asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	8f5581b0d0	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	10b3c2dc45	doc: Remove obsolete CHANGES-staging and UPGRADE-staging	2 years ago
Naveen Albert	cfae64a70d	asterisk.c: Fix option warning for remote console. 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	5c6d5ea38f	core: Cleanup gerrit and JIRA references. (#57 ) * 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	de15852ef0	Revert "app_queue: periodic announcement configurable start time." This reverts commit `3fd0b65bae`. Reason for revert: Causes segmentation fault. Change-Id: Ic189c6f7872943a5500d3e71142f0c09d54fcc31	2 years ago
Naveen Albert	b33f92cbb5	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	a0fd95ef52	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	ffe346b2de	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	fa635a872e	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	8d2ffc8aa5	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
Naveen Albert	3556ca239a	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
Mike Bradeen	405211eff7	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	3fd0b65bae	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
Holger Hans Peter Freyther	8f088aa0f7	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
Asterisk Development Team	93813c9dca	Update CHANGES and UPGRADE.txt for 20.2.0	2 years ago
Mike Bradeen	5c11d7adea	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
cmaj	5b0e3444c3	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	2308afed8e	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	98742388b6	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	88b2c741ca	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	8a45cd7af4	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	a1da8042d1	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
Sean Bright	ef16eaee36	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
Igor Goncharovsky	3526441e41	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	4710f37ef6	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	62ca063fca	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `d454801c2d`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86	2 years ago
Naveen Albert	e06fe8e344	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
George Joseph	d454801c2d	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
Asterisk Development Team	fefc236e7c	Update CHANGES and UPGRADE.txt for 20.1.0	2 years ago
Michael Kuron	5c114dcb4a	manager: AOC-S support for AOCMessage ASTERISK-21502 Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08	2 years ago
Michael Kuron	fee9012fe1	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	b9c031c1f8	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	531eacd6c9	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	b365ea8601	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	406143ae61	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	52c7d3ed07	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
Naveen Albert	691178c48e	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	81f10e847e	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	2 years ago
Naveen Albert	c7df5ee7c1	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	5ede4e217a	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	6e59b01e1a	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
Henning Westerholt	12445040d3	res_pjsip: return all codecs on a re-INVITE without SDP Currently chan_pjsip on receiving a re-INVITE without SDP will only return the codecs that are previously negotiated and not offering all enabled codecs. This causes interoperability issues with different equipment (e.g. from Cisco) for some of our customers and probably also in other scenarios involving 3PCC infrastructure. According to RFC 3261, section 14.2 we SHOULD return all codecs on a re-INVITE without SDP The PR proposes a new parameter to configure this behaviour: all_codecs_on_empty_reinvite. It includes the code, documentation, alembic migrations, CHANGES file and example configuration additions. ASTERISK-30193 #close Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148	3 years ago
Naveen Albert	40b52322e5	res_pjsip_notify: Add option support for AMI. The PJSIP notify CLI commands allow for using "options" configured in pjsip_notify.conf. This allows these same options to be used in AMI actions as well. Additionally, as part of this improvement, some repetitive common code is refactored. ASTERISK-30263 #close Change-Id: Ie4496b322b63b61eaf9672183a959ab99a04b6b5	3 years ago
Naveen Albert	c32b39d123	res_pjsip_logger: Add method-based logging option. Expands the pjsip logger to support the ability to filter by SIP message method. This can make certain types of SIP debugging easier by only logging messages of particular method(s). ASTERISK-30146 #close Co-authored-by: Sean Bright <sean@seanbright.com> Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f	3 years ago
Naveen Albert	b331caca30	cdr: Allow bridging and dial state changes to be ignored. Allows bridging, parking, and dial messages to be globally ignored for all CDRs such that only a single CDR record is generated per channel. This is useful when CDRs should endure for the lifetime of an entire channel and bridging and dial updates in the dialplan should not result in multiple CDR records being created for the call. With the ignore bridging option, bridging changes have no impact on the channel's CDRs. With the ignore dial state option, multiple Dials and their outcomes have no impact on the channel's CDRs. The last disposition on the channel is preserved in the CDR, so the actual disposition of the call remains available. These two options can reduce the amount of "CDR hacks" that have hitherto been necessary to ensure that CDR was not "spoiled" by these messages if that was undesired, such as putting a dummy optimization-disabled local channel between the caller and the actual call and putting the CDR on the channel in the middle to ensure that CDR would persist for the entire call and properly record start, answer, and end times. Enabling these options is desirable when calls correspond to the entire lifetime of channels and the CDR should reflect that. Current default behavior remains unchanged. ASTERISK-30091 #close Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832	3 years ago
Naveen Albert	e0e7f35730	res_tonedetect: Add ringback support to TONE_DETECT. Adds support for detecting audible ringback tone to the TONE_DETECT function using the p option. ASTERISK-30254 #close Change-Id: Ie2329ff245248768367d26749c285fbe823f6414	3 years ago
Maximilian Fridrich	0d2e140123	res_pjsip: Add mediasec capabilities. This patch adds support for mediasec SIP headers and SDP attributes. These are defined in RFC 3329, 3GPP TS 24.229 and draft-dawes-sipcore-mediasec-parameter. The new features are implemented so that a backbone for RFC 3329 is present to streamline future work on RFC 3329. With this patch, Asterisk can communicate with Deutsche Telekom trunks which require these fields. ASTERISK-30032 Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2	3 years ago
Asterisk Development Team	7f80830ced	Update CHANGES and UPGRADE.txt for 20.0.0	3 years ago
Naveen Albert	a5ec60e6c6	features: Add no answer option to Bridge. Adds the n "no answer" option to the Bridge application so that answer supervision can not automatically be provided when Bridge is executed. Additionally, a mechanism (dialplan variable) is added to prevent bridge targets (typically the target of a masquerade) from answering the channel when they enter the bridge. ASTERISK-30223 #close Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d	3 years ago
Naveen Albert	1e29607b5c	app_bridgewait: Add option to not answer channel. Adds the n option to not answer the channel when calling BridgeWait, so the application can be used without forcing answer supervision. ASTERISK-30216 #close Change-Id: I6b85ef300b1f7b5170f8537e2b10889cc2e6605a	3 years ago

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965 Commits (b4eb773d5ea9befa25e8045a57dba550a690e541)