asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	d014cedf5d	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	4 years ago
Mark Petersen	ac9607d03b	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	4 years ago
Naveen Albert	06eff7d5d0	ami: Add AMI event for Wink Adds an AMI event for a wink frame. ASTERISK-29830 #close Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56	4 years ago
Naveen Albert	ef00ff4e1e	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	4 years ago
Mark Petersen	0dbf313a76	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	ff0b048543	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	85dc5214e3	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	0eaa214be9	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	9eba9f44d6	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	a7688c0216	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	090c56d80c	xmldoc: Correct definition for XML element 'matchInfo'. ASTERISK-29791 Change-Id: I7c656498427fcadd0a5d61a54ff67e6036609725	4 years ago
Alexander Traud	b49ae92fe7	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	3162f6dfdd	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	191b92a69e	Update CHANGES and UPGRADE.txt for 16.23.0	4 years ago
Naveen Albert	876e647232	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	5d1e3cb17f	doc: Spelling fixes Correct typos of the following word families: transparent roughly ASTERISK-29714 Change-Id: I2b90c68dfde4aa3f0d58f64f8187465336acb1b3	4 years ago
Naveen Albert	530e89655e	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
George Joseph	ecb46511cc	ast_coredumper: Refactor to better find things The search for a running asterisk when --running is used has been greatly simplified and in the event it doesn't work, you can now specify a pid to use on the command line with --pid. The search for asterisk modules when --tarball-coredumps is used has been enhanced to have a better chance of finding them and in the event it doesn't work, you can now specify --libdir on the command line to indicate the library directory where they were installed. The DATEFORMAT variable was renamed to DATEOPTS and is now passed to the 'date' utility rather than running DATEFORMAT as a command. The coredump and output files are now renamed with DATEOPTS. This can be disabled by specifying --no-rename. Several confusing and conflicting options were removed: --append-coredumps --conffile --no-default-search --tarball-uniqueid The script was re-structured to make it easier for follow. Change-Id: I674be64bdde3ef310b6a551d4911c3b600ffee59	4 years ago
Ben Ford	669e16b3dc	STIR/SHAKEN: Option split and response codes. The stir_shaken configuration option now has 4 different choices to pick from: off, attest, verify, and on. Off and on behave the same way they do now. Attest will only perform attestation on the endpoint, and verify will only perform verification on the endpoint. Certain responses are required to be sent based on certain conditions for STIR/SHAKEN. For example, if we get a Date header that is outside of the time range that is considered valid, a 403 Stale Date response should be sent. This and several other responses have been added. Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7	4 years ago
Rodrigo Ramírez Norambuena	b12e8b5924	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	a68ef6aafd	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
Asterisk Development Team	448739fe63	Update CHANGES and UPGRADE.txt for 16.22.0	4 years ago
Naveen Albert	8ad4cb901b	chan_iax2: Add encryption for RSA authentication Adds support for encryption to RSA-authenticated calls. Also prevents crashes if an RSA IAX2 call is initiated to a switch requiring encryption but no secret is provided. ASTERISK-20219 Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40	4 years ago
Matthew Kern	39824c7a96	res_pjsip_t38: bind UDPTL sessions like RTP In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the fallback use of the transport's bind address solve problems sending media on systems that cannot send ipv4 packets on ipv6 sockets, and certain other situations. This change extends both of these behaviors to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific problems on these systems, introducing a new option endpoint/t38_bind_udptl_to_media_address. ASTERISK-29402 Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557	4 years ago
Joseph Nadiv	722b81904e	res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts The behavior of max_contacts and remove_existing are connected. If remove_existing is enabled, the soonest expiring contacts are removed. This may occur when there is an unavailable contact. Similarly, when remove_existing is not enabled, registrations from good endpoints are rejected in favor of retaining unavailable contacts. This commit adds a new AOR option remove_unavailable, and the effect of this setting will depend on remove_existing. If remove_existing is set to no, we will still remove unavailable contacts when they exceed max_contacts, if there are any. If remove_existing is set to yes, we will prioritize the removal of unavailable contacts before those that are expiring soonest. ASTERISK-29525 Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784	4 years ago
Naveen Albert	b0027c65f9	func_vmcount: Add support for multiple mailboxes Allows multiple mailboxes to be specified for VMCOUNT instead of just one. ASTERISK-29661 #close Change-Id: I9108528300795fd5b607efa9d4dd7b74be031813	4 years ago
