This commit fixes two possible scenarios:

* When recording name and if during recording you hangup, file is never
removed. This is due to the fact file location is nulled.
* When recording name and if you hangup during thank-you prompt, file
is never removed.

ASTERISK-27123 #close

Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625

When performing the "Queues" action via AMI, it outputs the same
text that the Asterisk CLI outputs when running a "queue show"
command, which does not conform with the AMI spec. "QueueStatus"
already does what the "Queues" action should do, so instead of
correcting the output, the "Queues" action will be removed and
"QueueStatus" should be used instead.

ASTERISK-27073 #close
Reported by: Brian

Change-Id: Id11743859758255b69cc3a557750d7a56c6d16f8

This API was not actively maintained, was not added to new modules
(such as res_pjsip), and there exist better alternatives to acquire the
same information, such as the ARI.

Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83

The primary focus of this patch is adding a missing call to
ast_odbc_release_obj(), but is also a general cleanup of the ODBC
related code in app_voicemail.

ASTERISK-27093 #close

Change-Id: I8e285142eaeb3146b4287a928276b70db76c902b

Fixed the following bugs:

* calls to stream_echo_write had the last two parameters swapped
* ast_read should have been ast_read_stream
* added a null check on the frame's subclass format

This also resets the update_sent flag upon receiving SRRCHANGE control frame.
This will then force a video update.

ASTERISK-26997

Change-Id: I6ad7c8253559b800800433c52339e7f5aa583566

Add priority to callers in AMI QueueStatus response

ASTERISK-27092 #close

Change-Id: I8d1f737a72c7c38f4cfe1a4ee3ecc0a4f85bd199

The fix for ASTERISK-25665 introduced a regression.
The return value of queue_exec used to be 0 in case of leavewhenempty
but it was changed to -1 (returned from wait_our_turn and passed
transparently by queue_exec), thus leading to hangup instead of returning
back to dialplan.

This commit resets the value back to 0 in this case, restoring
original behavior.

ASTERISK-27065 #close
Reported by: Marek Cervenka

Change-Id: Id9c83b75aeda463250155e88c5004be52bbca5ac

A new global option "imap_poll_logout" was added to specify whether need to
disconnect from the IMAP server after polling of mailboxes.

ASTERISK-27068 #close

Closing IMAP connection after loading mailbox from voicemail.conf

ASTERISK-24052 #close

Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a

Closing IMAP connection on module reload or unload.

ASTERISK-24052 #close

Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525

Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1

Closing IMAP connection on MWI unsubscribe.

ASTERISK-24052 #close

Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd

ASTERISK-27025

Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed

A previous commit added plumbing to bridge_softmix to allow for an SFU
experience with Asterisk. This commit adds an option to app_confbridge
that allows for a confbridge to actually make use of the SFU video mode.

SFU mode is implemented in a "set it and forget it" kind of way. That
is, when the bridge is created, if SFU mode is enabled, then the video
mode gets set to SFU and cannot be changed. Future improvements may
allow for a hybrid experience (e.g. forward multiple video streams,
specifically those of the most recent talkers), but for this addition,
no such capability is present.

Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020

This sets up the "plumbing" in bridge_softmix to
be able to accommodate Asterisk asking as an SFU
(selective forwarding unit) for conferences.

The way this works is that whenever a channel enters or leaves a
conference, all participants in the bridge get sent a stream topology
change request. The topologies consist of the channels' original
topology, along with video destination streams corresponding to each
participants' source video streams. So for instance, if Alice, Bob, and
Carol are in the conference, and each supplies one video stream, then
the topologies for each would look like so:

Alice:
Audio,
Source video(Alice),
Destination Video(Bob),
Destination video (Carol)

Bob:
Audio,
Source video(Bob)
Destination Video(Alice),
Destination video (Carol)

Carol:
Audio,
Source video(Carol)
Destination Video(Alice),
Destination video (Bob)

This way, video that arrives from a source video stream can then be
copied out to the destination video streams on the other participants'
channels.

Once the bridge gets told that a topology on a channel has changed, the
bridge constructs a map in order to get the video frames routed to the
proper destination streams. This is done using the bridge channel's
stream_map.

This change is bare-bones with regards to SFU support. Some key features
are missing at this point:

* Stream limits. This commit makes no effort to limit the number of
  streams on a specific channel. This means that if there were 50 video
  callers in a conference, bridge_softmix will happily send out topology
  change requests to every channel in the bridge, requesting 50+
  streams.

* Configuration. The plumbing has been added to bridge_softmix, but
  there has been nothing added as of yet to app_confbridge to enable SFU
  video mode.

* Testing. Some functions included here have unit tests.
  However, the functionality as a whole has only been verified by
  hand-tracing the code.

* Selectivenss. For a "selective" forwarding unit, this does not
  currently have any means of being selective.

* Features. Presumably, someone might wish to only receive video from
  specific sources. There are no external-facing functions at the moment
  that allow for users to select who they receive video from.

