asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	0007625ad0	logger: Add channel-based filtering. This adds the ability to filter console logging by channel or groups of channels. This can be useful on busy systems where an administrator would like to analyze certain calls in detail. A dialplan function is also included for the purpose of assigning a channel to a group (e.g. by tenant, or some other metric). ASTERISK-30483 #close Resolves: #242 UserNote: The console log can now be filtered by channels or groups of channels, using the logger filter CLI commands. (cherry picked from commit `a0fc8d1b5f`)	1 year ago
Sean Bright	c2680f63c5	chan_iax2.c: Don't send unsanitized data to the logger. This resolves an issue where non-printable characters could be sent to the console/log files. (cherry picked from commit `d2afb10eed`)	1 year ago
George Joseph	1d87c27cab	codec_ilbc: Disable system ilbc if version >= 3.0.0 Fedora 37 started shipping ilbc 3.0.4 which we don't yet support. configure.ac now checks the system for "libilbc < 3" instead of just "libilbc". If true, the system version of ilbc will be used. If not, the version included at codecs/ilbc will be used. Resolves: #84 (cherry picked from commit `d819a6bccb`)	1 year ago
Sean Bright	da35b6a244	resource_channels.c: Explicit codec request when creating UnicastRTP. Fixes #394 (cherry picked from commit `a83c761c95`)	1 year ago
Sean Bright	fb7d39db6d	doc: Update IP Quality of Service links. Fixes #328 (cherry picked from commit `26918d05f4`)	1 year ago
George Joseph	94f931a6d7	chan_pjsip: Add PJSIPHangup dialplan app and manager action See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range. (cherry picked from commit `cd77953172`)	1 year ago
Sean Bright	f96d7ef7b5	chan_iax2.c: Ensure all IEs are displayed when dumping frame contents. When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE in a frame was one that may not have any data - such as the CALLTOKEN IE in an NEW request - it was not getting displayed. (cherry picked from commit `0e126b3841`)	1 year ago
Naveen Albert	e75aebc9bc	chan_dahdi: Warn if nonexistent cadence is requested. If attempting to ring a channel using a nonexistent cadence, emit a warning, before falling back to the default cadence. Resolves: #409 (cherry picked from commit `4b9a4483fc`)	1 year ago
Holger Hans Peter Freyther	7699af00e1	stasis: Update the snapshot after setting the redirect The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest. (cherry picked from commit `56733c73b4`)	1 year ago
Holger Hans Peter Freyther	28f52d35f3	ari: Provide the caller ID RDNIS for the channels Provide the caller ID RDNIS when available. This will allow an application to follow the redirect. (cherry picked from commit `157389bc59`)	1 year ago
Brad Smith	089ddaaaed	main/utils: Implement ast_get_tid() for OpenBSD Implement the ast_get_tid() function for OpenBSD. OpenBSD supports getting the TID via getthrid(). (cherry picked from commit `e7943dd4d9`)	1 year ago
Brad Smith	fb3067dd71	res_rtp_asterisk.c: Fix runtime issue with LibreSSL The module will fail to load. Use proper function DTLS_method() with LibreSSL. (cherry picked from commit `65d38c8104`)	1 year ago
Naveen Albert	6a4fe8bdab	app_directory: Add ADSI support to Directory. This adds optional ADSI support to the Directory application, which allows callers with ADSI CPE to navigate the Directory system significantly faster than is possible using the audio prompts. Callers can see the directory name (and optionally extension) on their screenphone and confirm or reject a match immediately rather than waiting for it to be spelled out, enhancing usability. Resolves: #356 (cherry picked from commit `5046620fa3`)	1 year ago
Naveen Albert	b47a403b80	core_local: Fix local channel parsing with slashes. Currently, trying to call a Local channel with a slash in the extension will fail due to the parsing of characters after such a slash as being dial modifiers. Additionally, core_local is inconsistent and incomplete with its parsing of Local dial strings in that sometimes it uses the first slash and at other times it uses the last. For instance, something like DAHDI/5 or PJSIP/device is a perfectly usable extension in the dialplan, but Local channels in particular prevent these from being called. This creates inconsistent behavior for users, since using a slash in an extension is perfectly acceptable, and using a Goto to accomplish this works fine, but if specified through a Local channel, the parsing prevents this. This fixes this by explicitly parsing options from the last slash in the extension, rather than the first one, which doesn't cause an issue for extensions with slashes. ASTERISK-30013 #close Resolves: #248 (cherry picked from commit `2191a0d33f`)	1 year ago
