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

This patch fixes some wrongly formatted documentation for the AgentLogin
application. A couple of "see also" links should contain only the function
name, and no parameters.

Change-Id: I3f788b47dce3292e311f8a9856938d59a0bd0661

Dialplan has to be careful about passing an empty device list or empty
positions in the list.  As a result, dialplan has to check for these
conditions before using ChanIsAvail.  Simplify dialplan by making
ChanIsAvail handle these conditions gracefully.

* Made tolerate empty positions in the device list.

* Simplified the code and eliminated some unnecessary indention.

ASTERISK-28638

Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3

* Made BridgeAdd not hangup the call if there is a problem.
* Reduced message level from warning to verbose for normal exception
cases.
* Added a loop safety check to BridgeAdd.
* Made BridgeAdd set BRIDGERESULT with the status when dialplan is
resumed.

Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426

Dialplan has to be careful about passing an empty destination list or
empty positions in the list.  As a result, dialplan has to check for
these conditions before using Dial.  Simplify dialplan by making Dial
handle these conditions gracefully.

* Made tolerate empty positions in the dialed device list.

* Reduced some message log levels from notice to verbose.

ASTERISK-28638

Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9

Dialplan has to be careful about passing an empty destination list or
empty positions in the list.  As a result, dialplan has to check for
these conditions before using Page.  Simplify dialplan by making Page
handle these conditions gracefully.

* Made tolerate empty positions in the paged device list.

* Reduced some warnings associated with the 's' option to verbose
messages.  The warning level for those messages really serves no purpose
as that is why the 's' option exists.

ASTERISK-28638

Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3

Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a

Why log a warning for something your dialplan explicitly asked for?

Change-Id: I167b90daf4c7d75dd4b7ef94849f6cef05aa43a7

Temporary channel lifespan is very short and CDR deactivation request
through ast_cdr_set_property() may happen when CDR is not available
yet. Use CDR_PROP() dialplan function instead, it will first wait
for pending CDR insertion requests to be processed.

ASTERISK-28636

Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074

ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.

ASTERISK-28658

Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee

ASTERISK-28644

Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b

ASTERISK_28143 attempted to fix an issue where calls with no audio would never
timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its
calculations. Unfortunately these frames seem to show up at irregular time
intervals. This resulted in app_amd returning prematurely most of the time.

* Removed AST_FRAME_NULL from the calculations
* Added a check to see how much time has actually passed since app_amd began

ASTERISK-28608

Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42

ASTERISK-28614

Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664

This patch fixes several issues reported by the lgtm code analysis tool:

https://lgtm.com/projects/g/asterisk/asterisk

Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:

* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards

Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25

The following modules needed tweaks for API changes.

addons/cdr_mysql.c
addons/chan_ooh323.c
apps/app_meetme.c

ASTERISK-28604

Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d

If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav"
and are using realtime ODBC backend, only the first format gets stored
in the database. So when you forward a message later on, there is a bug
generating the email, related to the stored format (GSM) being different
than the desired email format (WAV) specified for the user. Sox can
handle this, but Asterisk needs to tell sox exactly what to do.

ASTERISK-22192

Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd

This reverts commit fd2e8d0d.

Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.

ASTERISK-28567 #close

Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41

Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a

ChanIsAvail() creates a temporary channel with ast_request() to test
resource availability. It should not generate a CDR when it hangs up
this temporary channel.

This patch disables CDR generation for the temporary channel with
ast_cdr_set_property().

ASTERISK-28527

Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1

There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:

 1. There is no rx and no tx audio, so return nothing.
 2. There is rx but no tx audio, so return rx.
 3. There is tx but no rx audio, so return tx.
 4. There is rx and tx audio, so mix them and return.

The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.

If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.

This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.

Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e

The function leave_voicemail checks if expungeonhangup is set,
but does not check if IMAP stream is closed,
so it could call imap function with NULL stream.
This leads to segfault.

ASTERISK-28505 #close

Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c

You now select voicemail backends like normal dialplan applications, so
there is no longer a need for their own menuselect category.

Reported by snuff-work in #asterisk-dev

Change-Id: Idfa4c9c8349726074318a9e6b68d24c374521005

There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:

unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);

would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.

ASTERISK-28480

Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1