The following message:

"Subscription request from endpoint <blah> rejected. Expiration of 0 is invalid"

Would sometimes spam the log with warnings if Asterisk restarted and a bunch
of clients sent unsubscribes. This patch changes it from a warning to a debug
message.

Change-Id: I841ec42f65559f3135e037df0e55f89b6447a467

When a stale item was being updated the object was being retrieved, but its
reference was not being decremented after the update. This patch makes it so
the object is now appropriately de-referenced.

ASTERISK-28523

Change-Id: I9d8173d3a0416a242f4eba92fa0853279c500ec7

You can currently capture backtraces of memory allocations but they
only get displayed when you stop asterisk and the atexit hooks
are enabled.  Now, if memory backtrace is on and you issue a
"memory show allocations" CLI command for a specific file, then
a backtrace will show for each allocation that occurred after
you turned "memory backtrace on".  The backtrace display is shown
only when a specific file's allocations are displayed to prevent
a massive CLI dump of every file's allocations.

Change-Id: Ic657afc1fc6ec7205e16eb36a97a611d235a2b4f

It is possible for topic->name to be NULL, this causes the allocation
reference to not be logged.  Use the name variable instead which has
been verified to be a non-empty.

Change-Id: I3d0031d03c8356e4808f00cdf2d5428712575883

* Release reference returned by cache_remove
* state_alloc unconditionally bumped state_topic even when it was
  locally allocated.

Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7

Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a

This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.

ASTERISK-28533

Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492

This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.

Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.

ASTERISK-28512

Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2

On FreeBSD using the clang/llvm compiler build fails to build due
to the switch statement argument being a non integer type expression.
Switch to an if/else if/else construct to sidestep the issue.

ASTERISK-28536 #close

Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8

When fax detection occurs on an outbound PJSIP channel the
redirect operation will result in a masquerade occurring and
the underlying channel on the session changing. The code
incorrectly relocked the new channel instead of the old
channel when returning. This resulted in the new channel
being locked indefinitely. The code now always acts on the
expected channel.

ASTERISK-28538

Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3

According to RFC3550, ALL RTCP packets must be sent in a compond packet
of at least two individual packets, including SR/RR and SDES. REMB,
FIR, and NACK were not following this format, and as a result, would
fail the packet check in ast_rtcp_interpret. This was found from writing
unit tests for RTCP. The browser would accept the way we were
constructing these RTCP packets, but when sending directly from one
Asterisk instance to another, the above mentioned problem would occur.

Change-Id: Ieb140e9c22568a251a564cd953dd22cd33244605

When modifying an already defined variable in some channel drivers they
add a new variable with the same name to the list, but that value is
never used, only the first one found.

Introduce ast_variable_list_replace() and use it where appropriate.

ASTERISK-23756 #close
Patches:
  setvar-multiplie.patch submitted by Michael Goryainov

Change-Id: Ie1897a96c82b8945e752733612ee963686f32839

This fix allows a realtime moh class to be unregistered from the command
line. This is useful when the contents of a directory referenced by a
realtime moh class have changed.
The realtime moh class is then reloaded on the next request and uses the
new directory contents.

ASTERISK-17808

Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce

Added unit tests for RTCP video stats. These tests include NACK, REMB,
FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR
tests are currently disabled due to a bug. We expect to receive a
compound packet, but the code sends this out as a single packet, which
the browser accepts, but makes Asterisk upset.

While writing these tests, I noticed an issue with NACK as well. Where
it is handling a received NACK request, it was reading in only the first
8 bits of following packets that were also lost. This has been changed
to the correct value of 16 bits.

Also made a minor fix to the data buffer unit test.

Change-Id: I56107c7411003a247589bbb6086d25c54719901b

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

When the remote ISDN party ends an ISDN call on a PRI link
(DISCONNECT), CHANNEL(hangupsource) information is not available.

chan_dahdi already contains an ast_set_hangupsource() in
__dahdi_exception() function but it seems that ISDN message processing
does not use this part of code.

Two other channel modules associate ast_queue_hangup() and
ast_set_hangupsource() functions calls:
- chan_pjsip in chan_pjsip_session_end() function,
- chan_sip in sip_queue_hangup_cause() function.
chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and
set_hangup_source_and_cause().

Thus, I propose to add ast_set_hangupsource() beside
ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and
chan_sip already do.

ASTERISK-28525

Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c

The Channel resource has a new sub-resource "externalMedia".
This allows an application to create a channel for the sole purpose
of exchanging media with an external server.  Once created, this
channel could be placed into a bridge with existing channels to
allow the external server to inject audio into the bridge or
receive audio from the bridge.
See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
for more information.

Change-Id: I9618899198880b4c650354581b50c0401b58bc46

The links in the deprecation notice were the shortened
variety but it makes better sense to show the unshortened
links as they're more descriptive.

I.E.
wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
rather than
wiki.asterisk.org/wiki/x/tAHOAQ

Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9

ASTERISK-28511

Change-Id: If0d58598ce14aad3c786a1c0127b5f7b200b737d

This change removes the assumption that a frame will always have
a src set on it. This assumption is incorrect.

Given a scenario where an RTP packet is received with no payload
the resulting audio frame will have no samples. If this frame goes
through a signed linear translation path an interpolated frame can
be created (if generic packet loss concealment is enabled) that has
minimal data on it, including no src. If this frame is given to a
translation path a crash will occur due to the lack of src.

ASTERISK-28499

Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8

After receiving a 200 OK with a declined stream in response to a T.38
initiated re-invite Asterisk would crash when attempting to dereference
a NULL session media object.

This patch checks to make sure the session media object is not NULL before
attempting to use it.

ASTERISK-28495
patches:
  ast-2019-004.patch submitted by Alexei Gradinari (license 5691)

Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572

Module res_adsi.so is deprecated, therefore it does not load by default.
Module not loaded causes it to yield a FAIL when tested by tests/test_utils.c.
This fix checks if the corresponding module is loaded at the start of the test,
and if not, it passes the test and exits with a message.

This fix is applied to all versions where the module is marked deprecated.

Change-Id: I52be64c8f6af222e15148a856d1f10cb113e1e94

On reading information about initial client packet unistim use dirty
implementation of destination ip address retrieval. This fix uses
CMSG_*(..) to get ip address and make clang compile without warning.

ASTERISK-25592 #close
Reported-by: Alexander Traud

Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1