This establishes the basic allocation/destruction of an SDP state
object, plus some of the simpler getter methods involved. Subsequent
tasks will deal with adding a state machine, creating SDPs from
capabilities and options, and merging SDPs into a joint SDP.

Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709

This creates the following:
* Asterisk's internal representation of an SDP
* An API for translating SDPs from one format to another
* An implementation of a translator for PJMEDIA

Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b

This is step one of adding an SDP API: defining some
configurable settings for SDPs. This is based on options
that are currently supported in Asterisk.

Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7

Original patch by John Covert, slight modifications by me.

ASTERISK-17428 #close
Reported by: John Covert
Patches:
	app_voicemail.c.patch (license #5512) patch uploaded by
        John Covert

Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6

When attempting to use VoiceMailPlayMsg with a realtime data backend
the message is located, but never retrieved. This patch adds the
required RETRIEVE and DISPOSE calls that will fetch the message from
the database (and IMAP storage as well for that matter).

Also, removed extraneous make_file call.

ASTERISK-26723 #close

Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c

When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.

This patch adds the 'u' option to Record() to override that behavior.

ASTERISK-18286 #close
Reported by: var
Patches:
	app_record-1.8.7.1.diff (license #6184) patch uploaded by var

Change-Id: Ia1cd163483235efe2db05e52f39054288553b957

ASTERISK-26775 #close

Change-Id: I86de4b1a699d6edc77fea9b70d839440e4088284

This change adds unit tests for the various API calls relating
to stream topologies. This includes creation, destruction,
inspection, and manipulation.

Through this a few bugs were uncovered in the implementation:

1. Creating a topology using a format capabilities would fail as
the code considered a return value of 0 from the append stream
function to indicate an error which is incorrect.

2. Not all functions which placed a stream into a topology
set the position on the stream itself.

3. Appending a stream would cause a frack if the position
provided was the last one. This occurred because the existing
stream was queried but the index was outside of what the
vector was currently at for size.

ASTERISK-26786

Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0

* app_minivm: Use built-in completion facilities to complete optional
arguments.

* app_voicemail: Use built-in completion facilities to complete
optional arguments.

* app_confbridge: Add missing colons after 'Usage' text.

* chan_alsa: Use built-in completion facilities to complete optional
arguments.

* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'

* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'

* func_odbc: Correct completions for 'odbc read' and 'odbc write'

* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.

* main/bridge: Correct completions for 'bridge kick.'

* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.

* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'

* main/pbx_app: Remove redundant completions for 'core show
applications.'

* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'

* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.

Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca

This change adds the media stream topology definition and API for
accessing and using it.

Some refactoring of the stream was also done.

ASTERISK-26786

Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568

When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com),
the user part is stripped down as it would be a trunk with a specified user,
and only the host part is called as a PJSIP endpoint and can't be found.
This is not correct in the case of a multidomain SIP account, so the stripping
after the @ sign is done only if the whole endpoint (in multidomain case
1000@test.com) can't be found.

ASTERISK-26248

Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6

The ast_waitfor_nandfds operation will manipulate the flags
of channels passed in. This was previously done without
the channel lock being held. This could result in incorrect
values existing for the flags if another thread manipulated
the flags at the same time.

This change locks the channel during flag manipulation.

ASTERISK-26788

Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed

The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE
but we have no authenticator registered to create the challenge.

Change-Id: I62368180d774b497411b80fbaabd0c80841f8512

In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c496. This patch
restores that behavior.

This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.

ASTERISK-26115 #close
Reported by: Nasir Iqbal

Change-Id: I8afbd06725e99610e02adb529137d4800c05345d

We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.

* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging.  Made hold the channel lock after the called
party answers while updating the caller channel staging.

* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.

* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.

* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.

Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a

This change adds the media stream definition and API for
accessing and using it. Unit tests have also been written
which exercise aspects of the API.

ASTERISK-26773

Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87

The entry for 'identify' was incorrectly placed in the
res_pjsip section when it should be in
res_pjsip_endpoint_identifier_ip.

ASTERISK-26785 #close

Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a

This reverts commit 6492e913.

The change in question was intended to prevent the need to reload in
order to update qualifies on contacts when an AOR changes. However, this
ended up causing a deadlock instead.

Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e

When performing an SRV lookup using the ast_srv_lookup function it
did not properly handle the situation where 0 records are returned.
If this happened it would wrongly assume that at least one record
was present.

This change fixes the code so it will exit early if an error occurs
or if 0 records are returned.

ASTERISK-26772
patches:
  srv_lookup.patch submitted by nappsoft (license 6822)

Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351

The adding and removing of device state subscriptions did not protect
fully against simultaneous manipulation. In particular the subscribe
case allowed a small window where two subscriptions could be added for
the same device state instead of just one.

This change makes the code hold the subscriptions lock for the entirety
of each operation to ensure that two are not occurring at the same time.

ASTERISK-26770

Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b

Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d

In ari.conf, when setting the option channelvars, every Stasis channel
snapshot would create a list of variable/value that would not be freed
when the snapshot is freed, resulting in a often-recurring memory
leak.

ASTERISK-26767 #close

Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d