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.
===
==============================================================================

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.9.0 to Asterisk 18.10.0 -----------
------------------------------------------------------------------------------

Applications
------------------
 * added support for Danish syntax, playing the correct plural sound file
   dependen on where you have 1 or multipe messages
   based on the existing SE/NO code

 * added that we set DIALEDPEERNUMBER on the outgoing channels
   so it is avalible in b(content^extension^line)
   this add the same behaviour as Dial

Core
------------------
 * Bundled PJProject Build

   The build process has been updated to make pjproject troubleshooting
   and development easier. See third-party/pjproject/README-hacking.md or
   https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject
   for more info.

ami
------------------
 * An AMI event now exists for "Wink".

app_mf
------------------
 * Adds MF receiver and sender applications to support
   the R1 MF signaling protocol, including integration
   with the Dial application.

app_queue
------------------
 * added that we set DIALEDPEERNUMBER on the outgoing channels
   so it is avalible in b(content^extension^line)
   this add the same behaviour as Dial

app_queues
------------------
 * adding support for playing the correct en/et for nordic languages

 * Don't play sound_thanks if there is no leading hold_time message
   When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience"

app_sendtext
------------------
 * A ReceiveText application has been added that can be
   used in conjunction with the SendText application.

app_voicemail
------------------
 * added support for Danish syntax, playing the correct plural sound file
   dependen on where you have 1 or multipe messages
   based on the existing SE/NO code

cdr
------------------
 * A new CDR option, channeldefaultenabled, allows controlling
   whether CDR is enabled or disabled by default on
   newly created channels. The default behavior remains
   unchanged from previous versions of Asterisk (new
   channels will have CDR enabled, as long as CDR is
   enabled globally).

chan_sip.c
------------------
 * resolve issue with pickup on device that uses "183" and not "180"

cli
------------------
 * The "module refresh" command has been added,
   which allows unloading and then loading a
   module with a single command.

func_json
------------------
 * The JSON_DECODE dialplan function can now be used
   to parse JSON strings, such as in conjunction with
   CURL for using API responses.

res_fax_spandsp
------------------
 * Adds support for spandsp 3.0.0.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.8.0 to Asterisk 18.9.0 ------------
------------------------------------------------------------------------------

ToneScan application
------------------
 * A new application, ToneScan, allows for
   synchronous detection of call progress
   signals such as dial tone, busy tone,
   Special Information Tones, and modems.

app_playback
------------------
 * A new option 'mix' is added to the Playback application that 
   will play by filename and say.conf. It will look on the format of the 
   name, if it is like say format it will play with say.conf if not it 
   will play the file name.

app_queue
------------------
 * Add field to save the time value when a member enter a queue.
   Shows this time in seconds using 'queue show' command and the
   field LoginTime for responses for AMI the events.

   The output for the CLI command `queue show` is changed by added a
   extra data field for the information of the time login time for each
   member.

apps
------------------
 * A new option 'mix' is added to the Playback application that 
   will play by filename and say.conf. It will look on the format of the 
   name, if it is like say format it will play with say.conf if not it 
   will play the file name.

ast_coredumper
------------------
 * New options:
    --pid=<asterisk_pid>
      Allows specification of an Asterisk instance when trying to
      and the script can't determine it itself.
    --libdir=<system library directory>
      Allows specification of a non-standard installation directory
      containing the Asterisk modules.
    --(no-)rename
      Renames the coredump and the output files with readable
      timestamps. This is the default.
   Removed unneeded or confusing options:
    --append-coredumps
    --conffile
    --no-default-search
    --tarball-uniqueid
   Changed Variables:
    COREDUMPS is now just "/tmp/core!(*.txt)"
    DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ'
   Changed behavior:
    If you use 'running' or 'RUNNING' you no longer need to specify
    '--no-default-search' to ignore existing coredumps.

