5290 Commits (a935133eebd58d7725c3519c87c9f46d1e9d6dec)

Author	SHA1	Message	Date
Jeremy Lainé	16bfde8b9b	docs: Fix minor typo in MixMonitor AMI action ... The `Options` argument was erroneously documented as lowercase `options`.	8 months ago
Sean Bright	b4156fecf0	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	8 months ago
George Joseph	85a4ab8390	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	9 months ago
George Joseph	a47b8e2d40	docs: Add version information to manager event instance XML elements ... * Do a git blame on the embedded XML managerEvent elements. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The events defined in res/res_pjsip/pjsip_manager.xml were done by hand because the XML was extracted from the C source file relatively recently. Two bugs were fixed along the way... * The get_documentation awk script was exiting after it processed the first DOCUMENTATION block it found in a file. We have at least 1 source file with multiple DOCUMENTATION blocks so only the first one in them was being processed. The awk script was changed to continue searching rather than exiting after the first block. * Fixing the awk script revealed an issue in logger.c where the third DOCUMENTATION block contained a XML fragment that consisted only of a managerEventInstance element that wasn't wrapped in a managerEvent element. Since logger_doc.xml already existed, the remaining fragments in logger.c were moved to it and properly organized.	9 months ago
George Joseph	a22dc33057	docs: Add version information to configObject and configOption XML elements ... Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `<since>/<version>` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep <summary>` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains <commit>` to find the tags that contain each commit. * Weed out all tags not <major>.<minor>.0. * Sort and discard any <major>.0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation.	9 months ago
Sean Bright	7f13966202	manager: Add `<since>` tags for all AMI actions.	9 months ago
Sperl Viktor	392558ee59	app_queue: indicate the paused state of a dynamically added member in queue_log. ... Fixes: #1021	10 months ago
Sperl Viktor	a80ae57cac	app_queue: allow dynamically adding a queue member in paused state. ... Fixes: #1007 UserNote: use the p option of AddQueueMember() for paused member state. Optionally, use the r(reason) option to specify a custom reason for the pause.	10 months ago
Ben Ford	8273eefd87	app_mixmonitor: Add 'D' option for dual-channel audio. ... Adds the 'D' option to app_mixmonitor that interleaves the input and output frames of the channel being recorded in the monitor output frame. This allows for two streams in the recording: the transmitted audio and the received audio. The 't' and 'r' options are compatible with this. Fixes: #945 UserNote: The MixMonitor application now has a new 'D' option which interleaves the recorded audio in the output frames. This allows for stereo recording output with one channel being the transmitted audio and the other being the received audio. The 't' and 't' options are compatible with this.	11 months ago
Naveen Albert	6a07e8e0f1	app_dial: Fix progress timeout calculation with no answer timeout. ... If to_answer is -1, simply comparing to see if the progress timeout is smaller than the answer timeout to prefer it will fail. Add an additional check that chooses the progress timeout if there is no answer timeout (or as before, if the progress timeout is smaller). Resolves: #821	12 months ago
Naveen Albert	97dfe4cd40	app_dial: Fix progress timeout. ... Under some circumstances, the progress timeout feature added in commit 320c98eec8 does not work as expected, such as if there is no media flowing. Adjust the waitfor call to explicitly use the progress timeout if it would be reached sooner than the answer timeout to ensure we handle the timers properly. Resolves: #821	1 year ago
Sean Bright	243f20a78d	res_agi.c: Ensure SIGCHLD handler functions are properly balanced. ... Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be balanced to ensure that we can capture the exit status of child processes when we need to. This extends to functions that call `ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as `ast_safe_fork()` and `ast_safe_fork_cleanup()`. The primary change here is ensuring that we do not call `ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously called `ast_safe_fork()`. Additionally we reinforce some of the documentation and add an assertion to, ideally, catch this sooner were this to happen again. Fixes #922	1 year ago
Naveen Albert	9423710a5e	app_voicemail: Fix ill-formatted pager emails with custom subject. ... Add missing end-of-headers newline to pager emails with custom subjects, since this was missing from this code path. Resolves: #902	1 year ago
George Joseph	48acf00eab	Fix application references to Background ... The app is actually named "BackGround" but several references in XML documentation were spelled "Background" with the lower case "g". This was causing documentation links to return "not found" messages.	1 year ago