* Efficiency. The current scheme treats all video streams as being
  unidirectional. We could be re-using a source video stream as a
  desetnation, too. But to simplify things on this first round, I did it
  this way.

Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d

A change was done which added an 'in_call' flag to queue
members that was set to true while talking to an agent.
Unfortunately in practice this does not accurately reflect
whether they are talking to an agent or not. If a Local
channel is involved and a transfer is performed then the
app_queue application would incorrectly think the agent
was still in a call with the caller. This was done to
fix a race condition between an agent becoming available
by device state and the checking of the last call information
for the wrapup time. There was a small window where the
last call information would be the previous value instead
of the new one.

This change goes about fixing the original issue in a
different way by considering the call completed if device
state is received which would make the agent available
and if they are currently in a call. If this occurs the
last call information is updated before the agent becomes
available ensuring that old information is not present
when checking if the member should be called. This also
improves the transfer situation by actually updating
and enforcing the wrapup time.

ASTERISK-26399
ASTERISK-26400
ASTERISK-26715
ASTERISK-26975

Change-Id: Ife1cb686e3173b3a6d368601adef9aff69d4beea

When user leaves a conference, its channel calls async_play_sound_file()
in order to play the name announcement and then unlinks the sound file.
The async_play_sound_file() function adds a task to conference playback queue,
which then runs playback_common() function in a different thread.

It leads to a race condition when, in some cases, channel thread may unlink
the sound file before playback_common() had a chance to open it.

This patch creates a file deletion task, that is queued after playback.

ASTERISK-27012 #close

Change-Id: I412f7922d412004b80917d4e892546c15bd70dd3

Additional variable to work alongside QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY,
including an extra parameter in queuerules.conf. This value causes lower
Agent penalty values to "raise up" so that they can join higher penalty agents
and be treated equally after a period of time.

ASTERISK-26995 #close

Change-Id: If1c6421a983667a5ac4c359f6dac25b212b4c459

If the channel does not have multi-stream support then this application acts
just like app_echo. If it does have multi-stream support then each stream is
echoed back to itself (one-to-one).

If a "num" is specified, then a new topology is made that contains clones (from
the channel's topology) of all media types that are not equal to the given
"type". If the media type differs then the first stream matching the "type" is
cloned into the new topology and then up to "num" - 1 of the same stream are
also cloned into it. Any additional streams from the original topology matching
the "type" are subsequently ignored (i.e. not added to the new topology).

For this same case when a frame is read from a stream that frame is still
echoed back like before, but now that frame is also echoed out to the
additional streams that matched on the specified "type".

ASTERISK-26997 #close

Change-Id: I254144486734178e196c7f590a26ffc13543ff2c

When manipulating flags on a channel the channel has to be
locked to guarantee that nothing else is also manipulating
the flags. This change introduces locking where necessary to
guarantee this. It also adds helper functions that manipulate
channel flags and lock to reduce repeated code.

ASTERISK-26789

Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10

There are 2 places in app_queue.c that log EXITEMPTY event: one in
wait_our_turn, and another one in queue_exec in the loop trying to
call an agent after wait_our_turn.

In most cases it leads to logging EXITEMPTY twice.

ABANDON is also logged on two places, and in the rare case when an agent
and caller hang up simultaneously it's also possible to get duplicates
in queue_log.

This commit changes wait_our_turn to return -1 ("the caller should exit
the queue") instead of 0 ("the caller's turn has arrived") in case of
leaving when empty, so queue_exec skips the agent calling loop.

Also, leave_queue is now executed only once in this case, because 2nd
time is just a noop when the queue entry has already been removed.

Also, it sets qe->handled to -1 to indicate that the call was not
answered by an agent, but the necessary handling has already been done
in order to avoid logging an extra ABANDON entry.

ASTERISK-25665 #close
Reported by: Ove Aursand

Change-Id: I4578dd383bf2ac41589cf167865e8aaebcd4c11e

menu_template_handler wasn't properly accounting for the fact that
it might be called both during a load/reload (which isn't really
valid but not prevented) and by a dialplan function.  In both cases
it was attempting to use the "pending" config which wasn't valid in
the latter case.  aco_process_config is also partly to blame because
it wasn't properly cleaning "pending" up when a reload was done and
no changes were made.  Both of these contributed to a crash if
CONFBRIDGE(menu,template) was called in a dialplan after a reload.

* aco_process_config now sets info->internal->pending to NULL
  after it unrefs it although this isn't strictly necessary in the
  context of this fix.
* menu_template_handler now uses the "current" config and silently
  ignores any attempt to be called as a result of someone uses the
  "template" parameter in the conf file.

Luckily there's no other place in the codebase where
aco_pending_config is used outside of aco_process_config.

ASTERISK-25506 #close
Reported-by: Frederic LE FOLL

Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7

This change extends the ast_request functionality by adding another
function and callback to create an outgoing channel with a requested
stream topology. Fallback is provided by either converting the
requested stream topology into a format capabilities structure if
the channel driver does not support streams or by converting the
requested format capabilities into a stream topology if the channel
driver does support streams.