Sean Bright	97ce647afd	message.c: Support 'To' header override with AMI's MessageSend. The MessageSend AMI action has been updated to allow the Destination and the To addresses to be provided separately. This brings the MessageSend manager command in line with the capabilities of the MessageSend dialplan application. ASTERISK-29663 #close Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c	4 years ago
Naveen Albert	228c97d3bf	func_channel: Add CHANNEL_EXISTS function. Adds a function to check for the existence of a channel by name or by UNIQUEID. ASTERISK-29656 #close Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb	4 years ago
Naveen Albert	41ba9f5f31	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
Asterisk Development Team	9b17523c4b	Update CHANGES and UPGRADE.txt for 16.21.0	4 years ago
Naveen Albert	425bd97f0d	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	249fe0f37e	func_strings: Add STRBETWEEN function Adds the STRBETWEEN function, which can be used to insert a substring between each character in a string. For instance, this can be used to insert pauses between DTMF tones in a string of digits. ASTERISK-29627 Change-Id: Ice23009d4a8e9bb9718d2b2301d405567087d258	4 years ago
Naveen Albert	b8e2b2ed5e	func_env: Add DIRNAME and BASENAME functions Adds the DIRNAME and BASENAME functions, which are wrappers around the corresponding C library functions. These can be used to safely and conveniently work with file paths and names in the dialplan. ASTERISK-29628 #close Change-Id: Id3aeb907f65c0ff96b6e57751ff0cb49d61db7f3	4 years ago
Naveen Albert	11516e4b8e	func_sayfiles: Retrieve say file names Up until now, all of the logic used to translate arguments to the Say applications has been directly coupled to playback, preventing other modules from using this logic. This refactors code in say.c and adds a SAYFILES function that can be used to retrieve the file names that would be played. These can then be used in other applications or for other purposes. Additionally, a SayMoney application and a SayOrdinal application are added. Both SayOrdinal and SayNumber are also expanded to support integers greater than one billion. ASTERISK-29531 Change-Id: If9718c89353b8e153d84add3cc4637b79585db19	4 years ago
Naveen Albert	698604a064	res_tonedetect: Tone detection module dsp.c contains arbitrary tone detection functionality which is currently only used for fax tone recognition. This change makes this functionality publicly accessible so that other modules can take advantage of this. Additionally, a WaitForTone and TONE_DETECT app and function are included to allow users to do their own tone detection operations in the dialplan. ASTERISK-29546 Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26	4 years ago
Sean Bright	5c28f881aa	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
Naveen Albert	38cc3160d3	chan_iax2: Add ANI2/OLI information element Adds an information element for ANI2 so that Originating Line Information can be transmitted over IAX2 channels. ASTERISK-29605 #close Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2	4 years ago
Naveen Albert	c5c5171ec8	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
Sebastien Duthil	166556961b	res_rtp_asterisk: Automatically refresh stunaddr from DNS This allows the STUN server to change its IP address without having to reload the res_rtp_asterisk module. The refresh of the name resolution occurs first when the module is loaded, then recurringly, slightly after the previous DNS answer TTL expires. ASTERISK-29508 #close Change-Id: I7955a046293f913ba121bbd82153b04439e3465f	4 years ago
Naveen Albert	220ce865db	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	5 years ago
George Joseph	18c467686c	res_pjproject: Allow mapping to Asterisk TRACE level Allow mapping pjproject log messages to the Asterisk TRACE log level. The defaults were also changes to log pjproject levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6 all went to DEBUG. ASTERISK-29582 Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d	5 years ago
Naveen Albert	9ee7b884f1	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	5 years ago
Naveen Albert	7f4433f8c6	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	5 years ago
Naveen Albert	1055660252	func_scramble: Audio scrambler function Adds a function to scramble audio on a channel using whole spectrum frequency inversion. This can be used as a privacy enhancement with applications like ChanSpy or other potentially sensitive audio. ASTERISK-29542 Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e	5 years ago
Naveen Albert	4606fb604a	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	5 years ago
Asterisk Development Team	4bf9b2d279	Update CHANGES and UPGRADE.txt for 16.20.0	5 years ago
Naveen Albert	af17973bda	func_frame_drop: New function Adds function to selectively drop specified frames in the TX or RX direction on a channel, including control frames. ASTERISK-29478 Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec	5 years ago
Naveen Albert	1298c02665	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	5 years ago
Igor Goncharovsky	8da1466478	res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request. It may be used to get all X- headers in case the actual set and names of headers unknown. ASTERISK-29389 Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b	5 years ago
Rijnhard Hessel	b40e97b1d7	res_statsd: handle non-standard meter type safely Meter types are not well supported, lacking support in telegraf, datadog and the official statsd servers. We deprecate meters and provide a compliant fallback for any existing usages. A flag has been introduced to allow meters to fallback to counters. ASTERISK-29513 Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7	5 years ago

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841 Commits (67a87c01ff38d52fc8a9b41fd3e703bf55a51eea)