Tinet-mucw	a721f99eb0	app_chanspy.c: resolving the issue writing frame to whisper audiohook. ... ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook. Resolves: #876	1 year ago
George Joseph	d6157aff5b	app_voicemail: Use ast_asprintf to create mailbox SQL query ... ...instead of trying to calculate the length of the buffer needed manually.	1 year ago
Tinet-mucw	e98127d540	app_chanspy.c: resolving the issue with audiohook direction read ... ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail. Resolves: #861	1 year ago
George Joseph	1b3a73cb24	app_voicemail_odbc: Allow audio to be kept on disk ... This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail to leave the message and greeting audio files on disk and only store the message metadata in the database. This option came from a concern that the database could grow to large and cause remote access and/or replication to become slow. In a clustering situation with this option, all asterisk instances would share the same database for the metadata and either use a shared filesystem or other filesystem replication service much more suitable for synchronizing files. The changes to app_voicemail to implement this feature were actually quite small but due to the complexity of the module, the actual source code changes were greater. They fall into the following categories: * Tracing. The module is so complex that it was impossible to figure out the path taken for various scenarios without the addition of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in code that's not related to the functional change. Making this worse was the fact that many "if" statements in this module didn't use braces. Since the tracing macros add multiple statements, many "if" statements had to be converted to use braces. * Excessive use of PATH_MAX. Previous maintainers of this module used PATH_MAX to allocate character arrays for filesystem paths and SQL statements as though they cost nothing. In fact, PATH_MAX is defined as 4096 bytes! Some functions had (and still have) multiples of these. One function has 7. Given that the vast majority of installations use the default spool directory path `/var/spool/asterisk/voicemail`, the actual path length is usually less than 80 bytes. That's over 4000 bytes wasted. It was the same for SQL statement buffers. A 4K buffer for statement that only needed 60 bytes. All of these PATH_MAX allocations in the ODBC related code were changed to dynamically allocated buffers. The rest will have to be addressed separately. * Bug fixes. During the development of this feature, several pre-existing ODBC related bugs were discovered and fixed. They had to do with leaving orphaned files on disk, not preserving original message ids when moving messages between folders, not honoring the "formats" config parameter in certain circumstances, etc. UserNote: This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail_odbc to leave the message and greeting audio files on disk and only store the message metadata in the database. Much more information can be found in the voicemail.conf.sample file.	1 year ago
Alexei Gradinari	192a848311	app_queue: Add option to not log Restricted Caller ID to queue_log ... Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event in the queue log if the Caller ID is restricted. Resolves: #765 UpgradeNote: Add a new column to the queues table: queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes') to control whether the Restricted Caller ID will be stored in the queue log. UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID will be stored in the queue log. If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.	1 year ago
Sean Bright	6914c93791	app_queue.c: Properly handle invalid strategies from realtime. ... The existing code sets the queue strategy to `ringall` but it is then immediately overwritten with an invalid one. Fixes #707	1 year ago
George Joseph	9e2179baa1	Fix incorrect application and function documentation references ... There were a few references in the embedded documentation XML where the case didn't match or where the referenced app or function simply didn't exist any more. These were causing 404 responses in docs.asterisk.org.	2 years ago
Naveen Albert	bdf9327a33	app_dial: Add dial time for progress/ringing. ... Add a timeout option to control the amount of time to wait if no early media is received before giving up. This allows aborting early if the destination is not being responsive. Resolves: #588 UserNote: The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received.	2 years ago
