34005 Commits (2be12e091d6f76fc001fdba555e10ba9f6fc7ccc)

 

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Author	SHA1	Message	Date
Sean Bright	3859b630a7	res_rtp_asterisk.c: Update for OpenSSL 3+. ... In 5ac5c2b0 we defined `OPENSSL_SUPPRESS_DEPRECATED` to silence deprecation warnings. This commit switches over to using non-deprecated API.	2 years ago
Sean Bright	0dcf03e844	alembic: Update list of TLS methods available on ps_transports. ... Related to #221 and #222. Also adds `*.ini` to the `.gitignore` file in ast-db-manage for convenience.	2 years ago
Naveen Albert	c222343ec6	func_channel: Expose previously unsettable options. ... Certain channel options are not set anywhere or exposed in any way to users, making them unusable. This exposes some of these options which make sense for users to manipulate at runtime. Resolves: #442	2 years ago
Sean Bright	f8212d4594	app.c: Allow ampersands in playback lists to be escaped. ... Any function or application that accepts a `&`-separated list of filenames can now include a literal `&` in a filename by wrapping the entire filename in single quotes, e.g.: ``` exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world) ``` Fixes #172 UpgradeNote: Ampersands in URLs passed to the `Playback()`, `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()` applications as filename arguments can now be escaped by single quoting the filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`.	2 years ago
Sean Bright	ff012323e8	uri.c: Simplify ast_uri_make_host_with_port()	2 years ago
Sean Bright	9e1a60727e	func_curl.c: Remove CURLOPT() plaintext documentation. ... I assume this was missed when initially converting to XML documentation and we've been kicking the can down the road since.	2 years ago
Sean Bright	f2961f048d	res_http_websocket.c: Set hostname on client for certificate validation. ... Additionally add a `assert()` to in the TLS client setup code to ensure that hostname is set when it is supposed to be. Fixes #433	2 years ago
Sean Bright	978d09fc35	live_ast: Add astcachedir to generated asterisk.conf. ... `astcachedir` (added in b0842713) was not added to `live_ast` so continued to point to the system `/var/cache` directory instead of the one in the live environment.	2 years ago
George Joseph	d10d4d9ddd	SECURITY.md: Update with correct documentation URL	2 years ago
Naveen Albert	12b353eae0	func_lock: Add missing see-also refs to documentation. ... Resolves: #423	2 years ago
Matthew Fredrickson	275f7911b5	app_followme.c: Grab reference on nativeformats before using it ... Fixes a crash due to a lack of proper reference on the nativeformats object before passing it into ast_request(). Also found potentially similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c Fixes: #388	2 years ago
Naveen Albert	1f19227eab	configs: Improve documentation for bandwidth in iax.conf. ... This improves the documentation for the bandwidth setting in iax.conf by making it clearer what the ramifications of this setting are. It also changes the sample default from low to high, since only high is compatible with good codecs that people will want to use in the vast majority of cases, and this is a common gotcha that trips up new users. Resolves: #425	2 years ago
Naveen Albert	a0fc8d1b5f	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.	2 years ago
Sean Bright	d2afb10eed	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.	2 years ago
George Joseph	d819a6bccb	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	2 years ago
Sean Bright	a83c761c95	resource_channels.c: Explicit codec request when creating UnicastRTP. ... Fixes #394	2 years ago
Sean Bright	26918d05f4	doc: Update IP Quality of Service links. ... Fixes #328	2 years ago
George Joseph	cd77953172	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.	2 years ago
Sean Bright	0e126b3841	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.	2 years ago
Naveen Albert	4b9a4483fc	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	2 years ago
Holger Hans Peter Freyther	56733c73b4	stasis: Update the snapshot after setting the redirect ... The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest.	2 years ago
Holger Hans Peter Freyther	157389bc59	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.	2 years ago
Brad Smith	e7943dd4d9	main/utils: Implement ast_get_tid() for OpenBSD ... Implement the ast_get_tid() function for OpenBSD. OpenBSD supports getting the TID via getthrid().	2 years ago
Brad Smith	65d38c8104	res_rtp_asterisk.c: Fix runtime issue with LibreSSL ... The module will fail to load. Use proper function DTLS_method() with LibreSSL.	2 years ago
Naveen Albert	5046620fa3	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	2 years ago
Naveen Albert	2191a0d33f	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	2 years ago
Mark Murawski	2ed8daa3cb	Remove files that are no longer updated ... Fixes: #360	2 years ago
Naveen Albert	cf9d7fa9f6	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	2 years ago
Sean Bright	fb937d1d89	app_queue.c: Emit unpause reason with PauseQueueMember event. ... Fixes #395	2 years ago
George Joseph	f301d4559e	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	2 years ago
Naveen Albert	8d9d0d3738	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	2 years ago
Mike Bradeen	c3e2bff36e	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.	2 years ago
Sean Bright	deeb1acffe	res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. ... Fixes #386	2 years ago
Sean Bright	d4e4942cf5	res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. ... Fixes #376	2 years ago
George Joseph	b619b64137	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.	2 years ago
Sean Bright	9d329da346	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.	2 years ago
Samuel Olaechea	ebc78a83be	configs: Fix typo in pjsip.conf.sample.	2 years ago
George Joseph	9efc4bdfbc	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	2 years ago
sungtae kim	f89e56c178	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.	2 years ago
George Joseph	be1e83a6ac	.github: PRSubmitActions: Fix adding reviewers to PR	2 years ago
George Joseph	289aa1840e	.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.	2 years ago
George Joseph	0c1c6e9ada	.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.	2 years ago
Mike Bradeen	79220e3f0c	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.	2 years ago
Holger Hans Peter Freyther	1171dcee02	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.	2 years ago
Sean Bright	acb2348f90	func_curl.c: Ensure channel is locked when manipulating datastores.	2 years ago
George Joseph	20398e8e95	.github: Fix job prereqs in PROpenedUpdated	2 years ago
George Joseph	e9abf11a26	.github: Block PR tests until approved	2 years ago
Joshua C. Colp	ef7b0f4c3b	Update config.yml	2 years ago
George Joseph	b52e07ee1b	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.	2 years ago
George Joseph	a9d4175e1d	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	2 years ago