asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	1ce9e1fa8f	sig_analog: Fix channel leak when mwimonitor is enabled. When mwimonitor=yes is enabled for an FXO port, the do_monitor thread will launch mwi_thread if it thinks there could be MWI on an FXO channel, due to the noise threshold being satisfied. This, in turns, calls analog_ss_thread_start in sig_analog. However, unlike all other instances where __analog_ss_thread is called in sig_analog, this call path does not properly set pvt->ss_astchan to the Asterisk channel, which means that the Asterisk channel is NULL when __analog_ss_thread starts executing. As a result, the thread exits and the channel is never properly cleaned up by calling ast_hangup. This caused issues with do_monitor on incoming calls, as it would think the channel was still owned even while receiving events, leading to an infinite barrage of warning messages; additionally, the channel would persist improperly. To fix this, the assignment is added to the call path where it is missing (which is only used for mwi_thread). A warning message is also added since previously there was no indication that __analog_ss_thread was exiting abnormally. This resolves both the channel leak and the condition that led to the warning messages. Resolves: #458	1 year ago
Sean Bright	83636e4b92	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.	1 year ago
Sean Bright	07e378e2c0	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.	1 year ago
Naveen Albert	c4bf59b781	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	1 year ago
Sean Bright	ca931c9436	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`.	1 year ago
Sean Bright	31c44d0634	uri.c: Simplify ast_uri_make_host_with_port()	1 year ago
Sean Bright	0b744cef15	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.	1 year ago
Sean Bright	611010d67d	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	1 year ago
Sean Bright	91c8866ff3	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.	1 year ago
George Joseph	f28047db36	SECURITY.md: Update with correct documentation URL	1 year ago
Naveen Albert	5f115e425c	func_lock: Add missing see-also refs to documentation. Resolves: #423	1 year ago
Matthew Fredrickson	21412fddcb	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	1 year ago
Naveen Albert	44b955e237	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	1 year ago
Naveen Albert	89709e2583	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.	1 year ago
Sean Bright	fb74ef1c7a	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.	1 year ago
George Joseph	36f8490901	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	1 year ago
Sean Bright	c19470497f	resource_channels.c: Explicit codec request when creating UnicastRTP. Fixes #394	1 year ago
Sean Bright	c040179fcf	doc: Update IP Quality of Service links. Fixes #328	1 year ago
George Joseph	f309ffad3d	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.	1 year ago
Sean Bright	52968038ed	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.	1 year ago
Naveen Albert	293577f1d9	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	1 year ago
Holger Hans Peter Freyther	69cf329681	stasis: Update the snapshot after setting the redirect The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest.	1 year ago
Holger Hans Peter Freyther	69590ba33e	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.	1 year ago
Brad Smith	5c22cfccd9	main/utils: Implement ast_get_tid() for OpenBSD Implement the ast_get_tid() function for OpenBSD. OpenBSD supports getting the TID via getthrid().	1 year ago
Brad Smith	6ec59e1e04	res_rtp_asterisk.c: Fix runtime issue with LibreSSL The module will fail to load. Use proper function DTLS_method() with LibreSSL.	1 year ago
Naveen Albert	4657163c56	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	82086545cc	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	fcb9feddfd	Remove files that are no longer updated Fixes: #360	2 years ago
Naveen Albert	d678370b54	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	6c6137028b	app_queue.c: Emit unpause reason with PauseQueueMember event. Fixes #395	2 years ago
George Joseph	ed4978ae8b	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	b94f8bb216	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	f666dd0dd8	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	99527745eb	res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. Fixes #386	2 years ago
Sean Bright	8283aa40a0	res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. Fixes #376	2 years ago
Joshua C. Colp	d6fa631d2a	Update issue guidelines link for bug reports.	2 years ago
George Joseph	dfb0efdcda	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	8a27d7ef89	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	3287585379	configs: Fix typo in pjsip.conf.sample.	2 years ago
George Joseph	c32d090e77	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	ddb5c377fd	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	1fc68b1d30	.github: PRSubmitActions: Fix adding reviewers to PR	2 years ago
George Joseph	7b19fdfa3d	.github: Remove start_version from Releaser	2 years ago
George Joseph	10204b9d31	.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	2694792e13	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	b99606955e	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	889b0aa375	func_curl.c: Ensure channel is locked when manipulating datastores.	2 years ago
George Joseph	87e053f6d1	.github: Fix job prereqs in PROpenedUpdated	2 years ago
George Joseph	344eb4f211	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
Mike Bradeen	2291f196c5	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	2 years ago

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