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
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>
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
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
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
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
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
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
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
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
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
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
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
There are a handful of files in the tree that
reference an SVN link for the coding guidelines.
This removes these because the links are dead
and the vast majority of source files do not
contain these links, so this is more consistent.
app_skel still maintains an (up to date) link
to the coding guidelines.
ASTERISK-30159 #close
Change-Id: I35bbb20f66982e98099cff3029ede20091ffdac7
The MeetmeList and MeetmeListRooms AMI
responses are currently completely undocumented.
This adds documentation for these responses.
ASTERISK-30018 #close
Change-Id: Id93135b7edf01de6f8fba266e2122989dc8996b8
Adjusts some logging levels to be more or less important,
that is more prominent when actual problems occur and less
prominent for less noteworthy things.
ASTERISK-30153 #close
Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b
Documents the ConfbridgeListRooms AMI response,
which is currently not documented.
ASTERISK-30020 #close
Change-Id: Id6fff7a936244bae7b52686301eb740c1169cdea
Currently, if multiple video-enabled ConfBridges are
conferenced together, we immediately get into a scenario
where an infinite sequence of video updates fills up
the taskprocessor queue and causes memory consumption
to climb unabated until Asterisk is killed. This is due
to the core bridging mechanism that provides video updates
(softmix_bridge_write_control in bridge_softmix.c)
continously updating all the channels in the bridge with
video updates.
The logic to do so in the core is that the video updates
should be provided if the video_update_discard property
for the bridge is 0, or if enough time has elapsed since
the last video update. Thus, we already have a safeguard
built in to ensure the scenario described above does not
happen. Currently, however, this safeguard is not being
adequately ensured.
In app_confbridge, the video_update_discard property
defaults to 2000, which is a healthy value that should
completely prevent this issue. However, this value is
only set onto the bridge in the SFU video mode. This
leaves video modes such as follow_talker completely
vulnerable, since video_update_discard will actually
be 0, since the default or set value was never applied.
As a result, the core bridging mechanism will always
try to provide video updates regardless of when the last
one was sent.
To prevent this issue from happening, we now always
set the video_update_discard property on the bridge
with the value from the bridge profile. The app_confbridge
defaults will thus ensure that infinite video updates
no longer happen in any video mode.
ASTERISK-29907 #close
Change-Id: I4accb2536ac62797950468e9930f12ef7dd486b2
ASTERISK_28638 caused a regression by incorrectly aborting
early and overwriting the status on certain calls.
This was exhibited by certain technologies such as DAHDI,
where DAHDI returns NULL for the request if a line is busy.
This caused the BUSY condition to be incorrectly treated
as CHANUNAVAIL because the DIALSTATUS was getting incorrectly
overwritten and call handling was aborted early.
This is fixed by instead checking if any valid peers have been
specified, as opposed to checking the list size of successful
requests. This is because the latter could be empty but this
does not indicate any kind of problem. This restores the
previous working behavior.
ASTERISK-29989 #close
Change-Id: I4d4b209b967816b1bc791534593ababa2b99bb88
The Dial application currently stops hook flashes
dead in their tracks from propagating through on
outbound calls. This fixes that so they can go
down the wire.
ASTERISK-30115 #close
Change-Id: Id4e78b29a049f35c5b1e7520eaa10d0eb5b7f97c
Finding an application and executing it if found is
a common task throughout Asterisk. This adds a helper
function around pbx_exec to do this, to eliminate
redundant code and make it easier for modules to
substitute variables and execute applications by name.
ASTERISK-30061 #close
Change-Id: Ifee4d2825df7545fb515d763d393065675140c84
Adds an option to VoiceMailMain that prevents the user
from deleting messages during that application invocation.
This can be useful for public or shared mailboxes, where
some users should be able to listen to messages but not
delete them.
ASTERISK-30063 #close
Change-Id: Icdfb8423ae8d1fce65a056b603eb84a672e80a26
Adds the CONFBRIDGE_CHANNELS function which can be used
to retrieve a comma-separated list of channels, filtered
by a particular type of participant category. This output
can then be used with functions like UNSHIFT, SHIFT, POP,
etc.
ASTERISK-30036 #close
Change-Id: I1950aff932437476dc1abab6f47fb4ac90520b83
Most issues were in stringfields and had to do with comparing
a pointer to an constant/interned string with NULL. Since the
string was a constant, a pointer to it could never be NULL so
the comparison was always "true". gcc now complains about that.
There were also a few issues where determining if there was
enough space for a memcpy or s(n)printf which were fixed
by defining some of the involved variables as "volatile".
There were also a few other miscellaneous fixes.
