5273 Commits (0939d0779c7931dc905b10a89db7ef9667fb22ed)

Author	SHA1	Message	Date
George Joseph	e16d498ce7	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. (cherry picked from commit 1b3a73cb24)	10 months ago
Alexei Gradinari	8a09a6ca6d	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. (cherry picked from commit 192a848311)	10 months ago
Sean Bright	5f07b095f2	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 (cherry picked from commit 6914c93791)	12 months ago
George Joseph	800f11649b	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. (cherry picked from commit 9e2179baa1)	12 months ago
Naveen Albert	6e5a6c176d	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. (cherry picked from commit bdf9327a33)	1 year ago
Naveen Albert	ae2fa8c5f0	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 (cherry picked from commit 2ac9c8fb5c)	1 year ago
Shaaah	d43d250e14	app_queue.c : fix "queue add member" usage string ... Fixing bracket placement in the "queue add member" cli usage string. (cherry picked from commit 6f99268f79)	1 year ago
Naveen Albert	6af036a855	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'. (cherry picked from commit 190b6eafb3)	1 year ago
George Joseph	f72d97b7b1	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 (cherry picked from commit a5ae546b88)	1 year ago
cmaj	1377ac9e89	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. (cherry picked from commit c863e0d77d)	1 year ago
Mike Bradeen	b510d681a1	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. (cherry picked from commit 69fe814813)	1 year ago
Naveen Albert	5e0f1bb5d2	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 (cherry picked from commit ed39406838)	1 year ago
Sean Bright	dbcd737302	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 (cherry picked from commit 03ad690276)	1 year ago
Mike Bradeen	ab1a9fa7d1	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. (cherry picked from commit a22db8fd60)	1 year ago
Naveen Albert	c148203225	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 (cherry picked from commit 1bf4493371)	1 year ago
Naveen Albert	f485d3cc8b	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 (cherry picked from commit 3bb34477d4)	1 year ago
Maximilian Fridrich	b3cff31e1a	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. (cherry picked from commit 366dc1e99f)	1 year ago
Sean Bright	77e8011291	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 (cherry picked from commit fbe92dce2b)	1 year ago
Sean Bright	0620c14eb6	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`. (cherry picked from commit 33213c1979)	1 year ago
Matthew Fredrickson	8c71aefa04	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 (cherry picked from commit b5c31b55c9)	1 year ago
Naveen Albert	0f33423107	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 (cherry picked from commit 4a356e984c)	1 year ago
Naveen Albert	8ad6e6e585	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 (cherry picked from commit 95bc661542)	1 year ago
Sean Bright	4feaf8a880	app_queue.c: Emit unpause reason with PauseQueueMember event. ... Fixes #395 (cherry picked from commit baf3ce25f5)	1 year ago
Naveen Albert	a11885989c	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 (cherry picked from commit 75620616f4)	1 year ago
Jaco Kroon	9eeee41fc5	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> (cherry picked from commit 130c3ab792)	1 year ago
Naveen Albert	3b7a0d617f	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 (cherry picked from commit c032d38d3f)	2 years ago
Mike Bradeen	5f7c8e75fd	app_voicemail: Fix for loop declarations ... Resolve for loop initial declarations added in cli changes. Resolves: #275 (cherry picked from commit e9b269692f)	2 years ago
zhengsh	d15a5c547f	app_audiosocket: Fixed timeout with -1 to avoid busy loop. ... Resolves: asterisk#234 (cherry picked from commit eab40e755e)	2 years ago
Matthew Fredrickson	64f76bdc60	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 (cherry picked from commit 1d7ae8d227)	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
Olaf Titz	128a9969a1	app_voicemail_imap: Fix message count when IMAP server is unavailable ... Some callers of __messagecount did not correctly handle error return, instead returning a -1 message count. This caused a notification with "Messages-Waiting: yes" and "Voice-Message: -1/0 (0/0)" if the IMAP server was unavailable. Fixes: #64	2 years ago
Nathan Bruning	292834d1ba	app_queue: Add force_longest_waiting_caller option. ... This adds an option 'force_longest_waiting_caller' which changes the global behavior of the queue engine to prevent queue callers from 'jumping ahead' when an agent is in multiple queues. Resolves: #108 Also closes old asterisk issues: - ASTERISK-17732 - ASTERISK-17570 Change-Id: I0f84e27903fefbe2018d0afa2d67b23aa0b321ce	2 years ago
alex2grad	8a6379f36b	app_followme: fix issue with enable_callee_prompt=no (#88) ... * app_followme: fix issue with enable_callee_prompt=no If the FollowMe option 'enable_callee_prompt' is set to 'no' then Asterisk incorrectly sets a winner channel to the channel from which any control frame was read. This fix sets the winner channel only to the answered channel. Resolves: #87 ASTERISK-30326	2 years ago
Niklas Larsson	c1f21b6f66	app_queue: Preserve reason for realtime queues ... When Asterisk is restarted it does not preserve paused reason for members of realtime queues. This was fixed for non-realtime queues in ASTERISK_25732 Resolves: #66 UpgradeNote: Add a new column to the queue_member table: reason_paused VARCHAR(80) so the reason can be preserved. UserNote: Make paused reason in realtime queues persist an Asterisk restart. This was fixed for non-realtime queues in ASTERISK_25732.	2 years ago
Naveen Albert	5dac935f61	app_sla: Migrate SLA applications out of app_meetme. ... This removes the dependency of the SLAStation and SLATrunk applications on app_meetme, in anticipation of the imminent removal of the deprecated app_meetme module. The user interface for the SLA applications is exactly the same, and in theory, users should not notice a difference. However, the SLA applications now use ConfBridge under the hood, rather than MeetMe, and they are now contained within their own module. Resolves: #50 ASTERISK-30309 UpgradeNote: The SLAStation and SLATrunk applications have been moved from app_meetme to app_sla. If you are using these applications and have autoload=no, you will need to explicitly load this module in modules.conf.	2 years ago
Joshua Colp	4ec4543332	Revert "app_queue: periodic announcement configurable start time." ... This reverts commit 71e317f68f. Reason for revert: Causes segmentation fault. Change-Id: I3beeda83249bffec2a8f246aa50a6b2f1b59ef59	2 years ago
Naveen Albert	fa41458298	app_queue: Fix minor xmldoc duplication and vagueness. ... The F option in the xmldocs for the Queue application was erroneously duplicated, causing it to display twice on the wiki. The two sections are now merged into one. Additionally, the description for the d option was quite vague. Some more details are added to provide context as to what this actually does. ASTERISK-30486 #close Change-Id: I6706cea708b5cc781f59f8652c2cb377e55aed7e	2 years ago
Mike Bradeen	c4a55322bc	res_mixmonitor: MixMonitorMute by MixMonitor ID ... While it is possible to create multiple mixmonitor instances on a channel, it was not previously possible to mute individual instances. This change includes the ability to specify the MixMonitorID when calling the manager action: MixMonitorMute. This will allow an individual MixMonitor instance to be muted via id. This id can be stored as a channel variable using the 'i' MixMonitor option. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor spy-type audiohooks on the channel. This is done via the new audiohook function: ast_audiohook_set_mute_all. ASTERISK-30464 Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98	2 years ago
Jaco Kroon	71e317f68f	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. ASTERISK-30437 #close Change-Id: Ia79984b6377ef78f167ad9ea2ac084bec29955d0 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Naveen Albert	95a41d231f	app_dial: Fix DTMF not relayed to caller on unanswered calls. ... DTMF frames are not handled in app_dial when sent towards the caller. This means that if DTMF is sent to the calling party and the call has not yet been answered, the DTMF is not audible. This is now fixed by relaying DTMF frames if only a single destination is being dialed. ASTERISK-29516 #close Change-Id: Iafd7430ac2915126d42dc48f0b73b262452ee027	2 years ago
Naveen Albert	cec0c5cd39	app_senddtmf: Add SendFlash AMI action. ... Adds an AMI action to send a flash event on a channel. ASTERISK-30440 #close Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9	2 years ago
Mike Bradeen	18a9540520	app_read: Add an option to return terminator on empty digits. ... Adds 'e' option to allow Read() to return the terminator as the dialed digits in the case where only the terminator is entered. ie; if "#" is entered, return "#" if the 'e' option is set and "" if it is not. ASTERISK-30411 Change-Id: I49f3221824330a193a20c660f99da0f1fc2cbbc5	2 years ago
Mike Bradeen	152239b978	app_directory: Add a 'skip call' option. ... Adds 's' option to skip calling the extension and instead set the extension as DIRECTORY_EXTEN channel variable. ASTERISK-30405 Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299	2 years ago
Mike Bradeen	cee3564b45	app_senddtmf: Add option to answer target channel. ... Adds a new option to SendDTMF() which will answer the specified channel if it is not already up. If no channel is specified, the current channel will be answered instead. ASTERISK-30422 Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085	2 years ago
Sean Bright	f9a27f028f	app_queue: Minor docs and logging fixes for UnpauseQueueMember. ... ASTERISK-30417 #close Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe	2 years ago
Sean Bright	632da7c11a	app_queue: Reset all queue defaults before reload. ... Several queue fields were not being set to their default value during a reload. Additionally added some sample configuration options that were missing from queues.conf.sample. Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4	2 years ago
Sean Bright	c85fc1278f	doxygen: Fix doxygen errors. ... Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	2 years ago