asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Fredrickson	eeae61b2f8	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	1f66a9fc28	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	318710cf6e	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	7b297cb90d	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	bee37b7404	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	59e296df01	resource_channels.c: Explicit codec request when creating UnicastRTP. Fixes #394	2 years ago
Sean Bright	092bd618fb	doc: Update IP Quality of Service links. Fixes #328	2 years ago
George Joseph	9a93ce0409	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	744bd4f9ac	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	beb9689288	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	ad46f59ca8	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	3c9e7ad4ba	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	d1f8978d91	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	9c70df1bb7	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	72c6e95ed1	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	63c5b119f4	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	f32a1fa64e	Remove files that are no longer updated Fixes: #360	2 years ago
Naveen Albert	70435a886c	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	6c3ad4ceaf	app_queue.c: Emit unpause reason with PauseQueueMember event. Fixes #395	2 years ago
George Joseph	11a586b65e	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	8397077e75	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	6aa80d1bb8	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	2aa6a63188	res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. Fixes #386	2 years ago
Sean Bright	47578d385a	res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. Fixes #376	2 years ago
George Joseph	848628b795	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	0f06787469	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	12017263c8	configs: Fix typo in pjsip.conf.sample.	2 years ago
George Joseph	3e4024ee20	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	96420f3d48	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	7524ef6b20	.github: PRSubmitActions: Fix adding reviewers to PR	2 years ago
George Joseph	0eda94525b	.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	de391bd8ab	.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	27283a9209	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	1ff540c75f	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	028ebdcc10	func_curl.c: Ensure channel is locked when manipulating datastores.	2 years ago
George Joseph	9e44d0e2d2	.github: Fix job prereqs in PROpenedUpdated	2 years ago
George Joseph	3b244926d6	.github: Block PR tests until approved	2 years ago
George Joseph	f74c84e978	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	761b143db3	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
Mike Bradeen	4d5b479244	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
Naveen Albert	4f99db350a	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	2 years ago
Bastian Triller	6edeb90485	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.	2 years ago
Mike Bradeen	792ad9fec8	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	2 years ago
Naveen Albert	c4d9e950bd	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	2 years ago
Eduardo	ed7fe7b02a	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	2 years ago
George Joseph	04df168656	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.	2 years ago
George Joseph	309ea22d8d	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	2 years ago
Tinet-mucw	3e3c2c833d	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	2 years ago
Mike Bradeen	ff4b5ed951	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	2 years ago
Naveen Albert	cd90c5a82b	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	2 years ago

1 2 3 4 5 ...

33734 Commits (eeae61b2f827bd3b32820344d66ba4a85e3ae3cc) All Branches Search

33734 Commits (eeae61b2f827bd3b32820344d66ba4a85e3ae3cc)

All Branches