Change includes move to netsock2 library.

ASTERISK-27164
Reported-by: Adam Secombe

Change-Id: Ia9e8dc3d153de7a291dbda4bd87fc827dd2bb846

Change-Id: I5e1d4a09e58b92b541bc8ed6f9e10e54c4e5101f

Change-Id: I7729ecceedceb12f52bf18dae259846aa1d993b3

If the review to be tested is in a project with restricted access,
we need to use the jenkins user's gerrit https credentials when we
do the checkout or the checkout will fail.

Change-Id: I9dc9994763c5ebfeb9f1cff60fb53f6902b7fd5f

If in the initial sdp the caller doesn't include the line
a=rtcp-mux

Then asterisk shoud not include rtcp-mux in the response regardless
of rtcp-mux being enabled on the endpoint

ASTERISK-28007 #close

Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7

Change-Id: I3f396316bb40d1ae6e91f5f688042420f1a540ed

Cleanup idle check scheduled events at shutdown.

Change-Id: I61bfbb56bac69fe840c3242927d31ff3593be461

This makes it possible for REF_DEBUG to report no leaks when loading
res_odbc.

Change-Id: I1a3dea786bd6e7f4820a6dd5cbaa197fa783ce93

sip_options_get_endpoint_state_compositor_state leaked a reference to
the first available endpoint state compositor that was found.

Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7

Change-Id: I11641d65592536dea9cbca5aa94a24c25d24dd5f

This change adds the "party" parameter to the Remote-Party-ID header
which indicates which party the header information is applicable
to. In Asterisk this is determined on whether we are the calling
or called party. This is added to improve interoperability with some
implementations.

ASTERISK-28006

Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca

When a conference contained a mixture of audio/video and audio-only
users, a NOTICE message would pop up stating there are no joint
capabilities between streams. This happens because streams can never be
removed, but they can be in a REMOVED state. If we have the scenario
where user A joins with audio/video, user B joins with audio-only, and
user C joins with audio/video, then user A leaves, the message would
be triggered. That removed stream is still in the SDP, but Asterisk
would pass it through, causing it to be seen as a ulaw stream. A check
has been added for removed streams, setting their status to REMOVED when
handling negotiated SDPs.

Also addressed an issue where user A joins, then user B joins but does
not receive video until much later. Full frames were not being sent,
causing some PLI from the browser. Because the video was flowing in one
direction, the browser sets the SSRC to 1, but Asterisk was dropping the
PLI because of that. Added a check to see if the SSRC is 1 or not, which
sends full frames and allows video to flow between user A and user B.
This should only happen when dealing with PSFB or FUR, and in the case
of PSFB, only for PLI.

ASTERISK-27398

Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e

When a call leaves a queue on leaveempty condition, QUEUESTATUS
must be set to LEAVEEMPTY, no matter whether Queue was executed with or
without the "c" (continue) option.

The regression was introduced in the fix for ASTERISK_25665.
The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was
overwritten in case when "c" is set, regardless of what was the cause
for leaving the queue.

ASTERISK-27973 #close
Reported-by: Valentin Safonov

Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c

It is valid for a config file to be empty or contain only comments, but
not valid for a config value to be set when no uncommented context
exists.  This caused an error to be loged numerous times during start
when loading the default pjsip.conf.

Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6

Enable coverage with `./tests/CI/buildAsterisk.sh --coverage`.  This
will cause Asterisk to be compiled with coverage support.  It also
initializes 'before' coverage data for all sources.  Accept
--tested-only to disable modules which are not run by any test.
Enabling coverage also sets tested-only true by default.  To build
everything with coverage enabled use `--coverage --tested-only=0`.

./tests/CI/processCoverage.sh is used to process the coverage and
generate HTML reports.

Fix utils/check_expr2 which failed to compiled with coverage enabled.

Add status output 5 times per stage of astobj2_test_perf to ensure
remote CLI does not timeout when compiled with coverage.  Remote CLI
disconnects if no output is received for 60 seconds.  When coverage is
enabled it takes about 70 seconds for my laptop to run the stages of
this test, so with the change a message is printed every 14 seconds.

Change-Id: I890f7d5665087426ad7d3e363187691b9afc2222

Change-Id: I9cf97bdc756012d1f552ab007f4aa85e0ddb4e62

When the stasis cache is used a hash is calculated for
retrieving or inserting messages. This change calculates
a hash when the message type is initialized that is then
used each time needed. This ensures that the hash is
calculated only once for the message type.

Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37

The authors of PJProject undef s_addr because of some issue in Microsoft
Windows. However in Oracle Solaris, s_addr is not a structure member, but
defined to map to the real structure member.

Updates the patch from ASTERISK_20366

ASTERISK-27997

Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11

* Don't include pjlib.h twice in res_pjsip.h
* Consistently use #include <> form for pjproject includes.
(pjsip.h and pjlib.h)

Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7

If asterisk offer an endpoint with SRTP and that endpoint respond
with non srtp, in that case channel(rtp,secure,audio) reply wrong
status.

Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key:
Currently this flag has being set redundantly. In either case identical
or different remote_key this flag has being set. So if we
don't set it while we receive identical remote_key or non SRTP SDP
response then we can take decision of srtp use by using that flag.

ASTERISK-27999

Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7

The script configure from Teluu expects shared libraries (.so) in a subfolder
called 'lib', when --with-xyz=PATH is specified. However for OpenSSL, the
default location is the root of the source folder = PATH. Furthermore, Asterisk
supports both, 'lib' and root. For consistency and because Asterisk is using
(only) OpenSSL in PJProject, it is enhanced to support both locations, just
like Asterisk.

ASTERISK-27995

Change-Id: I8eb916a88b6b8c22e29bb40bee8faaca6c73406f