Mark Murawski	3b70dfcced	Remove files that are no longer updated Fixes: #360 (cherry picked from commit `2ed8daa3cb`)	1 year ago
Naveen Albert	b5eb5e19c3	app_voicemail: Add AMI event for mailbox PIN changes. This adds an AMI event that is emitted whenever a mailbox password is successfully changed, allowing AMI consumers to process these. UserNote: The VoicemailPasswordChange event is now emitted whenever a mailbox password is updated, containing the mailbox information and the new password. Resolves: #398 (cherry picked from commit `cf9d7fa9f6`)	1 year ago
Sean Bright	3ded0c95fc	app_queue.c: Emit unpause reason with PauseQueueMember event. Fixes #395 (cherry picked from commit `fb937d1d89`)	1 year ago
George Joseph	e2f246c067	bridge_simple: Suppress unchanged topology change requests In simple_bridge_join, we were sending topology change requests even when the new and old topologies were the same. In some circumstances, this can cause unnecessary re-invites and even a re-invite flood. We now suppress those. Resolves: #384 (cherry picked from commit `f301d4559e`)	1 year ago
Naveen Albert	120dbba2b3	res_pjsip: Include cipher limit in config error message. If too many ciphers are specified in the PJSIP config, include the maximum number of ciphers that may be specified in the user-facing error message. Resolves: #396 (cherry picked from commit `8d9d0d3738`)	1 year ago
Mike Bradeen	0b4427d621	res_speech: allow speech to translate input channel * Allow res_speech to translate the input channel if the format is translatable to a format suppored by the speech provider. Resolves: #129 UserNote: res_speech now supports translation of an input channel to a format supported by the speech provider, provided a translation path is available between the source format and provider capabilites. (cherry picked from commit `c3e2bff36e`)	1 year ago
Sean Bright	ea74f942ce	res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. Fixes #386 (cherry picked from commit `deeb1acffe`)	1 year ago
Sean Bright	58f78d2d0e	res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. Fixes #376 (cherry picked from commit `d4e4942cf5`)	1 year ago
George Joseph	027c748e57	api.wiki.mustache: Fix indentation in generated markdown The '*' list indicator for default values and allowable values for path, query and POST parameters need to be indented 4 spaces instead of 2. Should resolve issue 38 in the documentation repo. (cherry picked from commit `b619b64137`)	1 year ago
Sean Bright	93697458a6	pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. Per RFC8827: Implementations MUST NOT implement DTLS renegotiation and MUST reject it with a "no_renegotiation" alert if offered. So we disable it when webrtc=yes is set. Fixes #378 UpgradeNote: The dtls_rekey will be disabled if webrtc support is requested on an endpoint. A warning will also be emitted. (cherry picked from commit `9d329da346`)	1 year ago
Samuel Olaechea	59181860ec	configs: Fix typo in pjsip.conf.sample. (cherry picked from commit `ebc78a83be`)	1 year ago
George Joseph	e52175341e	res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown Commit `f66f77f` last year prevents the res_pjsip_exten_state and res_pjsip_mwi modules from unloading due to possible pjproject asserts if the modules are reloaded. A side effect of the implementation is that the taskprocessors these modules use aren't being released. When asterisk is doing a graceful shutdown, it waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all taskprocessors to stop but since those 2 modules don't release theirs, the shutdown hangs for that amount of time. This change allows the modules to be unloaded and their resources to be released when ast_shutdown_final is true. Resolves: #379 (cherry picked from commit `9efc4bdfbc`)	1 year ago
sungtae kim	faf046ea86	res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters This commit introduces an extension to the endpoint and relevant resource sizes for PJSIP, transitioning from its current 40-character constraint to a more versatile 255-character capacity. This enhancement significantly overcomes limitations related to domain qualification and practical usage, ultimately delivering improved functionality. In addition, it includes adjustments to accommodate the expanded realm size within the ARI, specifically enhancing the maximum realm length. Resolves: #345 UserNote: With this update, the PJSIP realm lengths have been extended to support up to 255 characters. UpgradeNote: As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas. (cherry picked from commit `f89e56c178`)	1 year ago
