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
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
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
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
Naveen Albert
George Joseph
Michael Cargile
Maximilian Fridrich
Kfir Itzhak
Alexei Gradinari
Mark Petersen
Sean Bright
Steve Davies