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1.4.27.1
1.4.28
1.4.28-rc1
1.4.29
1.4.29-rc1
1.4.29.1
1.4.3
1.4.30
1.4.30-rc1
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1.4.30-rc3
1.4.31
1.4.31-rc1
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1.4.33
1.4.33-rc1
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1.4.33.1
1.4.34
1.4.34-rc1
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1.4.35
1.4.35-rc1
1.4.36
1.4.36-rc1
1.4.37
1.4.37-rc1
1.4.37.1
1.4.38
1.4.38-rc1
1.4.38.1
1.4.39
1.4.39-rc1
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1.4.39.2
1.4.4
1.4.40
1.4.40-rc1
1.4.40-rc2
1.4.40-rc3
1.4.40.1
1.4.40.2
1.4.41
1.4.41-rc1
1.4.41.1
1.4.41.2
1.4.42
1.4.42-rc1
1.4.42-rc2
1.4.43
1.4.44
1.4.5
1.4.6
1.4.7
1.4.7.1
1.4.8
1.4.9
1.6.0
1.6.0-beta1
1.6.0-beta2
1.6.0-beta3
1.6.0-beta4
1.6.0-beta5
1.6.0-beta6
1.6.0-beta7
1.6.0-beta7.1
1.6.0-beta8
1.6.0-beta9
1.6.0-rc1
1.6.0-rc2
1.6.0-rc3
1.6.0-rc4
1.6.0-rc5
1.6.0-rc6
1.6.0.1
1.6.0.10
1.6.0.11-rc1
1.6.0.11-rc2
1.6.0.12
1.6.0.13
1.6.0.13-rc1
1.6.0.14
1.6.0.14-rc1
1.6.0.15
1.6.0.16
1.6.0.16-rc1
1.6.0.16-rc2
1.6.0.17
1.6.0.18
1.6.0.18-rc1
1.6.0.18-rc2
1.6.0.18-rc3
1.6.0.19
1.6.0.2
1.6.0.20
1.6.0.20-rc1
1.6.0.21
1.6.0.21-rc1
1.6.0.22
1.6.0.23
1.6.0.23-rc1
1.6.0.23-rc2
1.6.0.24
1.6.0.25
1.6.0.26
1.6.0.26-rc1
1.6.0.27
1.6.0.27-rc1
1.6.0.27-rc2
1.6.0.27-rc3
1.6.0.28
1.6.0.28-rc1
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1.6.0.3
1.6.0.3-rc1
1.6.0.3.1
1.6.0.4-rc1
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1.6.0.5
1.6.0.6
1.6.0.6-rc1
1.6.0.7
1.6.0.7-rc1
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1.6.0.8
1.6.0.9
1.6.1-beta1
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1.6.1.0-rc2
1.6.1.0-rc3
1.6.1.0-rc4
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1.6.1.1
1.6.1.10
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1.6.1.12
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1.6.1.15-rc1
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1.8.0-rc4
1.8.0-rc5
1.8.1
1.8.1-rc1
1.8.1.1
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1.8.10.0-rc1
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1.8.10.0-rc3
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1.8.10.1
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1.8.7.0-rc1
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1.8.7.1
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1.8.8.0
1.8.8.0-rc1
1.8.8.0-rc2
1.8.8.0-rc3
1.8.8.0-rc4
1.8.8.0-rc5
1.8.8.1
1.8.8.2
1.8.9.0
1.8.9.0-rc1
1.8.9.0-rc2
1.8.9.0-rc3
1.8.9.1
1.8.9.2
1.8.9.3
10.0.0
10.0.0-beta1
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10.0.1
10.1.0
10.1.0-rc1
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10.1.1
10.1.2
10.1.3
10.10.0
10.10.0-digiumphones
10.10.0-digiumphones-rc1
10.10.0-digiumphones-rc2
10.10.0-rc1
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10.10.1
10.10.1-digiumphones
10.11.0
10.11.0-digiumphones
10.11.0-digiumphones-rc1
10.11.0-digiumphones-rc2
10.11.0-digiumphones-rc3
10.11.0-rc1
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10.11.1
10.11.1-digiumphones
10.12.0
10.12.0-digiumphones
10.12.0-digiumphones-rc1
10.12.0-digiumphones-rc2
10.12.0-rc1
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10.12.1
10.12.1-digiumphones
10.12.2
10.12.2-digiumphones
10.12.3
10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
10.2.0-rc1
10.2.0-rc2
10.2.0-rc3
10.2.0-rc4
10.2.1
10.3.0
10.3.0-rc1
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10.3.0-rc3
10.3.1
10.4.0
10.4.0-digiumphones-rc1
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10.4.0-rc1
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10.4.0-rc3
10.4.1
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10.5.0
10.5.0-digiumphones
10.5.0-digiumphones-rc1
10.5.0-digiumphones-rc2
10.5.0-rc1
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10.5.1
10.5.1-digiumphones
10.5.2
10.5.2-digiumphones
10.6.0
10.6.0-digiumphones
10.6.0-digiumphones-rc1
10.6.0-digiumphones-rc2
10.6.0-rc1
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10.6.1
10.6.1-digiumphones
10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
10.7.0-rc1
10.7.1
10.7.1-digiumphones
10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
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10.8.0-rc1
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10.9.0
10.9.0-digiumphones
10.9.0-digiumphones-rc1
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10.9.0-digiumphones-rc3
10.9.0-rc1
10.9.0-rc2
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11.0.0
11.0.0-beta1
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11.0.0-rc1
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11.0.1
11.0.2
11.1.0
11.1.0-rc1
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11.1.0-rc3
11.1.1
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11.10.0
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11.11.0
11.11.0-rc1
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11.13.1
11.14.0
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11.14.1
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11.15.0
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11.16.0
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11.21.0
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${ noResults }
5259 Commits (5216a1133d3e734d7c88c84a5a6254f2f0ffd9be)
Author | SHA1 | Message | Date |
---|---|---|---|
|
58b16a538d |
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 |
1 year ago |
|
d1fb397cfc |
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
|
1 year ago |
|
3d7a7b1a47 |
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. |
1 year ago |
|
3026ac08ab |
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 |
1 year ago |
|
ca931c9436 |
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`. |
1 year ago |
|
21412fddcb |
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 |
1 year ago |
|
4657163c56 |
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 |
|
d678370b54 |
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 |
|
6c6137028b |
app_queue.c: Emit unpause reason with PauseQueueMember event.
