asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	d1fb397cfc	general: Fix broken links. This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in `f28047db36`. Resolves: #430	1 year ago
George Joseph	c3c82441a2	Prepare master for Asterisk 22	2 years ago
InterLinked1	ad6ff4cbf2	asterisk.c: Fix option warning for remote console. (#103 ) Commit `09e989f972` categorized the T option as not being compatible with remote consoles, but they do affect verbose messages with remote console. This fixes this. Resolves: #102	2 years ago
Sean Bright	0d6b271831	core: Cleanup gerrit and JIRA references. (#58 ) * Remove .gitreview and switch to pulling the main asterisk branch version from configure.ac instead. * Replace references to JIRA with GitHub. * Other minor cleanup found along the way. Resolves: #39	2 years ago
Joshua Colp	4ec4543332	Revert "app_queue: periodic announcement configurable start time." This reverts commit `71e317f68f`. Reason for revert: Causes segmentation fault. Change-Id: I3beeda83249bffec2a8f246aa50a6b2f1b59ef59	2 years ago
Naveen Albert	ecf49ff746	pbx_dundi: Add PJSIP support. Adds PJSIP as a supported technology to DUNDi. To facilitate this, we now allow an endpoint to be specified for outgoing PJSIP calls. We also allow users to force a specific channel technology for outgoing SIP-protocol calls. ASTERISK-28109 #close ASTERISK-28233 #close Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf	2 years ago
George Joseph	1245e3330b	test.c: Fix counting of tests and add 2 new tests The unit test XML output was counting all registered tests as "run" even when only a subset were actually requested to be run and the "failures" attribute was missing. * The "tests" attribute of the "testsuite" element in the output XML now reflects only the tests actually requested to be executed instead of all the tests registered. * The "failures" attribute was added to the "testsuite" element. Also added 2 new unit tests that just pass and fail to be used for CI testing. Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69	2 years ago
Mike Bradeen	edd7f1b060	bridge_builtin_features: add beep via touch variable Add periodic beep option to one-touch recording by setting the touch variable TOUCH_MONITOR_BEEP or TOUCH_MIXMONITOR_BEEP to the desired interval in seconds. If the interval is less than 5 seconds, a minimum of 5 seconds will be imposed. If the interval is set to an invalid value, it will default to 15 seconds. A new test event PERIODIC_HOOK_ENABLED was added to the func_periodic_hook hook_on function to indicate when a hook is started. This is so we can test that the touch variable starts the hook as expected. ASTERISK-30446 Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661	2 years ago
Mike Bradeen	c4a55322bc	res_mixmonitor: MixMonitorMute by MixMonitor ID While it is possible to create multiple mixmonitor instances on a channel, it was not previously possible to mute individual instances. This change includes the ability to specify the MixMonitorID when calling the manager action: MixMonitorMute. This will allow an individual MixMonitor instance to be muted via id. This id can be stored as a channel variable using the 'i' MixMonitor option. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor spy-type audiohooks on the channel. This is done via the new audiohook function: ast_audiohook_set_mute_all. ASTERISK-30464 Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98	2 years ago
Mike Bradeen	9504d17ad2	format_sln: add .slin as supported file extension Adds '.slin' to existing supported file extensions: .sln and .raw ASTERISK-30465 Change-Id: Ice848addc03a64c8404b87cb5d3b13399c57e496	2 years ago
Mike Bradeen	76da3c71a7	cli: increase channel column width For 'core show channels', the Channel name field is increased to 64 characters and the Location name field is increased to 32 characters. For 'core show channels verbose', the Channel name field is increased to 80 characters, the Context is increased to 24 characters and the Extension is increased to 24 characters. ASTERISK-30455 Change-Id: Ibec3742ce360ffc93bc56e9984c2a21dabc4d5e1	2 years ago
Jaco Kroon	71e317f68f	app_queue: periodic announcement configurable start time. This newly introduced periodic-announce-startdelay makes it possible to configure the initial start delay of the first periodic announcement after which periodic-announce-frequency takes over. ASTERISK-30437 #close Change-Id: Ia79984b6377ef78f167ad9ea2ac084bec29955d0 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Naveen Albert	cec0c5cd39	app_senddtmf: Add SendFlash AMI action. Adds an AMI action to send a flash event on a channel. ASTERISK-30440 #close Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9	2 years ago
