Currently, if the bridge has created by the ARI, the video_mode
parameter was
not shown in the BridgeCreated event correctly.

Fixed it and added video_mode shown in the 'bridge show <bridge id>'
cli.

ASTERISK-28987

Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295

Allows a vector to be sorted in-place, rather than only during
insertion.

Change-Id: I22cba9ddf556a7e44dacc53c4431bd81dd2fa780

Change-Id: I7d951e75ad2d472fa096647dfb55670b11105e23

When dealing with a lot of video streams on WebRTC
the resulting SDPs can grow to be quite large. This
effectively doubles the maximum size to allow more
streams to exist.

The res_http_websocket module has also been changed
to use a buffer on the session for reading in packets
to ensure that the stack space usage is not excessive.

Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01

* Updated AMI version to 8.0.0
* Updated ARI version to 7.0.0
* Update make_ari_stubs.py to "Asterisk 19"

Change-Id: I51fb38c2e29f2db785f64a8bbd5565d56bea5af5

If an ACL is misconfigured in the realtime database (for instance, the
"rule" is blank) and Asterisk attempts to read the ACL, Asterisk will
crash.

ASTERISK-28978 #close

Change-Id: Ic1536c4df856231bfd2da00128f7822224d77610

I noticed this while looking at another issue and brought
it up with Teluu. It was possible for an uninitialized timer
to be cancelled, resulting in the invalid timer id of 0
being placed into the timer heap causing issues.

This change is a backport from the pjproject repository
preventing this from happening.

Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7

We read beyond the end of the buffer when copying the string out of the
buffer when we used ast_copy_string() because the original string was
not null terminated. Instead switch to ast_strndup() which does not
exhibit the same behavior.

ASTERISK-28975 #close

Change-Id: Ib4a75cffeb1eb8cf01136ef30306bd623e531a2a

Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.

Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.

Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.

Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1

session_on_rx_response wasn't checking for a NULL dialog before
attempting to get the invite session from it.

Change-Id: Id13534375966cc2eb7f2b55717c9813c63c10065

If your queues.conf had _no_ [general] section, they would default to
'yes'. Now, they always default to 'no'.

(Actually, commit ed615afb already
partially fixed it for shared_lastcall.)

ASTERISK-28951

Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6

Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.

No functional changes were made with this commit.

Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c

This commit adds the endpoint options required to control
Advanced Codec Negotiation.

incoming_offer_codec_prefs
outgoing_offer_codec_prefs
incoming_answer_codec_prefs
outgoing_answer_codec_prefs

The documentation may need tweaking and some additional edits
added, especially for the "answer" prefs.  That'll be handled
when things finalize.

This commit is safe to merge as it doens't alter any existing
functionality nor does it alter the previous codec negotiation
work which may now be obsolete.

Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11

Currently when the pjsip making an outgoing request, it keep adding the
rport parameter in a request message as a default.

This causes unexpected rport handle at the other end.

Added option for disable this behaviour in the pjsip.conf.

This is a system option, but working as a gloabl option.

ASTERISK-28959

Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc

 * ast_frame_subclass2str() and ast_frame_type2str() now return
   a pointer to the buffer that was passed in instead of void.
   This makes it easier to use these functions inline in
   printf-style debugging statements.

 * Added many missing control frame entries in
   ast_frame_subclass2str.

Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc

Tracing through synchronous tasks was a little troublesome because
the new thread's stack counter reset to 0.  This change allows
a synchronous task to set its trace level to be the same as the
thread that pushed the task.  For now, the task's level has to be
passed in the task's data structure but a future enhancement to the
taskprocessor subsystem could automatically set the trace level
of the servant to be that of the caller.

This doesn't really make sense for async tasks because you never
know when they're going to run anyway.

Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825

Do not return error if the client received ping frame
while looking for a string and just wait for another frame.

ASTERISK-28958 #close

Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922

When using the PSJIP_MEDIA_OFFER dialplan function it was not
overriding an endpoint's configured codecs on refresh unless
they had a shared codec between the two.

This patch makes it so whatever is set using PJSIP_MEDIA_OFFER
is used when creating the SDP for a refresh no matter what.

ASTERISK-28878 #close

Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6

This patch allows a user of AMI to now specify the type of message
content contained within by setting the 'Content-Type' parameter.

Note, the AMI version has been bumped for this change.

ASTERISK-28945 #close

Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb

The Streams API becomes the home for the core ACN capabilities.
These include...

 * Parsing and formatting of codec negotation preferences.
 * Resolving pending streams and topologies with those configured
   using configured preferences.
 * Utility functions for creating string representations of
   streams, topologies, and negotiation preferences.

For codec negotiation preferences:
 * Added ast_stream_codec_prefs_parse() which takes a string
   representation of codec negotiation preferences, which
   may come from a pjsip endpoint for example, and populates
   a ast_stream_codec_negotiation_prefs structure.
 * Added ast_stream_codec_prefs_to_str() which does the reverse.
 * Added many functions to parse individual parameter name
   and value strings to their respectrive enum values, and the
   reverse.