chan_iax2
------------------
 * Both a secret and an outkey may be specified at dial time,
   since encryption is possible with RSA authentication.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.7.0 to Asterisk 18.8.0 ------------
------------------------------------------------------------------------------

MessageSend
------------------
 * The MessageSend AMI action has been updated to allow the Destination
   and the To addresses to be provided separately. This brings the
   MessageSend manager command in line with the capabilities of the
   MessageSend dialplan application.

func_channel
------------------
 * Adds the CHANNEL_EXISTS function to check for the existence
   of a channel by name or unique ID.

func_vmcount
------------------
 * Multiple mailboxes may now be specified instead of just one.

logger
------------------
 * Added the ability to define custom log levels in logger.conf
   and use them in the Log dialplan application. Also adds a
   logger show levels CLI command.

res_pjsip_registrar
------------------
 * Adds new PJSIP AOR option remove_unavailable to either
   remove unavailable contacts when a REGISTER exceeds
   max_contacts when remove_existing is disabled, or
   prioritize unavailable contacts over other existing
   contacts when remove_existing is enabled.

res_pjsip_t38
------------------
 * In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
   fallback use of the transport's bind address solve problems sending
   media on systems that cannot send ipv4 packets on ipv6 sockets, and
   certain other situations. This change extends both of these behaviors
   to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
   problems on these systems, introducing a new option
   endpoint/t38_bind_udptl_to_media_address.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.6.0 to Asterisk 18.7.0 ------------
------------------------------------------------------------------------------

Channel-agnostic MF support
------------------
 * A SendMF application and PlayMF manager
   application are now included to send
   arbitrary standard R1 MF tones on the
   current channel or another specified channel.

app_milliwatt
------------------
 * The Milliwatt application's existing behavior is
   incorrect in that it plays a constant tone, which
   is not how digital milliwatt test lines actually
   work.

   An option is added so that a proper milliwatt test
   tone can be provided, including a 1 second silent
   interval every 10 seconds. However, for compatability
   reasons, the default behavior remains unchanged.

app_morsecode
------------------
 * Extends the Morsecode application by adding support for
   American Morse code and adds a configurable option
   for the frequency used in off intervals.

app_originate
------------------
 * Codecs can now be specified for dialplan-originated
   calls, as with call files and the manager action.
   By default, only the slin codec is now used, instead
   of all the slin* codecs.

app_queue
------------------
 * Reload behavior in app_queue has been changed so
   queue and agent stats are not reset during full
   app_queue module reloads. The queue reset stats
   CLI command may still be used to reset stats while
   Asterisk is running.

app_read
------------------
 * A new option allows the digit '#' to be read literally,
   rather than used exclusively as the input terminator
   character.

app_voicemail
------------------
 * Add a new 'S' option to VoiceMail which prevents the instructions
   (vm-intro) from being played if a busy/unavailable/temporary greeting
   from the voicemail user is played. This is similar to the existing 's'
   option except that instructions will still be played if no user
   greeting is available.

chan_iax2
------------------
 * ANI2 (OLI) is now transmitted over IAX2 calls
   as an information element.

func_env.c
------------------
 * Two new functions, DIRNAME and BASENAME, are now
   included which allow users to obtain the directory
   or the base filename of any file.

func_framedrop
------------------
 * New function to selectively drop specified frames
   in either direction on a channel.

func_scramble
------------------
 * Adds an audio scrambler function that may be used to
   distort voice audio on a channel as a privacy
   enhancement.

func_strings
------------------
 * A new STRBETWEEN function is now included which
   allows a substring to be inserted between characters
   in a string. This is particularly useful for transforming
   dial strings, such as adding pauses between digits
   for a string of digits that are sent to another channel.

res_pjproject
------------------
 * In pjproject.conf you can now map pjproject log levels
   to the Asterisk TRACE log level.  The default mappings
   have therefore changed so that only pjproject levels
   3 and 4 are mapped to DEBUG and 5 and 6 are now mapped
   to TRACE.  Previously 3, 4, 5, and 6 were all mapped to
   DEBUG.

