Allows multiple files comprising an agent announcement
to be played by separating on the ampersand, similar
to the multi-file support in other Asterisk applications.

ASTERISK-29528

Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a

Adds application to asynchronously collect digits
dialed on a channel in the TX or RX direction
using a framehook and stores them in a specified
variable, up to a configurable number of digits.

ASTERISK-29477

Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f

Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c

Adds an application to reload modules
from within the dialplan.

ASTERISK-29454

Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774

While several applications exist to wait for
a certain event to occur, none allow waiting
for any generic expression to become true.
This application allows for waiting for a condition
to become true, with configurable timeout and
checking interval.

ASTERISK-29444

Change-Id: I08adf2824b8bc63405778cf355963b5005612f41

Hitherto, the A option has made it possible to play
audio upon answer to the called party only. This option
is expanded to allow for playback of an audio file to
the caller instead of or in addition to the audio
played to the answerer.

ASTERISK-29442

Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e

Caller ID can now be set on the called channel and
Variables can now be set on the destination
using the Originate application, just as
they can be currently using call files
or the Manager Action.

ASTERISK-29450

Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66

Adds a new ConfKick() application, which may
be used to kick a specific channel, all channels,
or all non-admin channels from a specified
conference bridge, similar to existing CLI and
AMI commands.

ASTERISK-29446

Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b

A new user option, answer_channel, adds the capability to
prevent answering the channel if it hasn't already been
answered yet.

ASTERISK-29440

Change-Id: I26642729d0345f178c7b8045506605c8402de54b

Hitherto, VoiceMail() played a non-customizable beep tone to indicate
the caller could leave a message. In some cases, the beep may not
be desired, or a different tone may be desired.

To increase flexibility, a new option allows customization of the tone.
If the t option is specified, the default beep will be overridden.
Supplying an argument will cause it to use the specified file for the tone,
and omitting it will cause it to skip the beep altogether. If the option
is not used, the default behavior persists.

ASTERISK-29349

Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280

Also removed the sample documentation, and some oddly-placed
documentation about the timeout argument to the Queue() application
itself. There is a large section on the timeout behavior below.

ASTERISK-26614 #close

Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217

ASTERISK-27542 #close

Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a

If a queue member was updated with the same status multiple
times each time a QueueMemberStatus event would be sent
which would be a duplicate of the previous.

This change makes it so that the QueueMemberStatus event is
only sent if the status actually changes.

ASTERISK-29355

Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116

Some modules have a different support level documented in their
MODULEINFO XML and Asterisk module definition. This change
brings the two in sync for the modules which were not matching.

ASTERISK-29336

Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35

This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.

ASTERISK-29335

Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90

ASTERISK-29329 #close

Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601

ASTERISK-16799 #close

Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4

Queue members using dialplan hints as a state interface must handle
INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE.

ASTERISK-28369

Change-Id: I127e06943d4b4f1afc518f9e396de77449992b9f

ASTERISK-29244

Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5

Change-Id: I350939f2220f9e5d44ddf4c8d9a4c99fde4d169a

When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received

This allows applications to perform actions based on the failure
reason.

ASTERISK-29252 #close
Reported-by: Dan Cropp

Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4

launch_monitor_thread is responsible for creating and initializing
the mixmonitor, and dependent data structures. There was one off
nominal path after the datastore gets created that triggers when
the channel being monitored is hung up prior to monitor starting
itself.

If this happened the monitor thread would not "launch", and the
mixmonitor object and associated objects are freed, including the
underlying datastore data object. However, the datastore itself was
not removed from the channel, so when the channel eventually gets
destroyed it tries to access the previously freed datastore data
and crashes.

This patch removes and frees datastore object itself from the channel
before freeing the mixmonitor object thus ensuring the channel does
not call it when destroyed.

ASTERISK-28947 #close

Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8

ASTERISK-28992 #close

Change-Id: Ia7d608924036139ee2520b840d077762d02668d0

The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.

    https://wiki.asterisk.org/wiki/x/Xc5uAg

However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.

For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.

ASTERISK-28883 #close

Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f

When using this option, answering the channel is deferred until
all prompts/greetings have been played and the caller is about
to leave their message.