George Joseph	95656c409d	.github: PRSubmitActions: Fix adding reviewers to PR (cherry picked from commit `be1e83a6ac`)	1 year ago
George Joseph	13f2d126fd	.github: New PR Submit workflows The workflows that get triggered when PRs are submitted or updated have been replaced with ones that are more secure and have a higher level of parallelism. (cherry picked from commit `289aa1840e`)	1 year ago
George Joseph	2cb140f801	.github: New PR Submit workflows The workflows that get triggered when PRs are submitted or updated have been replaced with ones that are more secure and have a higher level of parallelism. (cherry picked from commit `0c1c6e9ada`)	1 year ago
Mike Bradeen	0f2eb00a52	res_stasis: signal when new command is queued res_statsis's app loop sleeps for up to .2s waiting on input to a channel before re-checking the command queue. This can cause delays between channel setup and bridge. This change is to send a SIGURG on the sleeping thread when a new command is enqueued. This exits the sleeping thread out of the ast_waitfor() call triggering the new command being processed on the channel immediately. Resolves: #362 UserNote: Call setup times should be significantly improved when using ARI. (cherry picked from commit `79220e3f0c`)	1 year ago
Holger Hans Peter Freyther	1f7e9bfad5	ari/stasis: Indicate progress before playback on a bridge Make it possible to start a playback and the calling party to receive audio on a bridge before the call is connected. Model the implementation after play_on_channel and deliver a AST_CONTROL_PROGRESS before starting the playback. For a PJSIP channel this will result in sending a SIP 183 Session Progress. (cherry picked from commit `1171dcee02`)	1 year ago
Sean Bright	d7a01afa50	func_curl.c: Ensure channel is locked when manipulating datastores. (cherry picked from commit `acb2348f90`)	1 year ago
George Joseph	fd487b48a3	.github: Fix job prereqs in PROpenedUpdated (cherry picked from commit `20398e8e95`)	1 year ago
George Joseph	e1ce35c3f5	.github: Block PR tests until approved (cherry picked from commit `e9abf11a26`)	1 year ago
Joshua C. Colp	b7cc7e804b	Update config.yml (cherry picked from commit `ef7b0f4c3b`)	1 year ago
George Joseph	f0edab9f3e	logger.h: Add ability to change the prefix on SCOPE_TRACE output You can now define the _TRACE_PREFIX_ macro to change the default trace line prefix of "file:line function" to something else. Full documentation in logger.h. (cherry picked from commit `b52e07ee1b`)	1 year ago
George Joseph	393d8cb947	Add libjwt to third-party The current STIR/SHAKEN implementation is not currently usable due to encryption issues. Rather than trying to futz with OpenSSL and the the current code, we can take advantage of the existing capabilities of libjwt but we first need to add it to the third-party infrastructure already in place for jansson and pjproject. A few tweaks were also made to the third-party infrastructure as a whole. The jansson "dest" install directory was renamed "dist" to better match convention, and the third-party Makefile was updated to clean all product directories not just the ones currently in use. Resolves: #349 (cherry picked from commit `a9d4175e1d`)	1 year ago
Mike Bradeen	0df2848147	res_pjsip: update qualify_timeout documentation with DNS note The documentation on qualify_timeout does not explicitly state that the timeout includes any time required to perform any needed DNS queries on the endpoint. If the OPTIONS response is delayed due to the DNS query, it can still render an endpoint as Unreachable if the net time is enough for qualify_timeout to expire. Resolves: #352 (cherry picked from commit `933490b758`)	1 year ago
Naveen Albert	c06fd10f0a	chan_dahdi: Clarify scope of callgroup/pickupgroup. Internally, chan_dahdi only applies callgroup and pickupgroup to FXO signalled channels, but this is not documented anywhere. This is now documented in the sample config, and a warning is emitted if a user tries configuring these settings for channel types that do not support these settings, since they will not have any effect. Resolves: #294 (cherry picked from commit `945babf25c`)	1 year ago
Bastian Triller	7a1208410a	func_json: Fix crashes for some types This commit fixes crashes in JSON_DECODE() for types null, true, false and real numbers. In addition it ensures that a path is not deeper than 32 levels. Also allow root object to be an array. Add unit tests for above cases. (cherry picked from commit `e6d5b8d8cf`)	1 year ago
Mike Bradeen	c47a386cf2	res_speech_aeap: add aeap error handling res_speech_aeap previously did not register an error handler with aeap, so it was not notified of a disconnect. This resulted in SpeechBackground never exiting upon a websocket disconnect. Resolves: #303 (cherry picked from commit `8c934fb7ed`)	1 year ago
