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

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

ASTERISK-26424 #close

Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0

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

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

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

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

This patch fixes a few compile warnings/errors that now occur when using gcc
10+.

Also, the Makefile.rules check to turn off partial inlining in gcc versions
greater or equal to 8.2.1 had a bug where it only it only checked against
versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures
any version above the specified version is correctly compared.

Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9

Ensure that output buffers for the osp_convert_inout
function have sufficient space for additional data
such as brackets and ports.

ASTERISK-28804

Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b

Add a new "masquarade" channel event, and use it in app_queue to track unique id's.

Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210

ASTERISK-28829 #close
ASTERISK-25844 #close

Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6

The gcc 10 -Wrestrict option was causing "overlap" errors when
snprintf was copying one char[256] structure member to another
char[256] member in the same structure.

Using ast_alloca instead of declaring the structure inline
solves the issue.

Here's a link to the "enhancement":
https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html

We may follow up with a gcc bug report.

Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534

Since Asterisk 14, app_fax did not compile at all because Asterisk
requires that not malloc but ast_malloc is used everywhere. However,
the system headers of SpanDSP use malloc. Because we cannot (and do
not need to) change system headers, let us ignore this.

ASTERISK-28848

Change-Id: I31f7a6b92a07032c5cef1c16b8901b107fe35546

When in a conference bridge it may be necessary to have
text messages disabled for specific participants or for
all. This change adds a configuration option, "text_messaging",
which can be used to enable or disable this on the
user profile. By default existing behavior is preserved
as it defaults to "yes".

ASTERISK-28841

Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13

ASTERISK-28838

Change-Id: I68b78e7e4718be82507247433127ce3992a5ba96

Users of this should set plugin dahdi.so in their options file.

ASTERISK-16676

Change-Id: I6d01ad0a10e9fea477876d0941c3f38aac357e91

Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.

This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.

ASTERISK-28780

Change-Id: I8044c06daa06f0f16607788c596f55623be26f58

Change-Id: Icba97905e331812f129e5966e91a59b104c7a748

The optional synchronization behavior created in
64906c4c is now the default for
MixMonitor.

* Add a new flag 'n' that allows for this behavior to be turned off

* Add a notice when the 'S' option is used indicating that it is no
  longer necessary

Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4

When opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.

* If the user wants to write to a wav49 file, make sure that it is
  reflected properly in MIXMONITOR_FILENAME.

* Add a note to the documentation describing this behavior.

* Add a note in main/file.c indicating that app_mixmonitor needs to be
  changed if the logic in build_filename was changed.

ASTERISK-24798 #close
Reported by: xrobau

Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c

* The MailboxExists dialplan application was deprecated on 2006-09-26
  in Asterisk 1.6.0 (commit ec83b111)

* The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in
  Asterisk 11.0.0 (commit fd64bb66)

Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f

In af90afd9, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.

Whitespace only, no functional change.

ASTERISK~23324
Reported by: Kevin McCoy

Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87

ast_store_realtime() is not NULL tolerant, so we need to initialize
the field values we pass to it to the empty string to avoid a crash.

ASTERISK-23739 #close
Reported by: Stas Kobzar

Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c

If voicemail.conf exists but is empty, the config parsing process will
default a number of global variables to non-zero values. On the other
hand, if voicemail.conf is missing (arguably semantically equivalent
to an empty file), this process is skipped and the globals are
defaulted to 0.

Set the globals to the same values they would be set to if a
configuration were present. This allows voicemail configuration to be
done completely by Realtime without the need to create an empty
voicemail.conf file.

ASTERISK-27622 #close
Reported by: Jim Van Meggelen

Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409

This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.

A description of the protocol can be found on the above referenced
GitHub page.  A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.

ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.

ASTERISK-28484 #close

Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5

The QueueMemberPause AMI event includes two fields that return the
reason a member was paused.

* In release branches, deprecate Reason in favor of PausedReason.
* In master, remove the Reason field entirely.

ASTERISK-28349 #close
Reported by: Niksa Baldun

Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296

Additionally alter the warning to mention that it was "beep" which could
not be streamed to give admins a better clue about what the warning
means.

ASTERISK-28682

Change-Id: If5aed21226a173117ed17589f44826dd1ba6576e