The Dial application has also been updated to request an outgoing
channel with the stream topology of the calling channel.

ASTERISK-26959

Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6

Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in
the format modules. Neither of these functions will ever return a value
less than 0, which we were checking for in some cases.

I've introduced a fair amount of duplication in the format modules, but
I plan to change how format modules work internally in a subsequent
patch set, so this is simply a stop-gap.

Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872

In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE.  This prevents asterisk from exiting
if a module can't be loaded.  If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.

A new API was added to logger: ast_is_logger_initialized().  This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout.  If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.

Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25

We needed the reason for our reporting when agents pause/unpause all of
their queues at once.  This is a small, simple patch that adds a reason
for PAUSEALL and UNPAUSEALL.  I have been using it in production for years.

ASTERISK-26920 #close

Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d

This change removes the old epoll support which has not been used or
maintained in quite some time.

The fixed number of file descriptors on a channel has also been removed.
File descriptors are now contained in a growable vector. This can be
used like before by specifying a specific position to store a file
descriptor at or using a new API call, ast_channel_fd_add, which adds
a file descriptor to the channel and returns its position.

Tests have been added which cover the growing behavior of the vector
and the new API call.

ASTERISK-26885

Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928

This reverts commit 163e9e53.

Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b

Queue member will get stuck in pending_members if queue calls a device
that is different from the one observed for state changes.

This patch removes members from pending_members as a result of channel stasis
events such as blind or attended transfers and hangup.

ASTERISK-26862 #close

Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727

The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.

Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088

Thanks to Chris Howard for pointing this out on the wiki.

Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705

Dereferencing struct ast_autochan.chan without first calling
ast_autochan_channel_lock() is unsafe because the pointer could change at
any time due to a masquerade.  Unfortunately, ast_autochan_channel_lock()
itself uses struct ast_autochan.chan unsafely and can result in a deadlock
if the original channel happens to get destroyed after a masquerade in
addition to the pointer getting changed.

The problem is more likely to happen with v11 and earlier because
masquerades are used to optimize out local channels on those versions.
However, it could still happen on newer versions if the channel is
executing a dialplan application when the channel is transferred or
redirected.  In this situation a masquerade still must be used.

* Added a lock to struct ast_autochan to safely be able to use
ast_autochan.chan while trying to get the channel lock in
ast_autochan_channel_lock().  The locking order is the channel lock then
the autochan lock.  Locking in the other direction requires deadlock
avoidance.

* Fix unsafe ast_autochan.chan usages in app_mixmonitor.c.

* Fix unsafe ast_autochan.chan usages in app_chanspy.c.

* app_chanspy.c: Removed unused autochan parameter from next_channel().

ASTERISK-26867

Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592

A caller can leave the Queue() application after being bridged with a
member in a few ways:

  * Caller or member hangup
  * Caller is transferred somewhere else (blind or atx)
  * Caller is externally redirected elsewhere

The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.

This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.

ASTERISK-26400 #close
Reported by: Etienne Lessard

Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334

* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.

ASTERISK-24562 #close

Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe

DTMF configuration options for the binaural softmix bridge:
toggle binaural rendering (per channel).

ASTERISK-26292

Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8

Adds binaural synthesis to bridge_softmix (via convolution using libfftw3).
Binaural synthesis is conducted at 48kHz.
For a conference, only one spatial representation is rendered.
The default rendering is applied for mono-capable channels.

ASTERISK-26292

Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf

ast_load_realtime_multientry() returns an ast_config structure whose
ast_categorys are keyed with the empty strings. Several modules were
giving semantic meaning to the category names causing problems at
runtime.

* app_directory: Treated the category name as the mailbox name, and
  would fail to direct calls to the appropriate extension after an
  entry was chosen.

* app_queue: Queues, queue members, and queue rules were all affected
  and needed to be updated.

* pbx_realtime: Pattern matching would never succeed because the
  extension entered by the user was always compared to the empty
  string.

Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7

vm_authenticate doesn't always set the passed ast_vm_user argument, so
we initialize to 0 before passing it in.

ASTERISK-25893 #close
Reported by: Filip Jenicek

Change-Id: Ia3cc0128f93d352ed9add8d5c2f0f7232c2cbe4a

Original patch by John Covert, slight modifications by me.

ASTERISK-17428 #close
Reported by: John Covert
Patches:
	app_voicemail.c.patch (license #5512) patch uploaded by
        John Covert

Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6

When attempting to use VoiceMailPlayMsg with a realtime data backend
the message is located, but never retrieved. This patch adds the
required RETRIEVE and DISPOSE calls that will fetch the message from
the database (and IMAP storage as well for that matter).

Also, removed extraneous make_file call.

ASTERISK-26723 #close

Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c

When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.

This patch adds the 'u' option to Record() to override that behavior.

ASTERISK-18286 #close
Reported by: var
Patches:
	app_record-1.8.7.1.diff (license #6184) patch uploaded by var

Change-Id: Ia1cd163483235efe2db05e52f39054288553b957