For streams:
 * Added ast_stream_create_resolved() which takes a "live" stream
   and resolves it with a configured stream and the negotiation
   preferences to create a new stream.
 * Added ast_stream_to_str() which create a string representation
   of a stream suitable for debug or display purposes.

For topology:
 * Added ast_stream_topology_create_resolved() which takes a "live"
   topology and resolves it, stream by stream, with a configured
   topology stream and the negotiation preferences to create a new
   topology.
 * Added ast_stream_topology_to_str() which create a string
   representation of a topology suitable for debug or display
   purposes.
 * Renamed ast_format_caps_from_topology() to
   ast_stream_topology_get_formats() to be more consistent with
   the existing ast_stream_get_formats().

Additional changes:
 * A new function ast_format_cap_append_names() appends the results
   to the ast_str buffer instead of replacing buffer contents.

Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56

Created new SCOPE_ functions that don't depend on RAII_VAR.  Besides
generating less code, the use of the explicit SCOPE_EXIT macros
capture the line number where the scope exited.  The RAII_VAR
versions can't do that.

 * SCOPE_ENTER(level, ...): Like SCOPE_TRACE but doesn't use
   RAII_VAR and therefore needs needs one of...

 * SCOPE_EXIT(...): Decrements the trace stack counter and optionally
   prints a message.

 * SCOPE_EXIT_EXPR(__expr, ...): Decrements the trace stack counter,
   optionally prints a message, then executes the expression.
   SCOPE_EXIT_EXPR(break, "My while got broken\n");

 * SCOPE_EXIT_RTN(, ...): Decrements the trace stack counter,
   optionally prints a message, then returns without a value.
   SCOPE_EXIT_RTN("Bye\n");

 * SCOPE_EXIT_RTN_VALUE(__return_value, ...): Decrements the trace
   stack counter, optionally prints a message, then returns the value
   specified.
   SCOPE_EXIT_RTN_VALUE(rc, "Returning with RC: %d\n", rc);

Create an ast_str helper ast_str_tmp() that allocates a temporary
ast_str that can be passed to a function that needs it, then frees
it.  This makes using the above macros easier.  Example:

   SCOPE_ENTER(1, Format Caps 1: %s  Format Caps 2: %s\n",
       ast_str_tmp(32, ast_format_cap_get_names(cap1, &STR_TMP),
       ast_str_tmp(32, ast_format_cap_get_names(cap2, &STR_TMP));

The calls to ast_str_tmp create an ast_str of the specified initial
length which can be referenced as STR_TMP.  It then calls the
expression, which must return a char *, ast_strdupa's it, frees
STR_TMP, then returns the ast_strdupa'd string.  That string is
freed when the function returns.

Change-Id: I44059b20d55a889aa91440d2f8a590865998be51

The outbound proxy for an AOR was not being applied to
any statically configured Contacts. This resulted in the
OPTIONS requests being sent to the wrong target.

This change sets the outbound proxy on statically configured
contacts once the AOR configuration is done being
applied.

ASTERISK-28965

Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0

Given a scenario where a module has a dependency on both
an external library and a module if the external library was
available and the module was not an infinite loop would
occur. This happened due to the code changing the dependecy
status to no failure on each dependency checking loop
iteration, resulting in the code thinking that it had
gone from no failure to failure each time triggering another
dependency check.

This change makes it so that the old dependency status is
preserved throughout the dependency checking allowing it to
determine that after the first iteration the dependency
status does not transition from no failure to failure.

ASTERISK-28930

Change-Id: Iea06d45d9fd6d8bfd068882a0bb7e23a53ec3e84

chan_sip handle_response() function, for a 400 response to an INVITE,
calls handle_response_invite() and does not generate ACK.
handle_response_invite() does not recognize 400 response and has no
default response processing for unexpected responses, thus it does not
generate ACK either.
The ACK on response repetition comes from handle_response() mechanism
"We must re-send ACKs to re-transmitted final responses".

According to code history, 400 response specific processing was
introduced with commit
"channels/chan_sip: Add improved support for 4xx error codes"
This commit added support for :
- 400/414/493 in handle_response_subscribe() handle_response_register()
  and handle_response().
- 414/493 only in handle_response_invite().

This fix adds 400 response support in handle_response_invite().

ASTERISK-28957

Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad

A patch made a reference to the PJSIP_SC_NULL enumeration value, which
was added to pjproject 2.8 and above thus making it so Asterisk would
fail to compile with prior versions of pjproject.

This patch removes the reference, and instead initializes the value
to '0'.

ASTERISK-28886 #close

Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7

1) Fix memory-leaks
   Added code to release ast_events extracted from corosync and stasis messages

2) Clean stasis cache when a member of the corosync cluster leaves the group
   Added code to remove from the stasis cache of the members remained on the
   group all the messages with the EID of the left member.
   If the device states of the left member remain in the stasis cache of other
   members, they will not be updated anymore and high priority cached values,
   like BUSY, will take precedence over current device states.

