33812 Commits (3c1a2daecd8221e5ed50c70011a931f795ed209c)

 

All Branches   

Search

Author	SHA1	Message	Date
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
George Joseph	c8a97d5f8c	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.	2 years ago
Naveen Albert	201d554b4e	chan_iax2: Improve authentication debugging. ... Improves and adds some logging to make it easier for users to debug authentication issues. Resolves: #286	2 years ago
Vitezslav Novy	8cf1db15c2	res_rtp_asterisk: fix wrong counter management in ioqueue objects ... In function rtp_ioqueue_thread_remove counter in ioqueue object is not decreased which prevents unused ICE TURN threads from being removed. Resolves: #301	2 years ago
George Joseph	55eca816b1	make_buildopts_h, et. al. Allow adding all cflags to buildopts.h ... The previous behavior of make_buildopts_h was to not add the non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS, REF_DEBUG, etc. to the buildopts.h file because "it caused ccache to invalidate files and extended compile times". They're only defined by passing them on the gcc command line with '-D' options. In practice, including them in the include file rarely causes any impact because the only time ccache cares is if you actually change an option so the hit occurrs only once after you change it. OK so why would we want to include them? Many IDEs follow the include files to resolve defines and if the options aren't in an include file, it can cause the IDE to mark blocks of "ifdeffed" code as unused when they're really not. So... * Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H which tells make_buildopts_h to include the non-ABI-breaking flags in buildopts.h as well as the ABI-breaking ones. The default is disabled to preserve current behavior. As before though, only the ABI-breaking flags appear in AST_BUILDOPTS and only those are used to calculate AST_BUILDOPT_SUM. A new AST_BUILDOPT_ALL define was created to capture all of the flags. * make_version_c was streamlined to use buildopts.h and also to create asterisk_build_opts_all[] and ast_get_build_opts_all(void) * "core show settings" now shows both AST_BUILDOPTS and AST_BUILDOPTS_ALL. UserNote: The "Build Options" entry in the "core show settings" CLI command has been renamed to "ABI related Build Options" and a new entry named "All Build Options" has been added that shows both breaking and non-breaking options.	2 years ago
Mike Bradeen	13da50219e	func_periodic_hook: Add hangup step to avoid timeout ... func_periodic_hook does not hangup after playback, relying on hangup which keeps the channel alive longer than necessary. Resolves: #325	2 years ago
Sean Bright	8be6f7b78c	res_stasis_recording.c: Save recording state when unmuted. ... Fixes #322	2 years ago
Mike Bradeen	15829bcef1	res_speech_aeap: check for null format on response ... * Fixed issue in res_speech_aeap when unable to provide an input format to check against.	2 years ago
George Joseph	209015a2ff	func_periodic_hook: Don't truncate channel name ... func_periodic_hook was truncating long channel names which causes issues when you need to run other dialplan functions/apps on the channel. Resolves: #319	2 years ago
George Joseph	c5a93ea93f	safe_asterisk: Change directory permissions to 755 ... If the safe_asterisk script detects that the /var/lib/asterisk directory doesn't exist, it now creates it with 755 permissions instead of 770. safe_asterisk needing to create that directory should be extremely rare though because it's normally created by 'make install' which already sets the permissions to 755. Resolves: #316	2 years ago
Maximilian Fridrich	0cd336a518	chan_rtp: Implement RTP glue for UnicastRTP channels ... Resolves: #298 UserNote: The dial string option 'g' was added to the UnicastRTP channel which enables RTP glue and therefore native RTP bridges with those channels.	2 years ago
Jaco Kroon	60b529ea00	app_queue: periodic announcement configurable start time. ... This newly introduced periodic-announce-startdelay makes it possible to configure the initial start delay of the first periodic announcement after which periodic-announce-frequency takes over. UserNote: Introduce a new queue configuration option called 'periodic-announce-startdelay' which will vary the normal (historic) behavior of starting the periodic announcement cycle at periodic-announce-frequency seconds after entering the queue to start the periodic announcement cycle at period-announce-startdelay seconds after joining the queue. The default behavior if this config option is not set remains unchanged. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Joshua C. Colp	039bbddd4f	variables: Add additional variable dialplan functions. ... Using the Set dialplan application does not actually delete channel or global variables. Instead the variables are set to an empty value. This change adds two dialplan functions, GLOBAL_DELETE and DELETE which can be used to delete global and channel variables instead of just setting them to empty. There is also no ability within the dialplan to determine if a global or channel variable has actually been set or not. This change also adds two dialplan functions, GLOBAL_EXISTS and VARIABLE_EXISTS which can be used to determine if a global or channel variable has been set or not. Resolves: #289 UserNote: Four new dialplan functions have been added. GLOBAL_DELETE and DELETE have been added which allows the deletion of global and channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been added which checks whether a global or channel variable has been set.	2 years ago
George Joseph	c14849ac17	ari-stubs: Fix more local anchor references ... Also allow CreateDocs job to be run manually with default branches.	2 years ago
George Joseph	0771c94e23	ari-stubs: Fix more local anchor references ... Also allow CreateDocs job to be run manually with default branches.	2 years ago
George Joseph	540c8b2794	ari-stubs: Fix broken documentation anchors ... All of the links that reference page anchors with capital letters in the ids (#Something) have been changed to lower case to match the anchors that are generated by mkdocs.	2 years ago
Bastian Triller	98665a14b3	res_pjsip_session: Send Session Interval too small response ... Handle session interval lower than endpoint's configured minimum timer when sending first answer. Timer setting is checked during this step and needs to handled appropriately. Before this change, no response was sent at all. After this change a response with 422 Session Interval too small is sent to UAC.	2 years ago
George Joseph	29967e1eea	.github: Update workflow-application-token-action to v2	2 years ago
Naveen Albert	1ada3c0747	app_dial: Fix infinite loop when sending digits. ... If the called party hangs up while digits are being sent, -1 is returned to indicate so, but app_dial was not checking the return value, resulting in the hangup being lost and looping forever until the caller manually hangs up the channel. We now abort if digit sending fails. ASTERISK-29428 #close Resolves: #281	2 years ago
Mike Bradeen	62480fd91b	app_voicemail: Fix for loop declarations ... Resolve for loop initial declarations added in cli changes. Resolves: #275	2 years ago
George Joseph	39b8b76f7c	alembic: Fix quoting of the 100rel column ... Add quoting around the ps_endpoints 100rel column in the ALTER statements. Although alembic doesn't complain when generating sql statements, postgresql does (rightly so). Resolves: #274	2 years ago
Naveen Albert	1241410bc3	pbx.c: Fix gcc 12 compiler warning. ... Resolves: #277	2 years ago
zhengsh	a2353f4803	app_audiosocket: Fixed timeout with -1 to avoid busy loop. ... Resolves: asterisk#234	2 years ago