ASTERISK-30044
Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570
Documents the Dial syntax for DAHDI, namely the channel group,
distinctive ring, answer confirmation, and digital call options
that are specified in the resource itself.
ASTERISK-24827 #close
Change-Id: Ib95e78497fb00dc5cbfde1c93a69f034bfd08c30
Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio
file.
ASTERISK-29931
Added by Michael Cargile
Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2
The admin_exec function in app_meetme is used by the SLA
applications for internal bridging. However, in these cases,
chan is NULL. Currently, this function will set some status
variables that are intended for a channel, but since channel
is NULL, this is erroneously creating meaningless global
variables, which shouldn't be happening. This sets these
variables only if chan is not NULL.
ASTERISK-30002 #close
Change-Id: I817df6c26f5bda131678e56791b0b61ba64fc6f7
Adds version information for applications, functions,
and manager events/actions.
This is not completely exhaustive by any means but
covers most new things added that have release
versioning information in the issue tracker.
ASTERISK-29940 #close
Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131
When executing dial, the topology of the incoming channel is cloned and
used for the outgoing channel. This creates issues when an incoming
stream is sendonly or recvonly as the stream state of the outgoing
channel will be the same as the stream state of the incoming channel.
Now the stream state is flipped for the outgoing stream in
dial_exec_full if the incoming stream topology is recvonly or sendonly.
ASTERISK-29655
Reported by: Michael Auracher
ASTERISK-29638
Reported by: Michael Auracher
Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1
The ReceiveMF and ReceiveSF applications currently always
return 0, even if a channel has hung up. The call will still
end but generally applications are expected to return -1 if
the channel has hung up.
We now return -1 if a hangup occured to bring this behavior
in line with this norm. This has no functional impact, but
merely increases conformity with how these modules interact
with the PBX core.
ASTERISK-29951 #close
Change-Id: I234d755050ab8ed58f197c6925b968ba26b14033
Adds the m option to the Queue application, which allows a
music on hold class to be specified at runtime which will
override the class configured in queues.conf.
This option functions like the m option to Dial.
ASTERISK-29876 #close
Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7
Currently, if a user tries to access a non-dynamic
MeetMe conference and the conference is not found,
the call simply silent hangs up. There is no indication
to the user that anything went wrong at all.
This changes the relevant debug message to a warning
so that the user is notified of this invalidity.
ASTERISK-29954 #close
Change-Id: Iebcfae3755d00f2150d676ee211c57bc59530048
Adds documentation for all of the possible return values
for the DIALSTATUS variable in the Dial application.
ASTERISK-25716
Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0
This adds a new AMI action called QueueWithdrawCaller.
This AMI action makes it possible to withdraw a caller from a queue,
in a safe and a generic manner.
This can be useful for retrieving a specific call and
dispatching it to a specific extension.
It works by signaling the caller to exit the queue application
whenever it can. Therefore, it is not guaranteed
that the call will leave the queue.
ASTERISK-29909 #close
Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec
Adds the since tag to the documentation DTD so
that individual applications, functions, etc.
can now specify when they were added to Asterisk.
This tag is added at the individual application,
function, etc. level as opposed to at the module
level because modules can expand over time as new
functionality is added, and granularity only
to the module level would generally not be useful.
This enables the ability to more easily determine
when new functionality was added to Asterisk, down
to minor version as opposed to just by major version.
This makes it easier for users to write more portable
dialplan if desired to not use functionality that may
not be widely available yet.
ASTERISK-29896 #close
Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5
Currently, if VoiceMailMain is called with a mailbox, if that
mailbox doesn't exist, then the application silently falls back
to prompting the user for the mailbox, as if no arguments were
provided.
However, if a specific mailbox is requested and it doesn't exist,
then no warning at all is emitted.
This fixes this behavior to now warn if a specifically
requested mailbox could not be accessed, before falling back to
prompting the user for the correct mailbox.
ASTERISK-29920 #close
Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4
mpg123 doesn't support HTTPS, but the MP3Player application
doesn't document this or warn the user about this. HTTPS
streams have become more common nowadays and users could
reasonably try to play them without being aware they should
use the HTTP stream instead.
This adds documentation to note this limitation. It also
throws a warning if users try to use the HTTPS stream to
tell them to use the HTTP stream instead.
ASTERISK-29900 #close
Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827
Adds an option to the ReceiveMF application to allow specifying a
maximum number of digits.
Originally, this capability was not added to ReceiveMF as it was
with ReceiveSF because typically a ST digit is used to denote that
sending of digits is complete. However, there are certain signaling
protocols which simply transmit a digit (such as Expanded In-Band
Signaling) and for these, it's necessary to be able to read a
certain number of digits, as opposed to until receiving a ST digit.