Fixes #395 |
2 years ago |
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d60c3c36e7 |
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 |
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4db98a38f1 |
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 |
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d60cec6249 |
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 |
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fce6821106 |
app_voicemail: Fix for loop declarations
Resolve for loop initial declarations added in cli changes. Resolves: #275 |
2 years ago |
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f4aaa4b9fb |
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Resolves: asterisk#234 |
2 years ago |
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27c5d27f01 |
Revert "app_stack: Print proper exit location for PBXless channels."
This reverts commit
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2 years ago |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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8a6379f36b
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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 |
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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 |
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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 |
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4ec4543332 |
Revert "app_queue: periodic announcement configurable start time."
This reverts commit
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2 years ago |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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f9a27f028f |
app_queue: Minor docs and logging fixes for UnpauseQueueMember.
ASTERISK-30417 #close Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe |
2 years ago |
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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 |
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c85fc1278f |
doxygen: Fix doxygen errors.
Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85 |
2 years ago |
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78f5a41d64 |
app_signal: Add signaling applications
Adds the Signal and WaitForSignal applications, which can be used for inter-channel signaling in the dialplan. Signal supports sending a signal to other channels listening for a signal of the same name, with an optional data payload. The signal is received by all channels waiting for that named signal. ASTERISK-29810 #close Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3 |
2 years ago |
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2a066b6c2f |
app_directory: add ability to specify configuration file
Adds option to app_directory to specify a filename from which to read configuration instead of voicemail.conf ie; same => n,Directory(,,c(directory.conf)) This configuration should contain a list of extensions using the voicemail.conf format, ie; 2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no ASTERISK-30404 Change-Id: Id58ccb1344ad1e563fa10db12f172fbd104a9d13 |
2 years ago |
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6b03d60c7d |
res_monitor: Remove deprecated module.
ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50 |
2 years ago |
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cbaba132a7 |
app_playback.c: Fix PLAYBACKSTATUS regression.
In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit
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2 years ago |
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e8f548c155 |
app_macro: Remove deprecated module.
For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6 |
2 years ago |
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f7726430b2 |
app_broadcast: Add Broadcast application
Adds a new application, Broadcast, which can be used for one-to-many transmission and many-to-one reception of channel audio in Asterisk. This is similar to ChanSpy, except it is designed for multiple channel targets instead of a single one. This can make certain kinds of audio manipulation more efficient and streamlined. New kinds of audio injection impossible with ChanSpy are also made possible. ASTERISK-30180 #close Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726 |
2 years ago |
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a46d5f9b76 |
app_cdr: Remove deprecated application and option.
This removes the deprecated NoCDR application, which was deprecated in Asterisk 12, having long been fully superseded by the CDR_PROP function. The deprecated e option to ResetCDR is also removed for the same reason. ASTERISK-30371 #close Change-Id: Id9ed094d8e4baf98bcbc610035c2295bfafe9ec0 |
2 years ago |
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4095a382da |
chan_sip: Remove deprecated module.
ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996 |
2 years ago |
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4168fa3466 |
app_voicemail_odbc: Fix string overflow warning.
Fixes a negative offset warning by initializing the buffer to empty. Additionally, although it doesn't currently complain about it, the size of a buffer is increased to accomodate the maximum size contents it could have. ASTERISK-30240 #close Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877 |
2 years ago |
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d60bd09851 |
app_sendtext: Remove references to removed applications.
Removes see-also references to applications that don't exist anymore (removed in Asterisk 19), so these dead links don't show up on the wiki. ASTERISK-30347 #close Change-Id: I9539bc30f57cd65aa4e2d5ce8185eafa09567909 |
2 years ago |
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a28421a676 |
app_if: Fix format truncation errors.
Fixes format truncation warnings in gcc 12.2.1. ASTERISK-30349 #close Change-Id: I42be4edf0284358b906e765d1966b6b9d66e1d3c |
2 years ago |
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e3ea1b88ff |
app_if: Adds conditional branch applications
Adds the If, ElseIf, Else, ExitIf, and EndIf applications for conditional execution of a block of dialplan, similar to the While, EndWhile, and ExitWhile applications. The appropriate branch is executed at most once if available and may be broken out of while inside. ASTERISK-29497 Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49 |
2 years ago |
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9e14523ca3 |
app_voicemail: Fix missing email in msg_create_from_file.
msg_create_from_file currently does not dispatch emails, which means that applications using this function, such as MixMonitor, will not trigger notifications to users (only AMI events are sent our currently). This is inconsistent with other ways users can receive voicemail. This is fixed by adding an option that attempts to send an email and falling back to just the notifications as done now if that fails. The existing behavior remains the default. ASTERISK-30283 #close Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7 |
2 years ago |
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d0140fc7fe |
app_osplookup: Remove deprecated module.
ASTERISK-30302 Change-Id: I2268189646fa0b587675d8619322818143172474 |
2 years ago |