3) Stop corosync event propagation when node is not joined to the group
   Updated dispatch_thread_handler code to detect when asterisk is not joined
   to the corosync group and added some condition in publish_event_to_corosync
   code to send corosync messages only when joined.
   When a node is not joined its corosync daemon can't send messages:
   the cpg_mcast_joined function append new messages to the FIFO buffer until
   it's full and then it blocks indefinitely.
   In this scenario if the stasis_message_cb callback, registered by
   res_corosync to handle stasis messages, try to send a corosync messages,
   the thread of the stasis thread-pool will be blocked until the node join
   the corosync cluster.

ASTERISK-28888
Reported by: Università di Bologna - CESIA VoIP

Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2

ASTERISK-28949

Change-Id: Id465030f2b1997b83d408933fdbabe01827469ca

When stream support was added to Asterisk the stream state
was used inconsistently, resulting in odd behavior. This
was then standardized to be the state of a stream from the
perspective of Asterisk.

This change updates the StreamEcho dialplan application
to use the correct state, send only, since we are only
sending to the endpoint and not expecting them to send us
multiple video streams.

ASTERISK-28954

Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56

The change to how setvar works for various channels performed in
ASTERISK~23756 missed some required change in the dahdi channel,
where the variables are actually set while reading configuration.
This change should fix the issue.

ASTERISK-28955

Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274

When a re-INVITE is received we create a new set of
streams that are then swapped in as the active streams.
We did not preserve the SDP label from the previous
streams, resulting in the label getting lost.

This change ensures that if an SDP label is present
on the previous stream then it is set on the new stream.

ASTERISK-28953

Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445

The AMI action and CLI command did not take into account the properties
of full backend caching. This resulted in an expired object remaining
removed until a full backend update occurred, instead of having the
object updated when needed.

This change makes it so that the AMI action and CLI command for object
expire will now fail instead of putting the cache into an undesired
state. If full backend caching is enabled then only operations
which act on the entire cache are available.

ASTERISK-28942

Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4

Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is
sent, the caller ID will be checked to see if there is a certificate
that corresponds to it. If so, that information will be retrieved and an
Identity header will be added to the SIP message. The format is:

header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken

Header, payload, and signature are all BASE64 encoded. The public key
URL is retrieved from the certificate. Currently the algorithm and ppt
are ES256 and shaken, respectively. This message is signed and can be
used for verification on the receiving end.

Two new configuration options have been added to the certificate object:
attestation and origid. The attestation is required and must be A, B, or
C. origid is the origination identifier.

A new utility function has been added as well that takes a string,
allocates space, BASE64 encodes it, then returns it, eliminating the
need to calculate the size yourself.

Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4

Because ring_entry() is not called, outgoing->chan is not touched here
either.

ASTERISK-28950
ASTERISK-28644

Change-Id: I564613715dfaf45af868251eb75a451f512af90f

Change-Id: I24b5453df412232cf7f9a171ea4a34b35ad3ae78

Before this changeset, it was possible that a queue member (agent) was
called even though they just got out of a call, and wrapuptime seconds
hadn't passed yet.

This could happen if a member ended a call _between_ a new call attempt
and asterisk trying that particular member for a new call.

In that case, Asterisk would check the hangup time of the
call-before-the-last-call instead of the hangup time of the-last-call.

ASTERISK-28952

Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3

This patch makes the usual necessary changes when upgrading to a new
version pjproject. For instance, version number bump, patches removed
from third-party, new *.md5 file added, etc..

This patch also includes a change to the Asterisk pjproject Makefile to
explicitly create the 'source/pjsip-apps/lib' directory. This directory
is no longer there by default so needs to be added so the Asterisk
malloc debug can be built.

This patch also includes some minor changes to Asterisk that were a result
of the upgrade. Specifically, there was a backward incompatibility change
made in 2.10 that modified the "expires header" variable field from a
signed to an unsigned value. This potentially effects comparison. Namely,
those check for a value less than zero. This patch modified a few locations
in the Asterisk code that may have been affected.

Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to
check a minimum version of pjproject at compile time.

ASTERISK-28899 #close

Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81

When requesting a Local channel the requested stream topology
or a converted stream topology will now be placed onto the
resulting channels.

Frames written in on streams will now also preserve the stream
identifier as they are queued on the opposite channel.

Finally when a stream topology change is requested it is
immediately accepted and reflected on both channels. Each
channel also receives a queued frame to indicate that the
topology has changed.

ASTERISK-28938

Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186

If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly.

Fixed it to parse the channelId, other_channel_id parameter correclty.

ASTERISK-28948

Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157

The "value" passed in when setting an RTP property determines
whether it should be enabled or disabled. The RTP send and
receive retrans props did not examine this to know if the
buffers should be enabled. They assumed they always should be.

This change makes it so that the "value" passed in is
respected.

ASTERISK-28939

Change-Id: I9244cdbdc5fd065c7f6b02cbfa572bc55c7123dc