0	3 years ago
Mark Petersen	95ee1d06d6	chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN resolve issue with pickup on device that uses "183" and not "180" ASTERISK-29832 Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841	3 years ago
Naveen Albert	6fc8453e96	cdr: allow disabling CDR by default on new channels Adds a new option, defaultenabled, to the CDR core to control whether or not CDR is enabled on a newly created channel. This allows CDR to be disabled by default on new channels and require the user to explicitly enable CDR if desired. Existing behavior remains unchanged. ASTERISK-29808 #close Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1	3 years ago
Mark Petersen	c8d89e7e1b	app_queue.c: Queue don't play "thank-you" when here is no hold time announcements if holdtime is (0 min, 0 sec) there is no hold time announcements we should then also not playing queue-thankyou ASTERISK-29831 Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c	3 years ago
George Joseph	3f093b8dda	bundled_pjproject: Make it easier to hack There are times when you need to troubleshoot issues with bundled pjproject or add new features that need to be pushed upstream but... * The source directory created by extracting the pjproject tarball is not scanned for code changes so you have to keep forcing rebuilds. * The source directory isn't a git repo so you can't easily create patches, do git bisects, etc. * Accidentally doing a make distclean will ruin your day by wiping out the source directory, and your changes. * etc. This commit makes that easier. See third-party/pjproject/README-hacking.md for the details. ASTERISK-29824 Change-Id: Idb1251040affdab31d27cd272dda68676da9b268	3 years ago
Mark Petersen	dea71ddbbf	app_queue.c: Support for Nordic syntax in announcements adding support for playing the correct en/et for nordic languages by adding 'n' for neuter gender in the relevant ast_say_number ASTERISK-29827 Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1	3 years ago
Naveen Albert	f133ae6ca2	ami: Add AMI event for Wink Adds an AMI event for a wink frame. ASTERISK-29830 #close Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56	3 years ago
Naveen Albert	1c2f311ba3	cli: Add module refresh command Adds a command to the CLI to unload and then load a module. This makes it easier to perform these operations which are often done subsequently to load a new version of a module. "module reload" already refers to reloading of configuration, so the name "refresh" is chosen instead. ASTERISK-29807 #close Change-Id: I595f6f11774a0de2565a1fba38da22309ce93a2c	3 years ago
Mark Petersen	3d71bcd2f4	app_queue.c: added DIALEDPEERNUMBER on outgoing channel added that we set DIALEDPEERNUMBER on the outgoing channels so it is avalible in b(content^extension^line) this add the same behaviour as Dial ASTERISK-29795 Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6	4 years ago
Mark Petersen	a4c42e70c1	app_voicemail.c: Support for Danish syntax in VM added support for playing the correct plural sound file dependen on where you have 1 or multipe messages based on the existing SE/NO code ASTERISK-29797 Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2	4 years ago
Naveen Albert	2f1eb56116	app_sendtext: Add ReceiveText application Adds a ReceiveText application that can be used in conjunction with SendText. Currently, there is no way in Asterisk to receive text in the dialplan (or anywhere else, really). This allows for Asterisk to be the recipient of text instead of just the sender. ASTERISK-29759 #close Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d	4 years ago
Naveen Albert	c0cdaf0246	func_json: Adds JSON_DECODE function Adds the JSON_DECODE function for parsing JSON in the dialplan. JSON parsing already exists in the Asterisk core and is used for many different things. This function exposes the basic parsing capability to the user in the dialplan, for instance, in conjunction with CURL for using API responses. ASTERISK-29706 #close Change-Id: Iea60c49a7358dfdc2db60803cdc9a742f808ba2c	4 years ago
Naveen Albert	2b2b708d43	app_mf: Add full tech-agnostic MF support Adds tech-agnostic support for MF signaling by adding MF sender and receiver applications as well as Dial integration. ASTERISK-29496-mf #do-not-close Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4	4 years ago
Alexander Traud	55dd77b921	xmldoc: Avoid whitespace around value for parameter/required. Otherwise, the value 'false' was not found in the enumerated set of the XML DTD for the XML attribute 'required' in the XML element 'parameter'. Therefore, DTD validation of the runtime XML failed. ASTERISK-29790 Change-Id: Id13f230ad65a70dd8c2e3ae9ac85d1e841aed03e	4 years ago
Alexander Traud	5b24edeb7c	xmldoc: Correct definition for XML element 'matchInfo'. ASTERISK-29791 Change-Id: I7c656498427fcadd0a5d61a54ff67e6036609725	4 years ago
Alexander Traud	d914e14420	progdocs: Update Makefile. In developer mode, use internal documentation as well. This should produce no warnings. Fix yours! In noisy mode, output all possible warnings of Doxygen. This creates zillion of warnings. Double-check your current module! Any warnings are in the file './doxygen.log'. Beside that, this change avoids deprecated parameters because the configuration file for Doxygen contains only those parameters which differ from the default. This avoids the need to update the file on each run. Furthermore, it adds AST_VECTOR to be expanded. Finally, the default name for that file is Doxyfile. Therefore, let us use that! ASTERISK-26991 ASTERISK-20259 Change-Id: I4129092a199d5e24c319a09cd088614b121015af	4 years ago
Dustin Marquess	b2e71b82e7	res_fax_spandsp: Add spandsp 3.0.0+ compatibility Newer versions of spandsp did refactoring of code to add new features like color FAXing. This refactoring broke backwards compatibility. Add support for the new version while retaining support for 0.0.6. ASTERISK-29729 #close Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1	4 years ago
Asterisk Development Team	868d2d5e53	Update CHANGES and UPGRADE.txt for 18.9.0	4 years ago
Naveen Albert	ca2e13e18f	res_tonedetect: Add call progress tone detection Makes basic call progress tone detection available in a tech-agnostic manner with the addition of the ToneScan application. This can determine if the channel has encountered a busy signal, SIT tones, dial tone, modem, fax machine, etc. A few basic async progress tone detect options are also added to the TONE_DETECT function. ASTERISK-29720 #close Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90	4 years ago
Josh Soref	135d51e55e	doc: Spelling fixes Correct typos of the following word families: transparent roughly ASTERISK-29714 Change-Id: I2b90c68dfde4aa3f0d58f64f8187465336acb1b3	4 years ago
Naveen Albert	bea08a563b	chan_iax2: Allow both secret and outkey at dial time Historically, the dial syntax for IAX2 has held that an outkey (used only for RSA authenticated calls) and a secret (used only for plain text and MD5 authenticated calls, historically) were mutually exclusive, and thus the same position in the dial string was used for both values. Now that encryption is possible with RSA authentication, this poses a limitation, since encryption requires a secret and RSA authentication requires an outkey. Thus, the dial syntax is extended so that both a secret and an outkey can be specified. The new extended syntax is backwards compatible with the old syntax. However, a secret can now be specified after the outkey, or the outkey can be specified after the secret. This makes it possible to spawn an encrypted RSA authenticated call without a corresponding peer being predefined in iax.conf. ASTERISK-29707 #close Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e	4 years ago

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881 Commits (ae08471d8b0bff35fef9b6aca565bd3e85f6568d)