CHANGES

==============================================================================
===
=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE
=== PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO
=== doc/CHANGES-staging/README.md FOR MORE DETAILS.
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
==============================================================================

------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 18.0.0 --------------------------
------------------------------------------------------------------------------

Core
------------------
 * 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.

app_bridgeaddchan
------------------
 * The BridgeAdd application now behaves more like the Bridge application.
   The application now sets the BRIDGERESULT channel variable to indicate
   what happened when the channel resumes in dialplan.  This is instead of
   hanging up the channel on failure conditions.

res_pjsip
------------------
 * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
   have been added to res_pjsip endpoints that specify the preferred order
   of codecs to use between those received/sent in an SDP offer and those
   set in the endpoint configuration.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * You can now specify an optional 'Content-Type' as an argument for the Asterisk
   SendText manager action.

ARI
------------------
 * A new parameter 'inhibitConnectedLineUpdates' is now available in the
   'bridges.addChannel' call. This prevents the identity of the newly connected
   channel from being presented to other bridge members.

ARI Channels
------------------
 * 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.

Core
------------------
 * H.265/HEVC is now a supported video codec and it can be used by
   specifying "h265" in the allow line.
   Please note however, that handling of the additional SDP parameters
   described in RFC 7798 section 7.2 is not yet supported.

Features
------------------
 * Adds support for AudioSocket, a very simple bidirectional audio streaming
   protocol. There are both channel and application interfaces.

   A description of the protocol can be found on the referenced wiki page. A
   short talk about the reasons and implementation can be found on YouTube at
   the link provided.

   ARI support has also been added via the existing "externalMedia" ARI
   functionality. The UUID is specified using the arbitrary "data" field.

   Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
   YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI

Messaging
------------------
 * In order to reduce the amount of AMI and ARI events generated,
   the global "Message/ast_msg_queue" channel can be set to suppress
   it's normal channel housekeeping events such as "Newexten",
   "VarSet", etc. This can greatly reduce load on the manager
   and ARI applications when the Digium Phone Module for Asterisk
   is in use.  To enable, set "hide_messaging_ami_events" in
   asterisk.conf to "yes"  In Asterisk versions <18, the default
   is "no" preserving existing behavior.  Beginning with
   Asterisk 18, the option will default to "yes".

STIR/SHAKEN
------------------
 * STIR/SHAKEN support has been added to Asterisk. Configuration is done in
   stir_shaken.conf. There is a sample configuration file to help you get
   started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's
   set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken
   to yes on the endpoint configuration object. This will add an Identity
   header on outgoing INVITEs, and check for an Identity header on incoming
   INVITEs. This option has been added to Alembic as well.

   The information received on an incoming INVITE can be checked using the
   STIR_SHAKEN dialplan function. There are two variations:

   STIR_SHAKEN(count)
   STIR_SHAKEN(0, verify_result)

   The first variation will tell you how many STIR/SHAKEN results are on the
   channel. The second fetches information for a specific result. The first
   parameter is the index, followed by what information you want to retrieve.
   The available options are 'verify_result', 'identity', and 'attestation'.

app_chanisavail
------------------
 * The ChanIsAvail application now tolerates empty positions in the supplied
   device list.  Dialplan can now be simplified by not having to check for
   empty positions in the device list.

app_confbridge
------------------
 * A new bridge profile option, maximum_sample_rate, has been added which sets
   a maximum sample rate that the bridge will be mixed at. This allows the bridge
   to move below the maximum sample rate as needed but caps it at the maximum.

 * A new option, "text_messaging", has been added to the user profile
   which allows control over whether text messaging is enabled or
   disabled for a user. If enabled (the default) text messages
   will be sent to the user. If disabled no text messages will be
   sent to the user.

app_dial
------------------
 * The Dial application now tolerates empty positions in the supplied
   destination list.  Dialplan can now be simplified by not having to check
   for empty positions in the destination list.  If there are no endpoints to
   dial then DIALSTATUS is set to CHANUNAVAIL.

app_mixmonitor
------------------
 * An option 'S' has been added to MixMonitor. If used in combination with
   the r() and/or t() options, if a frame is available to write to one of
   those files but not the other, a frame of silence if written to the file
   that does not have an audio frame. This should prevent the two files
   from "drifting" when mixed after the fact.

 * If the 'filename' argument to MixMonitor() ended with '.wav49,'
   Asterisk would silently convert the extension to '.WAV' when opening
   the file for writing. This caused the MIXMONITOR_FILENAME variable to
   reference the wrong file. The MIXMONITOR_FILENAME variable will now
   reflect the name of the file that Asterisk actually used instead of
   the filename that was passed to the application.

app_page
------------------
 * The Page application now tolerates empty positions in the supplied