The default is 600 seconds.
Also added timeouts to the *TestGroups.json files.

Change-Id: I8ab6a69e704b6a10f06a0e52ede02312a2b72fe0

With tls and udp enabled asterisk generates a warning about sending
message via udp instead of tls.
sip notify command via cli works as expected and without warning.

asterisk has to set the connection information accordingly to connection
and not on presumption

ASTERISK-28057 #close

Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e

Use json_vsprintf from versions which contain fix for va_copy leak.

Apply fixes from jansson master:
* va_copy leak fix.
* Avoid potential invalid memory read in json_pack.
* Rename variable that shadowed another.

Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539

* Use "o*" format specifier for optional fields in ast_json_party_id.
* Stop using ast_json_deep_copy on immutable objects, it is now thread
  safe to just use ast_json_ref.

Additional changes to ast_json_pack calls in the vicinity:
* Use "O" when an object needs to be bumped.  This was previously
  avoided as it was not thread safe.
* Use "o?" and "O?" to replace NULL with ast_json_null().  The
  "?" is a new feature of ast_json_pack starting with Asterisk 16.

Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48

app_voicemail wasn't properly cleaning up the stasis cache or the
mwi topic pool when the module was unloaded or when a user was
deleted as a result of a reload.  This resulted in leaks in both
areas.

* app_voicemail now calls ast_delete_mwi_state_full when it frees
  a user structure and ast_delete_mwi_state_full in turn now calls
  the new stasis_topic_pool_delete_topic function to clear the topic
  from the pool.

Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8

When writing an RTCP report to json the code attempts to pack the "ssrc" and
"source_ssrc" unsigned integer values as a signed int value type. This of course
means if the ssrc's unsigned value is greater than that which can fit into a
signed integer value it gets converted to a negative number. Subsequently, the
negative value goes out in the json report.

This patch now packs the value as a json_int_t, which is the widest integer type
available on a given system. This should make it so the value no longer
overflows.

Note, this was caught by two failing tests hep/rtcp-receiver/ and
hep/rtcp-sender.

Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0

The append_mailbox function wasn't calculating the correct length
to pass to ast_alloca and it wasn't handling the case where context
might be empty.

Found by the Address Sanitizer.

Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161

does_id_conflict() was passing a pointer to an int to a callback
that expected a pointer to a size_t.

Found by the Address Sanitizer.

Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503

ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call
the destroy callback.  Remove call to ast_free from ast_rtp_new, leave
it to rtp_engine.c to initiate the full cleanup.  Add error detection
for the ssrc_mapping vector initialization.  In rtp_allocate_transport
set rtp->s = -1 in the failure path where we close that FD to ensure we
don't try closing it twice.

ASTERISK-27854 #close

Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451

'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults
if they are not present in the updated configuration file.

Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670

There's been a long standing leak when using topic pools.  The
topics in the pool get cleaned up when the last pool reference is
released but you can't remove a topic specifically.  If you reloaded
app_voicemail for instance, and mailboxes went away, their topics
were left in the pool.

* Added stasis_topic_pool_delete_topic() so modules can clean up
  topics from pools.
* Registered the topic pool containers so it can be examined from
  the CLI when AO2_DEBUG is enabled.  They'll be named
  "<topic_pool_name>-pool".

Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25

The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.

* No longer allocate memory from the stack in a loop to parse the header
values.  NOTE: There is a slight API change when using the passed in
strings as is.  We now require the passed in strings to no longer have
leading or trailing whitespace.  This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.

ASTERISK-28013 #close

Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a

Fixes random asterisk crash on start or reload with TLS phones.

ASTERISK-28034 #close
Reported-by: David Hajek

Change-Id: I2a859f97dc80c348e2fa56e918214ee29521c4ac

This adds a module which registers a CLI command that can set the
REMB bitrate value for REMB as it enters or exits Asterisk. This
allows you to ignore what Asterisk or a client produces and is
useful for demonstrations.

This does not generate REMB frames, however, but just modifies
them as they flow to or from a channel.

Change-Id: Ib089427c46a4a36d645cecfe02406adb38c17bec

Stasis message types are global ao2 objects and we make stasis messages
and cache entries hold references to them.  Since there are currently
situations where cache objects are never deleted, the reference count on
the types can exceed 100000 and generate a FRACK assertion message.  The
stasis message cache could conceivably also have that many messages
legitimately on large systems.

The only down side to not holding the message type ref in the stasis
message is it only makes a crash either at shutdown or when manually
unloading a busy module slightly more likely.  However, this is more
exposing a pre-existing stasis shutdown ordering issue than a problem with
not holding a message type ref in stasis messages.

* Made stasis messages and cache entries no longer hold a ref to the
message type.

Change-Id: Ibaa28efa8d8ad3836f0c65957192424c7f561707

ASTERISK-28059

Change-Id: I027472f2753391646dde594a709a75f14422db93

* Create the stasis message object without a lock as it is immutable.
* Create the stasis message type object without a lock as it is immutable.
* Creating the stasis message type could crash if the passed in type name
is NULL and REF_DEBUG is enabled.  Added missing NULL check when passing
the ao2 object tag string.

Change-Id: I28763c58bb9f0b427c11971d0103bf94055e7b32

This change brings in PJSIP 2.8, removes all the patches
that were merged upstream, and makes a minor change to
support a breaking change that was done.

ASTERISK-28059

Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189

In the original commit introducing the feature the column in the alembic
script was called 'suppress_q850_reason_header'.
In the code however the option is called 'suppress_q850_reason_headers'
(trailing 's'). This leads to errors when ARI push configuration is used.

Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f

app_voicemail was using the stasis cache to build and maintain a
list of mailboxes that had subscribers.  It then used this list
to determine if a mailbox should be polled for new messages if
polling was enabled.  For this to work, stasis had to cache every
subscription and unsubscription to the mailbox which caused a lot of
overhead, both cpu and memory related.

Since polling is only required when changes are being made to
mailboxes outside of app_voicemail and since the number of mailboxes
that don't have any subscribers is likely to be very low, all
mailboxes are now polled instead of just the ones with subscribers.

This paves the way for disabling the caching of stasis subscription
change messages.

Also fixed cleanup in some of the unit tests that not only left
test users in the users list but also caused segfaults if the tests
were run more than once.

ASTERISK-27121

Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee

Given a scenario where a session refresh was done with a removed
stream we would always add a removed stream to the outgoing SDP
even if one did not already exist.

This change makes it so that a removed stream is only placed into
the SDP if one already exists.

ASTERISK-28047

Change-Id: Ibb97d21cdeb87a8acae0c720861b0ff255708442