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 change to res_config_odbc that allowed empty strings to be
returned to realtime consumers¹ causes a warning to be emitted when
loading MoH classes. So we need to treat an empty 'format' as if it
was not specified to avoid the warning.

ASTERISK-28735 #close
Reported by: Ross Beer

[1] https://gerrit.asterisk.org/c/asterisk/+/13722

Change-Id: I9a271d721e1a0973e80ebe7d75b46a0d8fa0e5a5

For reasons that are not clear to me - this only appears for me when
_not_ building in dev-mode.

Change-Id: Ib45c54daaea8e0d571cb470cab1daaae2edba968

Given a scenario where session refreshes occur close to
each other while another is finishing it was possible for
the session refreshes to occur out of order. It was
also possible for session refreshes to be delayed for
quite some time if a session refresh did not result in
a topology change.

For the out of order session refreshes the first session
refresh would be queued due to a transaction in progress.
This transaction would then finish. When finished a
separate task to process the delayed requests queue
would be queued for handling. A second refresh would
be requested internally before this delayed request
queued task was processed. As no transaction was in
progress this session refresh would be immediately
handled before the queued session refresh.

The code will now check if any delayed requests exist
before allowing a session refresh to immediately occur.
If any exist then the session refresh is queued.

For the delayed session refreshes if a session refresh
did not result in a topology change the attempt would
be immediately stopped and no other delayed requests would
be processed.

The code will now go through the entire delayed requests
queue until a delayed request results in a request
actually being sent.

ASTERISK-28730

Change-Id: Ied640280133871f77d3f332be62265e754605088

Although the wiki page for the new CHANGES and UPGRADE scheme
states that the files must have the ".txt" suffix, the READMEs
didn't.

Change-Id: I490306aa2cc24d6f014738e9ebbc78592efe0f05
(cherry picked from commit 7416703f)

ASTERISK-26955 #close
Reported by: Peter Sokolov

Change-Id: Ib2803640905a77b65d0cee2d0ed2c7b310d470ac

Currently aptitude is installed using interactive mode. This patch
changes this to use the non-interactive mode as it can block
automatic dependencies installation, ex: CI, Docker build.

ASTERISK-28726 #close

Change-Id: I271ee00d230513a6f044810351a32d83b2181133

The code assumed that when the transport-cc feedback
function was called at least one packet will have been
received. In practice this isn't always true, so now
we just reschedule the sending and do nothing.

Change-Id: Iabe7b358704da446fc3b0596b847bff8b8a0da6a

In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use.  To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes"  In Asterisk versions <18, the default
is "no" preserving existing behavior.  Beginning with
Asterisk 18, the option will default to "yes".

NOTE:  This change does not affect UserEvents or the ARI
TextMessageReceived events.

* Added the "hide_messaging_ami_events" option to asterisk.conf.

* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
  the "Message/ast_msg_queue" channel if the option is set in
  asterisk.conf.  This suppresses the reporting of the events.

Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b

If you're for some reason out of RTP ports, chan_sip would previously
responde to an INVITE with a 403, which will fail the call.

Now, it returns a 503, allowing the device/proxy to retry the call on a
different machine.

ASTERISK-28718

Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90

When res_config_odbc (and perhaps other realtime backends) reads a SQL
NULL from the database, it coalesces the value to the empty string
which prevents it from being returned to the realtime core.

However, if it instead reads the empty string from the database, it
needs a way to encode that fact without having the value omitted
entirely. It does this by changing the value to a string with a single
space. The realtime code in main/config.c recognizes this special case
and _turns the string back into the empty string_ before passing it to
realtime API consumers.

For all of this to work, we need to ensure that we actually pass the
single-space-string back to the realtime core, which is currently
failing because we are trimming the value before checking its
content. So instead we now special case the single-space-string case
so that empty values are returned properly.

ASTERISK-28719 #close
Reported by: EDV O-TON

Change-Id: I673ed8c31ad037aa224e80c78c7a1dc4e4a4e3de

Incrementing stasis_app_playback.media_index directly in our playback
loop means that when we reach the end of our playlist the index into
the vector will be outside of the bounds of the vector.

Instead use a temporary variable and only assign when we're sure that
we are in bounds.

ASTERISK-28713 #close
Reported by: Sébastien Duthil

Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928

Each subscription needs to have a reference to the persisted data
for it, as well as the main JSON contained within the tree. When
recreating a subscription this did not occur and they both shared
the same reference.

ASTERISK-28714

Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a

ASTERISK-26082 #close
Reported by: Alex

Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb

It said "restrict [...] which peers should be able to pass [audio]
to each other".

However, these settings are not global (for which you would expect
signaling IPs to be checked). These settings are available per peer
only, and the IPs being checked, are the RTP IPs.

Change-Id: I2a6c6cd7c2f5f30d1df4844e3e0308a077021660

Calling 'app_send' eventually calls the app's message handler. It's possible
for a handler to obtain a lock on another object, and then need/want to lock
the app object. If the caller of 'app_send' locks the app object prior to
calling then there's a potential for a deadlock, if another thread calls
'app_send' without locking.

This patch makes it so 'app_send' is not called with the app object locked in
the section of code doing such.

ASTERISK-28423 #close

Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27

The cleanup code in stasis shuts down applications if they are in a deactivated
state, and no longer have explicit subscriptions. When registering an app the
cleanup code was running before calling 'update'. When it should be executed
after 'update' since a call to register may re-activate the app. We don't want
it to shutdown before the 'update' otherwise the app won't be re-activated,
or registered.

This patch makes it so the cleanup code is executed post 'update'.

ASTERISK-28679 #close

Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b

We need to wait for the message sending callback to finish to know if
we succeeded or failed.

ASTERISK-25421 #close
Reported by:  Dmitriy Serov

Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059

Fixes no-audio issues when the media source is changed and
strictrtp is enabled (default).

If the peer media source changes, the SDP session version also changes.
If it is lower than the one we had stored, chan_sip would ignore it.

This changeset keeps track of the remote media origin identifier,
comparing that as well. If it changes, the session version needn't be
higher for us to accept the SDP.

Common scenario where this would've caused problems: a separate media
gateway that informs the caller about premium rates before handing off
the call to the final destination.

(An alternative fix would be to set ignoresdpversion=yes on the peer.)

ASTERISK-28686

Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee

If chan_pjsip receives an RTP packet whose payload differs from the
channel's native format, and asymmetric_rtp_codec is disabled (the
default), Asterisk will switch the channel's native format to match
that of the incoming packet without regard to the negotiated payloads.

We now check that the received frame is in a format we have negotiated
before switching payloads which results in these packets being dropped
instead of causing the session to terminate.

ASTERISK-28139 #close
Reported by: Paul Brooks

Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3

When Alice calls Bob and Bob does a blind transfer to Charlie,
Bob's bridge leave event generates a finalize on both the party_a
and party_b CDRs but while the party_a CDR has the correct end time
set from the event time, party_b's leg did not. This caused that
CDR's end time to be equal to the answered time and resulted in a
billsec of 0.

* We now pass the bridge leave message event time to
cdr_object_party_b_left_bridge_cb() and set it on that CDR before
calling cdr_object_finalize() on it.

NOTE:  This issue affected transfers using chan_sip most of the
time but also occasionally affected chan_pjsip probably due to
message timing.

ASTERISK-28677
Reported by: Maciej Michno

Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca