5268 Commits (ba30b87e9172d1ab265178cdeda0010d8fa4fe69)

Author	SHA1	Message	Date
Naveen Albert	c70587e43e	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 (cherry picked from commit 485ccb7429)	11 months ago
Naveen Albert	6c840886a9	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 (cherry picked from commit 62b6ea3202)	11 months ago
Sean Bright	7973b7474a	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 (cherry picked from commit c80ce750bb)	11 months ago
Naveen Albert	ec990d6a6e	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 (cherry picked from commit 44a6349b69)	11 months ago
George Joseph	3b73137f15	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. (cherry picked from commit 29d6a62768)	11 months ago
Tinet-mucw	8d3bd2cc31	app_chanspy.c: resolving the issue writing frame to whisper audiohook. ... ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook. Resolves: #876 (cherry picked from commit 03983b2ff9)	1 year ago
Jean-Denis Girard	beee92c909	app_voicemail: Fix sql insert mismatch caused by cherry-pick ... When commit e8c9cb80 was cherry-picked in from master, the fact that the 20 and 18 branches still had the old "macrocontext" column wasn't taken into account so the number of named parameters didn't match the number of '?' placeholders. They do now. We also now use ast_asprintf to create the full mailbox query SQL statement instead of trying to calculate the proper length ourselves. Resolves: #831 (cherry picked from commit 83b3783499)	1 year ago
Tinet-mucw	463178f0d5	app_chanspy.c: resolving the issue with audiohook direction read ... ChanSpy(${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 (cherry picked from commit 4263f7b58d)	1 year ago
George Joseph	c9d5fd450e	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 2b9dc329bd)	1 year ago
Alexei Gradinari	1108f61bfb	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 624c23310e)	1 year ago
Sean Bright	e49049b474	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 574d6986ce)	1 year ago
George Joseph	ca6590426a	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 9c2cc5bf24)	1 year ago
Naveen Albert	74bf9821a3	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 17d6a08beb)	2 years ago
Naveen Albert	aacf779b32	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 564d22acc6)	2 years ago
Shaaah	742440029c	app_queue.c : fix "queue add member" usage string ... Fixing bracket placement in the "queue add member" cli usage string. (cherry picked from commit 157dff87b3)	2 years ago
Naveen Albert	11d7b9d224	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 166fc3c3d0)	2 years ago
George Joseph	d9e4b080f2	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 57d30c17b3)	2 years ago
cmaj	f89a394658	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 d2de6a2158)	2 years ago
Mike Bradeen	86b3ed84a5	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 f31b981d92)	2 years ago
Naveen Albert	c1a495002f	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 aa5b9d1d9f)	2 years ago
Sean Bright	295ef9d052	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 9dd9615edf)	2 years ago
Naveen Albert	4f4d997aa1	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 f4edb3b87e)	2 years ago
Naveen Albert	7af4934943	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 8cc5fc2e39)	2 years ago
Maximilian Fridrich	94de48c80c	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 8d7665db7b)	2 years ago
Sean Bright	5288c25dc1	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 242a0685b1)	2 years ago
Sean Bright	a8015816b1	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 6b4d7f2125)	2 years ago
Matthew Fredrickson	d8434ca398	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 eeae61b2f8)	2 years ago
Naveen Albert	1f200b936d	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 72c6e95ed1)	2 years ago
Naveen Albert	af91eb9ea0	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 70435a886c)	2 years ago
Sean Bright	9c1eaa01e0	app_queue.c: Emit unpause reason with PauseQueueMember event. ... Fixes #395 (cherry picked from commit 6c3ad4ceaf)	2 years ago
Naveen Albert	2d6143068e	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 c4d9e950bd)	2 years ago
Jaco Kroon	b90f11fc70	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 60b529ea00)	2 years ago
Naveen Albert	fd98f7c893	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 1ada3c0747)	2 years ago
Mike Bradeen	3e2cb7ecb1	app_voicemail: Fix for loop declarations ... Resolve for loop initial declarations added in cli changes. Resolves: #275 (cherry picked from commit 62480fd91b)	2 years ago
zhengsh	e879a5534a	app_audiosocket: Fixed timeout with -1 to avoid busy loop. ... Resolves: asterisk#234 (cherry picked from commit a2353f4803)	2 years ago
Matthew Fredrickson	9cec96a04c	app_macro: Fix locking around datastore access ... app_macro sometimes would crash due to datastore list corruption on the channel because of lack of locking around find and create process for the macro datastore. This patch locks the channel lock prior to protect against this problem. Resolves: #265 (cherry picked from commit ec4e0340ce)	2 years ago
Matthew Fredrickson	e15c994186	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 3f80f9ce0b)	2 years ago
Joshua C. Colp	248e724928	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. (cherry picked from commit 71558e23d0)	2 years ago
Mike Bradeen	a7093ba51f	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. (cherry picked from commit 74f62dbfe9)	2 years ago
Mike Bradeen	b05e0fd05c	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> (cherry picked from commit 6aff1c1065)	2 years ago
Mike Bradeen	0337c76b4e	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 (cherry picked from commit 5f01983abd)	2 years ago
Olaf Titz	f33d5c2f1e	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 (cherry picked from commit 3b23ee4203)	2 years ago
Nathan Bruning	90fbc26bce	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 (cherry picked from commit c88a8aa8b0)	2 years ago
alex2grad	cf2f3a1516	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 (cherry picked from commit 591351d4ad)	2 years ago
Niklas Larsson	5fe7fd5577	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. (cherry picked from commit 12fb5d8589)	2 years ago
Joshua Colp	6c15249f1d	Revert "app_queue: periodic announcement configurable start time." ... This reverts commit c405630810. Reason for revert: Causes segmentation fault. Change-Id: Ib0c8c592f8d4f0a5e3889aeadfe8bdcde800ba42 (cherry picked from commit a8ce41b7f7)	2 years ago
Naveen Albert	0e02fd15cb	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 (cherry picked from commit 1ed147ffc1)	2 years ago
Mike Bradeen	03da795e3e	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 (cherry picked from commit b2e9419961)	2 years ago
Jaco Kroon	2383e8a598	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> (cherry picked from commit c405630810)	2 years ago
Naveen Albert	6d5998ae5c	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 (cherry picked from commit 48e3da5cdf)	2 years ago