Naveen Albert	2ac9c8fb5c	app_voicemail: Properly reinitialize config after unit tests. ... Most app_voicemail unit tests were not properly cleaning up after themselves after running. This led to test mailboxes lingering around in the system. It also meant that if any unit tests in app_voicemail that create mailboxes were executed and the module was not unloaded/loaded again prior to running the test_voicemail_vm_info unit test, Asterisk would segfault due to an attempt to copy a NULL string. The load_config test did actually have logic to reinitialize the config after the test. However, this did not work in practice since load_config() would not reload the config since voicemail.conf had not changed during the test; thus, additional logic has been added to ensure that voicemail.conf is truly reloaded, after any unit tests which modify the users list. This prevents the SEGV due to invalid mailboxes lingering around, and also ensures that the system state is restored to what it was prior to the tests running. Resolves: #629	2 years ago
Shaaah	6f99268f79	app_queue.c : fix "queue add member" usage string ... Fixing bracket placement in the "queue add member" cli usage string.	2 years ago
Naveen Albert	190b6eafb3	app_voicemail: Allow preventing mark messages as urgent. ... This adds an option to allow preventing callers from leaving messages marked as 'urgent'. Resolves: #619 UserNote: The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.	2 years ago
George Joseph	a5ae546b88	Reduce startup/shutdown verbose logging ... When started with a verbose level of 3, asterisk can emit over 1500 verbose message that serve no real purpose other than to fill up logs. When asterisk shuts down, it emits another 1100 that are of even less use. Since the testsuite runs asterisk with a verbose level of 3, and asterisk starts and stops for every one of the 700+ tests, the number of log messages is staggering. Besides taking up resources, it also makes it hard to debug failing tests. This commit changes the log level for those verbose messages to 5 instead of 3 which reduces the number of log messages to only a handful. Of course, NOTICE, WARNING and ERROR message are unaffected. There's also one other minor change... ast_context_remove_extension_callerid2() logs a DEBUG message instead of an ERROR if the extension you're deleting doesn't exist. The pjsip_config_wizard calls that function to clean up the config and has been triggering that annoying error message for years. Resolves: #582	2 years ago
cmaj	c863e0d77d	app_speech_utils.c: Allow partial speech results. ... Adds 'p' option to SpeechBackground() application. With this option, when the app timeout is reached, whatever the backend speech engine collected will be returned as if it were the final, full result. (This works for engines that make partial results.) Resolves: #572 UserNote: The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs.	2 years ago
Mike Bradeen	69fe814813	app_chanspy: Add 'D' option for dual-channel audio ... Adds the 'D' option to app chanspy that causes the input and output frames of the spied channel to be interleaved in the spy output frame. This allows the input and output of the spied channel to be decoded separately by the receiver. If the 'o' option is also set, the 'D' option is ignored as the audio being spied is inherently one direction. Fixes: #569 UserNote: The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored.	2 years ago
Naveen Albert	ed39406838	app_if: Fix next priority calculation. ... Commit fa3922a4d2 fixed a branching issue but "overshoots" when calculating the next priority. This fixes that; accompanying test suite tests have also been extended. Resolves: #560	2 years ago
Sean Bright	03ad690276	app_confbridge: Don't emit warnings on valid configurations. ... The numeric bridge profile options `internal_sample_rate` and `maximum_sample_rate` are documented to accept the special values `auto` and `none`, respectively. While these values currently work, they also emit warnings when used which could be confusing for users. In passing, also ensure that we only accept the documented range of sample rate values between 8000 and 192000. Fixes #546	2 years ago
Mike Bradeen	a22db8fd60	app_voicemail_odbc: remove macrocontext from voicemail_messages table ... When app_macro was deprecated, the macrocontext column was removed from the INSERT statement but the binds were not renumbered. This broke the insert. This change removes the macrocontext column via alembic and re-numbers the existing columns in the INSERT. Fixes: #527 UserNote: The fix requires removing the macrocontext column from the voicemail_messages table in the voicemail database via alembic upgrade. UpgradeNote: The fix requires that the voicemail database be upgraded via alembic. Upgrading to the latest voicemail database via alembic will remove the macrocontext column from the voicemail_messages table.	2 years ago
Naveen Albert	1bf4493371	app_if: Fix faulty EndIf branching. ... This fixes faulty branching logic for the EndIf application. Instead of computing the next priority, which should be done for false conditionals or ExitIf, we should simply advance to the next priority. Resolves: #341	2 years ago
Naveen Albert	3bb34477d4	general: Fix broken links. ... This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in f28047db36. Resolves: #430	2 years ago