Boris P. Korzun	e971396546	http.c: Minor simplification to HTTP status output. Change the HTTP status page (located at /httpstatus by default) by: * Combining the address and port into a single line. * Changing "SSL" to "TLS" ASTERISK-30433 #close Change-Id: Id2ccb1218f00a68424aca2b651647d8b1f549bcb	2 years ago
Holger Hans Peter Freyther	92849c8c62	res_http_media_cache: Introduce options and customize Make the existing CURL parameters configurable and allow to specify the usable protocols, proxy and DNS timeout. ASTERISK-30340 Change-Id: I2eb02ef44190e026716720419bcbdbcc8125777b	2 years ago
cmaj	fd95998b56	res_phoneprov.c: Multihomed SERVER cache prevention Phones moving between subnets on multi-homed server have their initially connected interface IP cached in the SERVER variable, even when it is not specified in the configuration files. This prevents phones from obtaining the correct SERVER variable value when they move to another subnet. ASTERISK-30388 #close Reported-by: cmaj Change-Id: I1d18987a9d58e85556b4c4a6814ce7006524cc92	2 years ago
Mike Bradeen	18a9540520	app_read: Add an option to return terminator on empty digits. Adds 'e' option to allow Read() to return the terminator as the dialed digits in the case where only the terminator is entered. ie; if "#" is entered, return "#" if the 'e' option is set and "" if it is not. ASTERISK-30411 Change-Id: I49f3221824330a193a20c660f99da0f1fc2cbbc5	2 years ago
Mike Bradeen	152239b978	app_directory: Add a 'skip call' option. Adds 's' option to skip calling the extension and instead set the extension as DIRECTORY_EXTEN channel variable. ASTERISK-30405 Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299	2 years ago
Mike Bradeen	cee3564b45	app_senddtmf: Add option to answer target channel. Adds a new option to SendDTMF() which will answer the specified channel if it is not already up. If no channel is specified, the current channel will be answered instead. ASTERISK-30422 Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085	2 years ago
Naveen Albert	78f5a41d64	app_signal: Add signaling applications Adds the Signal and WaitForSignal applications, which can be used for inter-channel signaling in the dialplan. Signal supports sending a signal to other channels listening for a signal of the same name, with an optional data payload. The signal is received by all channels waiting for that named signal. ASTERISK-29810 #close Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3	2 years ago
Naveen Albert	055e0a1571	func_json: Enhance parsing capabilities of JSON_DECODE Adds support for arrays to JSON_DECODE by allowing the user to print out entire arrays or index a particular key or print the number of keys in a JSON array. Additionally, adds support for recursively iterating a JSON tree in a single function call, making it easier to parse JSON results with multiple levels. A maximum depth is imposed to prevent potentially blowing the stack. Also fixes a bug with the unit tests causing an empty string to be printed instead of the actual test result. ASTERISK-29913 #close Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b	2 years ago
Naveen Albert	d1bec3623e	res_pjsip_session: Add overlap_context option. Adds the overlap_context option, which can be used to explicitly specify a context to use for overlap dialing extension matches, rather than forcibly using the context configured for the endpoint. ASTERISK-30262 #close Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db	2 years ago
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	2 years ago
Sean Bright	cbaba132a7	app_playback.c: Fix PLAYBACKSTATUS regression. In Asterisk 11, if a channel was redirected away during Playback(), the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 (specifically commit `7d9871b394`) that behavior was inadvertently changed and the same operation would result in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 behavior has been restored. Partial fix for ASTERISK~25661. Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff	2 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	2 years ago
Igor Goncharovsky	410150235a	res_pjsip_rfc3326: Add SIP causes support for RFC3326 Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	2 years ago
George Joseph	7dc8773178	res_rtp_asterisk: Asterisk Media Experience Score (MES) ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	2 years ago
George Joseph	3a3d6c7dcb	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `e66c5da145`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I12853c5b1d3a77f5b9200f41908fd238a17159dc	2 years ago