Naveen Albert	0a834c8d75	app_voicemail: Disable ADSI if unavailable. If ADSI is available on a channel, app_voicemail will repeatedly try to use ADSI, even if there is no CPE that supports it. This leads to many unnecessary delays during the session. If ADSI is available but ADSI setup fails, we now disable it to prevent further attempts to use ADSI during the session. Resolves: #354 (cherry picked from commit `c04923fcda`)	1 year ago
Eduardo	7a2305a08b	codec_builtin: Use multiples of 20 for maximum_ms Some providers require a multiple of 20 for the maxptime or fail to complete calls, e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used. Resolves: #260 (cherry picked from commit `11d87713eb`)	1 year ago
George Joseph	9afa54b3b3	lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS. Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just causes the lock calls to loop over trylock in 200us intervals until the lock is obtained and spits out log messages if it takes more than 5 seconds. From a code perspective, the only reason they were tied together was for logging. So... The ifdefs in lock.c were refactored to allow DETECT_DEADLOCKS to be enabled without also enabling DEBUG_THREADS. Resolves: #321 UserNote: You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS. This removes a significant amount of overhead if you just want to detect possible deadlocks vs needing full lock tracing. (cherry picked from commit `7e2243f9e1`)	1 year ago
George Joseph	97674e1d20	asterisk.c: Use the euid's home directory to read/write cli history The CLI .asterisk_history file is read from/written to the directory specified by the HOME environment variable. If the root user starts asterisk with the -U/-G options, or with runuser/rungroup set in asterisk.conf, the asterisk process is started as root but then it calls setuid/setgid to set the new user/group. This does NOT reset the HOME environment variable to the new user's home directory though so it's still left as "/root". In this case, the new user will almost certainly NOT have access to read from or write to the history file. * Added function process_histfile() which calls getpwuid(geteuid()) and uses pw->dir as the home directory instead of the HOME environment variable. * ast_el_read_default_histfile() and ast_el_write_default_histfile() have been modified to use the new process_histfile() function. Resolves: #337 (cherry picked from commit `07cf37531a`)	1 year ago
Tinet-mucw	a86376fe68	res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes. From the gdb information, ast_websocket_read reads a message successfully, then transport_read is called in the serializer. During execution of pjsip_transport_down, ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop. After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages. This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue. In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop. Resolves: asterisk#299 (cherry picked from commit `edc674a6ca`)	1 year ago
Mike Bradeen	3b61583c20	cel: add publish user event helper Add a wrapper function around ast_cel_publish_event that packs event and extras into a blob before publishing Resolves:#330 (cherry picked from commit `248b92563c`)	1 year ago
Naveen Albert	3fef46e776	chan_console: Fix deadlock caused by unclean thread exit. To terminate a console channel, stop_stream causes pthread_cancel to make stream_monitor exit. However, commit `5b8fea93d1` added locking to this function which results in deadlock due to the stream_monitor thread being killed while it's holding the pvt lock. To resolve this, a flag is now set and read to indicate abort, so the use of pthread_cancel and pthread_kill can be avoided altogether. Resolves: #308 (cherry picked from commit `71215561d8`)	1 year ago
George Joseph	d5a285c8f2	file.c: Add ability to search custom dir for sounds To better co-exist with sounds files that may be managed by packages, custom sound files may now be placed in AST_DATA_DIR/sounds/custom instead of the standard AST_DATA_DIR/sounds/<lang> directory. If the new "sounds_search_custom_dir" option in asterisk.conf is set to "true", asterisk will search the custom directory for sounds files before searching the standard directory. For performance reasons, the "sounds_search_custom_dir" defaults to "false". Resolves: #315 UserNote: A new option "sounds_search_custom_dir" has been added to asterisk.conf that allows asterisk to search AST_DATA_DIR/sounds/custom for sounds files before searching the standard AST_DATA_DIR/sounds/<lang> directory. (cherry picked from commit `4493d2b2fc`)	1 year ago

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