ASTERISK-29118 #close

Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb

Operations that update queues when shared_lastcall is set lock the
queue in question, then have to lock the queues container to find the
other queues with the same member. On the other hand, __queues_show
(which is called by both the CLI and AMI) does the reverse. It locks
the queues container, then iterates over the queues locking each in
turn to display them.  This creates a deadlock.

* Moved queue print logic from __queues_show to a separate function
  that can be called for a single queue.

* Updated __queues_show so it doesn't need to lock or traverse
  the queues container to show a single queue.

* Updated __queues_show to snap a copy of the queues container and iterate
  over that instead of locking the queues container and iterating over
  it while locked.  This prevents us from having to hold both the
  container lock and the queue locks at the same time.  This also
  allows us to sort the queue entries.

ASTERISK-29155

Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41

ASTERISK-29144

Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62

ASTERISK-26424 #close

Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0

app_confbridge now has the ability to set the estimated bitrate on an
SFU bridge.  To use it, set a bridge profile's remb_behavior to "force"
and set remb_estimated_bitrate to a rate in bits per second.  The
remb_estimated_bitrate parameter is ignored if remb_behavior is something
other than "force".

Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a

This fixes a bug introduced mistakenly in ASTERISK-25665:
If leave-empty is enabled, a call may sometimes be removed from
a queue without recording it as abandoned.
This causes Asterisk to not generate an abandon event for that
call, and for the queue abandoned counter to be incorrect.

ASTERISK-29043 #close

Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7

This fixes the reseting members lastpause problem when realtime members is being used,
the function rt_handle_member_record was forcing the reset members lastpause because it
does not exist in realtime

ASTERISK-29034 #close

Change-Id: Ic9107e4456732a1f78412a32adb2ef87f5da40b5

Rather than putting messages into INBOX and then moving them to Urgent
later, put them directly in to the Urgent folder. This prevents
mailcmd from being skipped.

ASTERISK-27273 #close

Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5

The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages
at the same level as the scope level.  This allows the same
messages to be printed to the debug log when AST_DEVMODE
isn't enabled.

Also added a few variants of the SCOPE_EXIT macros that will
also call ast_log instead of ast_debug to make it easier to
use scope tracing and still print error messages.

Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21

Allow passing a topology from the called channel back to the
calling channel.

 * Added a new function ast_queue_answer() that accepts a stream
   topology and queues an ANSWER CONTROL frame with it as the
   data.  This allows the called channel to indicate its resolved
   topology.

 * Added a new virtual function to the channel tech structure
   answer_with_stream_topology() that allows the calling channel
   to receive the called channel's topology.  Added
   ast_raw_answer_with_stream_topology() that invokes that virtual
   function.

 * Modified app_dial.c and features.c to grab the topology from the
   ANSWER frame queued by the answering channel and send it to
   the calling channel with ast_raw_answer_with_stream_topology().

 * Modified frame.c to automatically cleanup the reference
   to the topology on ANSWER frames.

Added a few debugging messages to stream.c.

Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c

If your queues.conf had _no_ [general] section, they would default to
'yes'. Now, they always default to 'no'.

(Actually, commit ed615afb already
partially fixed it for shared_lastcall.)

ASTERISK-28951

Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6

Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.

No functional changes were made with this commit.

Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c

When stream support was added to Asterisk the stream state
was used inconsistently, resulting in odd behavior. This
was then standardized to be the state of a stream from the
perspective of Asterisk.

This change updates the StreamEcho dialplan application
to use the correct state, send only, since we are only
sending to the endpoint and not expecting them to send us
multiple video streams.

ASTERISK-28954

Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56

Because ring_entry() is not called, outgoing->chan is not touched here
either.

ASTERISK-28950
ASTERISK-28644

Change-Id: I564613715dfaf45af868251eb75a451f512af90f

Before this changeset, it was possible that a queue member (agent) was
called even though they just got out of a call, and wrapuptime seconds
hadn't passed yet.

This could happen if a member ended a call _between_ a new call attempt
and asterisk trying that particular member for a new call.

In that case, Asterisk would check the hangup time of the
call-before-the-last-call instead of the hangup time of the-last-call.

ASTERISK-28952

Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3

When send_events is enabled for a user, we were leaking a reference
to the bridge channel in confbridge_manager.c:send_message().  This
also caused the bridge snapshot to not be destroyed.

Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97