Naveen Albert	f7726430b2	app_broadcast: Add Broadcast application Adds a new application, Broadcast, which can be used for one-to-many transmission and many-to-one reception of channel audio in Asterisk. This is similar to ChanSpy, except it is designed for multiple channel targets instead of a single one. This can make certain kinds of audio manipulation more efficient and streamlined. New kinds of audio injection impossible with ChanSpy are also made possible. ASTERISK-30180 #close Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726	2 years ago
Naveen Albert	a46d5f9b76	app_cdr: Remove deprecated application and option. This removes the deprecated NoCDR application, which was deprecated in Asterisk 12, having long been fully superseded by the CDR_PROP function. The deprecated e option to ResetCDR is also removed for the same reason. ASTERISK-30371 #close Change-Id: Id9ed094d8e4baf98bcbc610035c2295bfafe9ec0	2 years ago
Mike Bradeen	4095a382da	chan_sip: Remove deprecated module. ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996	2 years ago
George Joseph	e66c5da145	res_rtp_asterisk: Asterisk Media Experience Score (MES) This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	2 years ago
Naveen Albert	f86d2a211c	pbx_builtins: Remove deprecated and defunct functionality. This removes the ImportVar and SetAMAFlags applications which have been deprecated since Asterisk 12, but were never removed previously. Additionally, it removes remnants of defunct options that themselves were removed years ago. ASTERISK-30335 #close Change-Id: I749520c7b08d4c9d5eebbf640d4fbc81950eda8d	2 years ago
Mike Bradeen	de3ce178ab	chan_alsa: Remove deprecated module. ASTERISK-30298 Change-Id: I5c8afb781528afdf55d237e3bffa5e4a862ae8c7	2 years ago
Michael Kuron	6b8d3cb89a	manager: AOC-S support for AOCMessage ASTERISK-21502 Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08	2 years ago
Mike Bradeen	89a7d30a97	chan_mgcp: Remove deprecated module. Also removes res_pktcops to avoid merge conflicts with ASTERISK~30301. ASTERISK-30299 Change-Id: I41a316d327646a197b6f112f7f637aceb5111b41	2 years ago
Michael Kuron	841107f294	res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip chan_sip supported sending AOC-D and AOC-E information in SIP INFO messages in an "AOC" header in a format that was originally defined by Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC format that is supported by devices from multiple vendors, including Snom phones with firmware >= 8.4.2 (released in 2010). This commit adds a new res_pjsip_aoc module that inserts AOC information into outgoing messages or sends SIP INFO messages as described below. It also fixes a small issue in res_pjsip_session which didn't always call session supplements on outgoing_response. * AOC-S in the 180/183/200 responses to an INVITE request * AOC-S in SIP INFO (if a 200 response has already been sent or if the INVITE was sent by Asterisk) * AOC-D in SIP INFO * AOC-D in the 200 response to a BYE request (if the client hangs up) * AOC-D in a BYE request (if Asterisk hangs up) * AOC-E in the 200 response to a BYE request (if the client hangs up) * AOC-E in a BYE request (if Asterisk hangs up) The specification defines one more, AOC-S in an INVITE request, which is not implemented here because it is not currently possible in Asterisk to have AOC data ready at this point in call setup. Once specifying AOC-S via the dialplan or passing it through from another SIP channel's INVITE is possible, that might be added. The SIP INFO requests are sent out immediately when the AOC indication is received. The others are inserted into an appropriate outgoing message whenever that is ready to be sent. In the latter case, the XML is stored in a channel variable at the time the AOC indication is received. Depending on where the AOC indications are coming from (e.g. PRI or AMI), it may not always be possible to guarantee that the AOC-E is available in time for the BYE. Successfully tested AOC-D and both variants of AOC-E with a Snom D735 running firmware 10.1.127.10. It does not appear to properly support AOC-S however, so that could only be tested by inspecting SIP traces. ASTERISK-21502 #close Reported-by: Matt Jordan <mjordan@digium.com> Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333	2 years ago
Naveen Albert	1c5738771d	res_hep: Add support for named capture agents. Adds support for the capture agent name field of the Homer protocol to Asterisk by allowing users to specify a name that will be sent to the HEP server. ASTERISK-30322 #close Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b	2 years ago
Naveen Albert	e3ea1b88ff	app_if: Adds conditional branch applications Adds the If, ElseIf, Else, ExitIf, and EndIf applications for conditional execution of a block of dialplan, similar to the While, EndWhile, and ExitWhile applications. The appropriate branch is executed at most once if available and may be broken out of while inside. ASTERISK-29497 Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49	2 years ago