This capability is added as an option, as opposed to as a parameter,
to remain compatible with existing usage (and not shift the
parameters).
ASTERISK-29877 #close
Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b
There are a lot of Queue AMI actions and Queue applications
which do not load queue and queue members from Realtime.
AMI actions
QueuePause - if queue not in memory - response "Interface not found".
QueueStatus/QueueSummary - if queue not in memory - empty response.
Applications:
PauseQueueMember - if queue not in memory
Attempt to pause interface %s, not found
UnpauseQueueMember - if queue not in memory
Attempt to unpause interface xxxxx, not found
This patch adds a new function load_realtime_queues
which loads queue and queue members for desired queue
or all queues and all members if param 'queuename' is NULL or empty.
Calls the function load_realtime_queues when needed.
Also this patch fixes leak of ast_config in function set_member_value.
Also this patch fixes incorrect LOG_WARNING when pausing/unpausing
already paused/unpaused member.
The function ast_update_realtime returns 0 when no record modified.
So 0 is not an error to warn about.
ASTERISK-29873 #close
ASTERISK-18416 #close
ASTERISK-27597 #close
Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c
if holdtime is (0 min, 0 sec) there is no hold time announcements
we should then also not playing queue-thankyou
ASTERISK-29831
Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c
gethostbyname() and gethostbyname_r() are deprecated in favor of
getaddrinfo() which we use in the ast_sockaddr family of functions.
ASTERISK-29819 #close
Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7
adding support for playing the correct en/et for nordic languages
by adding 'n' for neuter gender in the relevant ast_say_number
ASTERISK-29827
Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1
Currently, the MP3Player application doesn't
emit a warning if attempting to play a stream
which no longer exists. This can be a common
scenario as many mp3 streams are valid at some
point but can disappear at any time.
Now a warning is thrown if attempting to play
a nonexistent MP3 stream, instead of silently
exiting.
ASTERISK-29829 #close
Change-Id: I53a0bf1ed1740166655eb66fe7675f6f808bf535
Adds missing documentation for some channel,
bridge, and queue events.
ASTERISK-24427
ASTERISK-29515
Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64
Adds tech-agnostic support for SF signaling
by adding SF sender and receiver applications
as well as Dial integration.
ASTERISK-29802 #close
Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad
A previous patch for ASTERISK_29578 caused a 'leak' of
extension state information across queues, causing the
state of the first member of unrelated queues to be
updated in addition to the correct member. Which queues
and members depended on the order of queues in the
iterator.
Additionally, it is possible to use the same 'hint:' on
multiple queue members, so the update cannot break out
of the update loop early when a match is found.
ASTERISK-29806 #close
Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad
added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial
ASTERISK-29795
Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6
added support for playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code
ASTERISK-29797
Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2
Adds a ReceiveText application that can be used in
conjunction with SendText. Currently, there is no
way in Asterisk to receive text in the dialplan
(or anywhere else, really). This allows for Asterisk
to be the recipient of text instead of just the sender.
ASTERISK-29759 #close
Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d
Adds tech-agnostic support for MF signaling by adding
MF sender and receiver applications as well as Dial
integration.
ASTERISK-29496-mf #do-not-close
Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4
changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING
ASTERISK-28053
Reported by: roadkill
Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89
Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.
This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.
ASTERISK-29777 #close
Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b
Refactors generic functions used for email generation
into utils.c so that they can be used by multiple
modules, including app_voicemail and app_minivm,
to avoid code duplication.
ASTERISK-29715 #close
Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee
Fixes a deadlock in app_morsecode caused by locking
the channel twice when reading variables from the
channel. The duplicate lock is simply removed.
ASTERISK-29744 #close
Change-Id: I204000701f123361d7f85e0498fedc90243c75e4
Currently, when the t option is specified with no arguments,
the # character is still treated as a terminator, even though
no character should be treated as a terminator.
This is because a previous regression fix was modified to
remove the use of NULL as a default altogether. However,
NULL and an empty string actually refer to different
arrangements and should be treated differently. NULL is the
default terminator (#), while an empty string removes the
terminator altogether. This is the behavior being used by
the rest of the core.
Additionally, since S_OR catches empty strings as well as
NULL (not intended), this is changed to a ternary operator
instead, which fixes the behavior.
ASTERISK-29705 #close
Change-Id: I9b6b72196dd04f5b1e0ab5aa1b0adf627725e086
If users are able to press # for options while leaving
a message and then press 3 to rerecord the message, if
the caller hangs up during the rerecord prompt but before
Asterisk starts recording a message, then an "empty"
voicemail gets processed whereby an email gets sent out
notifying the user of a 0:00 duration message. The file
doesn't actually exist, so playback will fail since there
was no message to begin with.
