asterisk

Commit Graph

Author	SHA1	Message	Date
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	2 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
Naveen Albert	8c791f9a65	app_amd: Add option to play audio during AMD. Adds an option that will play an audio file to the party while AMD is running on the channel, so the called party does not just hear silence. ASTERISK-30179 #close Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6	3 years ago
Naveen Albert	1ed4518328	func_export: Add EXPORT function Adds the EXPORT function, which allows write access to variables and functions on other channels. ASTERISK-29432 #close Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf	3 years ago
Maximilian Fridrich	5bbad0d27c	res_pjsip: Add 100rel option "peer_supported". This patch adds a new option to the 100rel parameter for pjsip endpoints called "peer_supported". When an endpoint with this option receives an incoming request and the request indicated support for the 100rel extension, then Asterisk will send 1xx responses reliably. If the request did not indicate 100rel support, Asterisk sends 1xx responses normally. ASTERISK-30158 Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad	3 years ago
Naveen Albert	ab1dbfef75	func_strings: Add trim functions. Adds TRIM, LTRIM, and RTRIM, which can be used for trimming leading and trailing whitespace from strings. ASTERISK-30222 #close Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554	3 years ago
Asterisk Development Team	f01ed3eea4	Update CHANGES and UPGRADE.txt for 20.0.0	3 years ago
Mike Bradeen	7a44296ca9	res_pjsip: Add user=phone on From and PAID for usereqphone=yes Adding user=phone to local-side uri's when user_eq_phone=yes is set for an endpoint. Previously this would only add the header to the To and R-URI. ASTERISK-30178 Change-Id: Id3bfb5d225d762e7d2668c023fe09e4541ae8600	3 years ago
sungtae kim	80bc844fd6	res_musiconhold: Add option to not play music on hold on unanswered channels This change adds an option, answeredonly, that will prevent music on hold on channels that are not answered. ASTERISK-30135 Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08	3 years ago
Philip Prindeville	d13afaf302	res_crypto: Don't load non-regular files in keys directory ASTERISK-30046 Change-Id: Ie77e0648f8b0b1c2159fb24662d1989cfd4cc36d	3 years ago
Naveen Albert	c487425620	lock.c: Add AMI event for deadlocks. Adds an AMI event to indicate that a deadlock has likely started, when Asterisk is compiled with DETECT_DEADLOCKS enabled. This can make it easier to perform automated deadlock detection and take appropriate action (such as doing a core dump). Unlike the deadlock warnings, the AMI event is emitted only once per deadlock. ASTERISK-30161 #close Change-Id: Ifc6ed3e390f8b4cff7f8077a50e4d7a5b54e42fb	3 years ago
Naveen Albert	205c7c8d21	app_confbridge: Add end_marked_any option. Adds the end_marked_any option, which can be used to kick a user from a conference if any marked user leaves. ASTERISK-30211 #close Change-Id: I9e8da7ccb892e522546c0f2b5476d172e022c2f5	3 years ago
George Joseph	05f42806cc	res_geolocation: Add two new options to GEOLOC_PROFILE Added an 'a' option to the GEOLOC_PROFILE function to allow variable lists like location_info_refinement to be appended to instead of replacing the entire list. Added an 'r' option to the GEOLOC_PROFILE function to resolve all variables before a read operation and after a Set operation. Added a few missing parameters to the ones allowed for writing with GEOLOC_PROFILE. Fixed a bug where calling GEOLOC_PROFILE to read a parameter might actually update the profile object. Cleaned up XML documentation a bit. ASTERISK-30190 Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c	3 years ago
George Joseph	c799db6a21	res_geolocation: Allow location parameters on the profile object You can now specify the location object's format, location_info, method, location_source and confidence parameters directly on a profile object for simple scenarios where the location information isn't common with any other profiles. This is mutually exclusive with setting location_reference on the profile. Updated appdocsxml.dtd to allow xi:include in a configObject element. This makes it easier to link to complete configOptions in another object. This is used to add the above fields to the profile object without having to maintain the option descriptions in two places. ASTERISK-30185 Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569	3 years ago
George Joseph	4ffc5561c4	res_geolocation: Add profile parameter suppress_empty_ca_elements Added profile parameter "suppress_empty_ca_elements" that will cause Civic Address elements that are empty to be suppressed from the outgoing PIDF-LO document. Fixed a possible SEGV if a sub-parameter value didn't have a value. ASTERISK-30177 Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd	3 years ago
George Joseph	2d5a6498dd	res_geolocation: Add built-in profiles The trigger to perform outgoing geolocation processing is the presence of a geoloc_outgoing_call_profile on an endpoint. This is intentional so as to not leak location information to destinations that shouldn't receive it. In a totally dynamic configuration scenario however, there may not be any profiles defined in geolocation.conf. This makes it impossible to do outgoing processing without defining a "dummy" profile in the config file. This commit adds 4 built-in profiles: "<prefer_config>" "<discard_config>" "<prefer_incoming>" "<discard_incoming>" The profiles are empty except for having their precedence set and can be set on an endpoint to allow processing without entries in geolocation.conf. "<discard_config>" is actually the best one to use in this situation. ASTERISK-30182 Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a	3 years ago

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951 Commits (20.2)