Naveen Albert	c3cf0cd388	res_pjsip_header_funcs: Add custom parameter support. Adds support for custom URI and header parameters in the From header in PJSIP. Parameters can be both set and read using this function. ASTERISK-30150 #close Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428	2 years ago
Naveen Albert	9e14523ca3	app_voicemail: Fix missing email in msg_create_from_file. msg_create_from_file currently does not dispatch emails, which means that applications using this function, such as MixMonitor, will not trigger notifications to users (only AMI events are sent our currently). This is inconsistent with other ways users can receive voicemail. This is fixed by adding an option that attempts to send an email and falling back to just the notifications as done now if that fails. The existing behavior remains the default. ASTERISK-30283 #close Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7	2 years ago
Naveen Albert	1da5eb3795	xmldoc: Allow XML docs to be reloaded. The XML docs are currently only loaded on startup with no way to update them during runtime. This makes it impossible to load modules that use ACO/Sorcery (which require documentation) if they are added to the source tree and built while Asterisk is running (e.g. external modules). This adds a CLI command to reload the XML docs during runtime so that documentation can be updated without a full restart of Asterisk. ASTERISK-30289 #close Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07	2 years ago
Mike Bradeen	d0140fc7fe	app_osplookup: Remove deprecated module. ASTERISK-30302 Change-Id: I2268189646fa0b587675d8619322818143172474	2 years ago
Mike Bradeen	8d652ab4be	chan_skinny: Remove deprecated module. ASTERISK-30300 Change-Id: I8be11455010b8ec552e62b0719368342e8a1bae9	2 years ago
Naveen Albert	1c8acdb1c0	app_mixmonitor: Add option to use real Caller ID for voicemail. MixMonitor currently uses the Connected Line as the Caller ID for voicemails. This is due to the implementation being written this way for use with Digium phones. However, in general this is not correct for generic usage in the dialplan, and people may need the real Caller ID instead. This adds an option to do that. ASTERISK-30286 #close Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c	2 years ago
Mike Bradeen	c59eb7e6d8	manager: prevent file access outside of config dir Add live_dangerously flag to manager and use this flag to determine if a configuation file outside of AST_CONFIG_DIR should be read. ASTERISK-30176 Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404 (cherry picked from commit `81f10e847e`)	2 years ago
Naveen Albert	b1d21f7667	pbx_builtins: Allow Answer to return immediately. The Answer application currently waits for up to 500ms for media, even if users specify a different timeout. This adds an option to not wait for media on the channel by doing a raw answer instead. The default 500ms threshold is also documented. ASTERISK-30308 #close Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4	2 years ago
Naveen Albert	67186aad56	chan_dahdi: Allow FXO channels to start immediately. Currently, chan_dahdi will wait for at least one ring before an incoming call can enter the dialplan. This is generally necessary in order to receive the Caller ID spill and/or distinctive ringing detection. However, if neither of these is required, then there is nothing gained by waiting for one ring and this unnecessarily delays call setup. Users can now use immediate=yes to make FXO channels (FXS signaled) begin processing dialplan as soon as Asterisk receives the call. ASTERISK-30305 #close Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5	2 years ago
Naveen Albert	7e1340eb77	app_mixmonitor: Add option to delete files on exit. Adds an option that allows MixMonitor to delete its copy of any recording files before exiting. This can be handy in conjunction with options like m, which copy the file elsewhere, and the original files may no longer be needed. ASTERISK-30284 #close Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e	3 years ago
Naveen Albert	5e35862109	translate.c: Prefer better codecs upon translate ties. If multiple codecs are available for the same resource and the translation costs between multiple codecs are the same, ties are currently broken arbitrarily, which means a lower quality codec would be used. This forces Asterisk to explicitly use the higher quality codec, ceteris paribus. ASTERISK-29455 Change-Id: I4b7297e1baca7aac14fe4a3c7538e18e2dbe9fd6	3 years ago

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972 Commits (d2b6248196b018f724e877c0d456152b2925af07)