This adds a check after the streaming of the rerecord
announcement to see if the caller has hung up. If so,
we bail out early so that we can clean up properly.
ASTERISK-29391 #close
Change-Id: Id965d72759a2fd3b39afb76fec08aaebebe75c31
Add a time_t logintime to storage a time when a member is added into a
queue.
Also, includes show this time (in seconds) using a 'queue show' command
and the field LoginTime for response for AMI events.
ASTERISK-18069 #close
Change-Id: Ied6c3a300f78d78eebedeb3e16a1520fc3fff190
I am adding a mix option that will play by filename and say.conf unlike
say option that will only play with say.conf. It
will look on the format of the name, if it is like say it play with
say.conf if not it will play the file name.
ASTERISK-29662
Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a
If the terminator character is not explicitly specified
and an indications tone is used for reading a digit,
there is no null pointer check so Asterisk crashes.
This prevents null usage from occuring.
ASTERISK-29673 #close
Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac
Previously, if custom hints were used with the hint:
format in app_queue, when device state changes occured,
app_queue would only do a literal string comparison of
the context used for the hint in app_queue and the context
of the hint which just changed state. This caused hints
to not update and become stale if the context associated
with the agent included the context which actually changes
state, essentially completely breaking device state for
any such agents defined in this manner.
This fix adds an additional check to ensure that included
contexts are also compared against the context which changed
state, so that the behavior is correct no matter whether the
context is specified to app_queue directly or indirectly.
ASTERISK-29578 #close
Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78
Adds the ability for users to log to custom log levels
by providing custom log level names in logger.conf. Also
adds a logger show levels CLI command.
ASTERISK-29529
Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702
In new mpg123 versions (since 1.26) the default output is 32 bits
Asterisk expects the output in 16 bits, so we force the output to be on 16 bits.
It will work wit new and old versions of mpg123.
Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key!
ASTERISK-29635 #close
Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816
Adds a SendMF application and PlayMF manager
event to send arbitrary R1 MF tones on the
current or specified channel.
ASTERISK-29496
Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4
Previously, the error emitted when app_stack tries
to branch to a dialplan location that doesn't exist
has included only the information about the attempted
branch in the error log. This adds the current location
as well so users can see where the branch failed in
the logs.
ASTERISK-29626
Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce
There is an option to silence voicemail instructions but it does not
take into consideration if a recorded greeting exists or not. Add a
new 'S' option that does that.
ASTERISK-29632 #close
Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4
There are 3 separate changes here but they are all closely related:
* Only try to set matchfield attributes on 'field' nodes
* We need to adjust how we treat the category pointer based on the
value of the category_match, to avoid memory corruption. We now
generate a regex-like string when match types other than
ACO_WHITELIST and ACO_BLACKLIST are used.
* Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
ACO_BLACKLIST_EXACT since we only have one category we need to
ignore, not two.
ASTERISK-29614 #close
Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e
Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.
ASTERISK-18454 #close
Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b
Prevents reloads of app_queue from also resetting
queue statistics.
Also preserves individual queue agent statistics
if we're just reloading members.
ASTERISK-28701
Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1
The Milliwatt application uses incorrect tone timings
that cause it to play the 1004 Hz tone constantly.
This adds an option to enable the correct timing
behavior, so that the Milliwatt application can
be used for milliwatt test lines. The default behavior
remains unchanged for compatability reasons, even
though it is incorrect.
ASTERISK-29575 #close
Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c
Previously, the Morsecode application only supported international
Morse code. This adds support for American Morse code and adds an
option to configure the frequency used in off intervals.
Additionally, the application checks for hangup between tones
to prevent application execution from continuing after hangup.
ASTERISK-29541
Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4
A list of codecs to use for dialplan-originated calls can
now be specified in Originate, similar to the ability
in call files and the manager action.
Additionally, we now default to just using the slin codec
for originated calls, rather than all the slin* codecs up
through slin192, which has been known to cause issues
and inconsistencies from AMI and call file behavior.
ASTERISK-29543
Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883
Mike Bradeen
Jaco Kroon
Naveen Albert
Sean Bright
George Joseph
Michael Cargile
Maximilian Fridrich
Kfir Itzhak
Alexei Gradinari
Mark Petersen
Sean Bright
Steve Davies
Alexander Traud
Josh Soref
Rodrigo Ramírez Norambuena
Shloime Rosenblum
Carlos Oliva
Joshua C. Colp