Maximilian Fridrich	366dc1e99f	app_dial: Add option "j" to preserve initial stream topology of caller ... Resolves: #462 UserNote: The option "j" is now available for the Dial application which uses the initial stream topology of the caller to create the outgoing channels.	2 years ago
Sean Bright	fbe92dce2b	app_voicemail.c: Completely resequence mailbox folders. ... Resequencing is a process that occurs when we open a voicemail folder and discover that there are gaps between messages (e.g. `msg0000.txt` is missing but `msg0001.txt` exists). Resequencing involves shifting the existing messages down so we end up with a sequential list of messages. Currently, this process stops after reaching a threshold based on the message limit (`maxmsg`) configured on the current folder. However, if `maxmsg` is lowered when a voicemail folder contains more than `maxmsg + 10` messages, resequencing will not run completely leaving the mailbox in an inconsistent state. We now resequence up to the maximum number of messages permitted by `app_voicemail` (currently hard-coded at 9999 messages). Fixes #86	2 years ago
Sean Bright	33213c1979	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
Matthew Fredrickson	b5c31b55c9	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	4a356e984c	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	95bc661542	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	baf3ce25f5	app_queue.c: Emit unpause reason with PauseQueueMember event. ... Fixes #395	2 years ago
Naveen Albert	75620616f4	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
Jaco Kroon	130c3ab792	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
Naveen Albert	c032d38d3f	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	e9b269692f	app_voicemail: Fix for loop declarations ... Resolve for loop initial declarations added in cli changes. Resolves: #275	2 years ago
zhengsh	eab40e755e	app_audiosocket: Fixed timeout with -1 to avoid busy loop. ... Resolves: asterisk#234	2 years ago
Matthew Fredrickson	1d7ae8d227	Revert "app_stack: Print proper exit location for PBXless channels." ... This reverts commit 617dad4cba. apps/app_stack.c: Revert buggy gosub patch This seems to break the case when a predial macro calls a gosub. When the gosub calls return, the Return function outputs: app_stack.c:423 return_exec: Return without Gosub: stack is empty This returns -1 to the calling macro, which returns to app_dial and causes the call to hangup instead of proceeding with the macro that invoked the gosub. Resolves: #253	2 years ago
Joshua C. Colp	6b6880072b	app_queue: Add support for applying caller priority change immediately. ... The app_queue module provides both an AMI action and a CLI command to change the priority of a caller in a queue. Up to now this change of priority has only been reflected to new callers into the queue. This change adds an "immediate" option to both the AMI action and CLI command which immediately applies the priority change respective to the other callers already in the queue. This can allow, for example, a caller to be placed at the head of the queue immediately if their priority is sufficient. Resolves: #202 UserNote: The 'queue priority caller' CLI command and 'QueueChangePriorityCaller' AMI action now have an 'immediate' argument which allows the caller priority change to be reflected immediately, causing the position of a caller to move within the queue depending on the priorities of the other callers.	2 years ago
Mike Bradeen	eef5a0b7bf	Adds manager actions to allow move/remove/forward individual messages ... in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's. Resolves: #181 UserNote: The following manager actions have been added VoicemailBoxSummary - Generate message list for a given mailbox VoicemailRemove - Remove a message from a mailbox folder VoicemailMove - Move a message from one folder to another within a mailbox VoicemailForward - Copy a message from one folder in one mailbox to another folder in another or the same mailbox.	2 years ago
Mike Bradeen	48c6e3fb1d	app_voicemail: add CLI commands for message manipulation ... Adds CLI commands to allow move/remove/forward individual messages from a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a show mailbox, required to retrieve message ID's. Resolves: #170 UserNote: The following CLI commands have been added to app_voicemail voicemail show mailbox <mailbox> <context> Show contents of mailbox <mailbox>@<context> voicemail remove <mailbox> <context> <from_folder> <messageid> Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context> voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder> Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder> voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder> Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to mailbox <mailbox>@<context> <to_folder>	2 years ago
Mike Bradeen	250b3a2f3c	app_voicemail: fix imap compilation errors ... Fixes two compilation errors in app_voicemail_imap, one due to an unsed variable and one due to a new variable added in the incorrect location in _163. Resolves: #174	2 years ago