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

OpenSSL can not tolerate if the packet sent out does not
match the length that it provided to the sender. This change
lies and says that each time the full packet was sent. If
a problem does occur then a retransmission will occur as
appropriate.

ASTERISK-28576

Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449

If the last line of a file does not have a terminating EOL sequence, we
potentially add garbage to the value returned from the FILE() function.

There is no overflow potential here as we are reading from a buffer of a
known size, we are just reading too much of it.

ASTERISK-26481 #close

Change-Id: I50dd4fcf416fb3c83150040a1a79a59d9eb1ae01

MOH continues to play to a channel if that channel was on hold prior to
entering a softmix bridge. MOH will not stop even if the original "holder"
attempts an unhold.

For the most part a softmix bridge ignores holds, so a participating channel
shouldn't join while on hold. This patch checks to see if the channel joining
the softmix bridge is currently on hold. If so then it indicates an unhold.

ASTERISK-28618

Change-Id: I66ccd4efc80f5b4c3dd68186b379eb442916392b

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

resource_events:stasis_app_message_handler() was locking the session,
then attempting to determine if the app had debug enabled which
locked the app_registry container.  res_stasis:__stasis_app_register
was locking the app_registry container then calling app_update
which caused app_handler (which locks the session) to run.
The result was a deadlock.

* Updated resource_events:stasis_app_message_handler() to determine
  if debug was set (which locks the app_registry) before obtaining the
  session lock.

* Updated res_stasis:__stasis_app_register to release the app_registry
  container lock before calling app_update (which locks the sesison).

ASTERISK-28423
Reported by Ross Beer

Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4

There exists a scenario where a thread can hold a lock on the
channels container while trying to lock a bridge. At the same
time another thread can hold the lock for said bridge while
attempting to retrieve a channel. This causes a deadlock.

This change fixes this scenario by retrieving a channel snapshot
instead of a channel, as information present in the snapshot
is all that is needed.

ASTERISK-28616

Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2

Found during some testing, there is a race condition between selecting an
appropriate bridge type for a call versus the applying of media on the callee's
session. In some instances a native bridge type would have been chosen, but
due to the callee's media not yet being established at bridge compatibility
check time the simple bridge type is picked instead.

When using chan_pjsip this initiates a topology change event. The topologies
are then compared for the two sessions. However, when the topology was created
for the caller its streams are initialized to "inactive". This topology is then
used as a base when creating the callee's topology, and streams. Soon after
the caller's topology's stream(s) get updated based on the sdp (get set to
sendrecv in the failing scenario).

Now when the topology change event is raised, and the two topologies are
compared, the comparison fails due to a stream state mismatch (sendrecv vs
inactive). And since they differ a reinvite is sent out (to the caller in
this case).

This patch makes it such that when the caller's topology is initially created
it gets created based on its configured endpoint's media topology. When the
endpoint's topology is created its stream's state(s) are initialized to
sendrecv instead of inactive. Subsequently, now when the callee's topology is
created its topology streams are now initialized to sendrecv. Thus when the
topology change event occurs due to the mentioned scenario the stream states
match for the given sessions, and the reinvite is not sent unless due to some
other valid mismatch.

Note, this patch only changes one pending media state's creation point. It's
possible other places *could* be changed, however for now it was deemed best
to only alter what's here.

Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7

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 the "max_retries" option is set to 0 then upon failure no
further attemps are made, so explicitly document the behavior.

ASTERISK-28602

Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf

Calling ne_uri_parse allocates memory that needs to be freed with a
corresponding call to ne_uri_free.

ASTERISK-28572 #close

Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61

This change ensures that the module isn't unloaded when a
WebSocket is open. Previously it was possible to unload the
module manually or during shutdown which could cause a crash
when any active WebSockets were terminated.

ASTERISK-28585

Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c

ASTERISK-28590 #close

Change-Id: I51abce00c04d0a06550bda5205580705185b9c1c

When we created the External Media addition to ARI we created an
ExternalMedia object to be returned from the channels/externalMedia
REST endpoint.  This object contained the channel object that was
created plus local_address and local_port attributes (which are
also in the Channel variables).  At the time, we thought that
creating an ExternalMedia object would give us more flexibility
in the future but as we created the sample speech to text
application, we discovered that it doesn't work so well with ARI
client libraries that a) don't have the ExternalMedia object
defined and/or b) can't promote the embedded channel structure
to a first-class Channel object.

This change causes the channels/externalMedia REST endpoint to
return a Channel object (like channels/create and channels/originate)
instead of the ExternalMedia object.

Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9

During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.

ASTERISK-28578

Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f

This was only supposed to be for testing, so now it can be
removed.

Change-Id: I3dfc2e776e70b3196aeed5688372ea80c0214b59

ASTERISK-28586 #close

Change-Id: I43dc4e8bd9dc685b17695b215a5360314074734f

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

Now that both FIR and REMB are being sent in compound packets
these tests can be enabled.

This also extends the REMB nominal test to cover the REMB
contents itself.

Change-Id: Ibfee526ad780eefcce5dd787f53785382210024a

PostgreSQL 12 finally removed column adsrc from table pg_catalog.pg_attrdef
(column default values), which has been deprecated since version 8.0.
Since then, the official/correct/supported way to retrieve the column
default value from the catalog is function pg_catalog.pg_get_expr().

This change breaks compatibility with pre-8.0 PostgreSQL servers,
but has reached end-of-support more than a decade ago.
cdr_pgsql and res_config_pgsql still have support for pre-7.3
servers, but cleaning that up is perhaps a topic for a major release,
not this bugfix.

ASTERISK-28571

Change-Id: I834cb3addf1937e19e87ede140bdd16cea531ebe

When creating an unsolicited MWI aggregate subscription it was possible for
the subscription object to be double unref'ed. This patch removes the explicit
unref as it is not needed since the RAII_VAR will handle it at function end.

Less concerning there was also a bug that could potentially allow the aggregate
subscription object to be added to the unsolicited container twice. This patch
ensures it is added only once.

ASTERISK-28575

Change-Id: I9ccfdb5ea788bc0c3618db183aae235e53c12763

Fixes a failure in /main/taskprocesor unit test, only occurring in Ubuntu.
Newer versions of GCC require variable initialization.

Change-Id: I2994d8aab9307a8c2c7330584f287a27144a580c

* changes:
  res_pjsip_mwi: use an ao2_global object for mwi containers
  res_pjsip/res_pjsip_mwi: use centralized serializer pools