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)
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)
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)
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)
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)
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)
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)
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)
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)
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)
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)
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)
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)
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)
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)
* 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)
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)
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)
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)
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)
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)
Adds an AMI action to send a flash event
on a channel.
ASTERISK-30440 #close
Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9
(cherry picked from commit 0c8b42ede3)
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
Adds 's' option to skip calling the extension and instead set the
extension as DIRECTORY_EXTEN channel variable.
ASTERISK-30405
Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299
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
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
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
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
In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b394) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.
Partial fix for ASTERISK~25661.
Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff
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
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
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
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
MixMonitor currently uses the Connected Line as the Caller ID
for voicemails. This is due to the implementation being written
this way for use with Digium phones. However, in general this
is not correct for generic usage in the dialplan, and people
may need the real Caller ID instead. This adds an option to do that.
ASTERISK-30286 #close
Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c
SLAStation currently autoservices the station channel before
creating a thread to actually dial the trunk. This leads
to duplicate servicing of the channel which causes assertions,
deadlocks, crashes, and moreover not the correct behavior.
Removing the autoservice prevents the crash, but if the station
hangs up before the trunk answers, the call hangs since the hangup
was never serviced on the channel.
This is fixed by not autoservicing the channel, but instead
servicing it in the thread dialing the trunk, since it is doing
so synchronously to begin with. Instead of sleeping for 100ms
in a loop, we simply use the channel for timing, and abort
if it disappears.
The same issue also occurs with SLATrunk when a call is answered,
because ast_answer invokes ast_waitfor_nandfds. Thus, we use
ast_raw_answer instead which does not cause any conflict and allows
the call to be answered normally without thread blocking issues.
ASTERISK-29998 #close
Change-Id: Icc237d50354b5910000d2305901e86d2c87bb9d8
Adds an option that allows MixMonitor to delete
its copy of any recording files before exiting.
This can be handy in conjunction with options
like m, which copy the file elsewhere, and the
original files may no longer be needed.
ASTERISK-30284 #close
Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e
When gosub is executed on channels without a PBX, the context,
extension, and priority are initialized to the channel driver's
default location for that endpoint. As a result, the last Return
will restore this location and the Gosub logs will print out bogus
information about our exit point.
To fix this, on channels that don't have a PBX, the execution
location is left intact on the last return if there are no
further stack frames left. This allows the correct location
to be printed out to the user, rather than the bogus default
context.
ASTERISK-30076 #close
Change-Id: I1d42a99c9aa9e3708d32718863175158a894e414
(cherry picked from commit 12d18b0a40)
Adds the n option to not answer the channel when calling
BridgeWait, so the application can be used without
forcing answer supervision.
ASTERISK-30216 #close
Change-Id: I6b85ef300b1f7b5170f8537e2b10889cc2e6605a
Adds an option that will play an audio file
to the party while AMD is running on the
channel, so the called party does not just
hear silence.
ASTERISK-30179 #close
Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6
Adds the end_marked_any option, which can be used
to kick a user from a conference if any marked user
leaves.
ASTERISK-30211 #close
Change-Id: I9e8da7ccb892e522546c0f2b5476d172e022c2f5
Fixes a few coding guideline violations:
* Use of C99 comments
* Opening brace on same line as function prototype
ASTERISK-30163 #close
Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7
If the CONFBRIDGE function is used to dynamically set
menu options, a memory leak occurs when a menu option
that has been set is overridden, since the menu entry
is not destroyed before being freed. This ensures that
it is.
Additionally, logic that duplicates the destroy function
is removed in lieu of the destroy function itself.
ASTERISK-28422 #close
Change-Id: I71cfb5c24e636984d41086d1333a416dc12ff995
The MeetmeList and MeetmeListRooms AMI
responses are currently completely undocumented.
This adds documentation for these responses.
ASTERISK-30018 #close
Change-Id: Id93135b7edf01de6f8fba266e2122989dc8996b8