res_rtp_asterisk
------------------
 * When the address of the STUN server (stunaddr) is a name resolved via DNS, the
   stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL)
   expires. This allows the STUN server to change its IP address without having to
   reload the res_rtp_asterisk module.

res_tonedetect
------------------
 * Arbitrary tone detection is now available through a
   WaitForTone application (blocking) and a TONE_DETECT
   function (non-blocking).

say.c
------------------
 * Adds SAYFILES function to retrieve the file names that would
   be played by corresponding Say applications, such as
   SayDigits, SayAlpha, etc.

   Additionally adds SayMoney and SayOrdinal applications.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.5.0 to Asterisk 18.6.0 ------------
------------------------------------------------------------------------------

Handle non-standard Meter metric type safely
------------------
 * A meter_support flag has been introduced that defaults to true to maintain current behaviour.
   If disabled, a counter metric type will be used instead wherever a meter metric type was used,
   the counter will have a "_meter" suffix appended to the metric name.

app_dtmfstore
------------------
 * New application which collects digits
   dialed and stores them into
   a specified variable.

app_queue.c
------------------
 * Allow multiple files to be streamed for agent announcement.

chan_pjsip
------------------
 * Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do.

   Add ability to read header by pattern using PJSIP_HEADER().

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.5.0 to Asterisk 18.5.1 ------------
------------------------------------------------------------------------------

New Reload application
------------------
 * Adds an application to reload modules

PlaybackFinished has a new error state
------------------
 * The PlaybackFinished event now has a new state "failed"
   that is used when the sound file was not played due to an error.
   Before the state on PlaybackFinished was always "done".

   In case of multiple sound files to be played,
   the PlaybackFinished is sent only once in the end of the list,
   even in case of error.

WaitForCondition application
------------------
 * This application provides a way to halt
   dialplan execution until a provided
   condition evaluates to true.

app_dial announcement option
------------------
 * The A option for Dial now supports
   playing audio to the caller as well
   as the called party.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.4.0 to Asterisk 18.5.0 ------------
------------------------------------------------------------------------------

AMI Flash event
------------------
 * Hook flash events are now exposed as AMI events.

Add variable support to Originate
------------------
 * The Originate application now allows
   variables to be set on the new channel
   through a new option.

MessageSend
------------------
 * The MessageSend dialplan application now takes an
   optional third argument that can set the message's
   "To" field on outgoing messages.  It's an alternative
   to using the MESSAGE(to) dialplan function.

   To prevent confusion with the first argument, currently
   named "to", it's been renamed to "destination".
   Its function, creating the request URI, hasn't changed.

   The online documentation has also been enhanced to
   explain the behavior.

   Despite the changes in this commit, there should be
   no impact to current users of MessageSend.

New ConfKick application
------------------
 * Adds a ConfKick() application, which allows
   a specific channel, all users, or all non-admin
   users to be kicked from a conference bridge.

app_confbridge answer supervision control
------------------
 * app_confbridge now provides a user option to prevent
   answer supervision if the channel hasn't been
   answered yet. To use it, set a user profile's
   answer_channel option to no.

app_voicemail
------------------
 * You can now customize the "beep" tone or omit it entirely.

func_math: Three new dialplan functions
------------------
 * Introduce three new functions, MIN, MAX, and ABS, which can be used to
   obtain the minimum or maximum of up to two integers or absolute value.

func_volume now can be read
------------------
 * The VOLUME function can now also be used
   to read existing values previously set.

res_pjsip
------------------
 * PJSIP support of registrations of endpoints in multidomain
   scenarios, where the endpoint contains the domain info
   in pjsip.conf.

res_pjsip_dialog_info_body_generator
------------------
 * PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and
   remote elements by iterating through ringing channels and inserting
   that info into NOTIFY packet sent to the endpoint.

res_pjsip_messaging
------------------
 * Implemented the new "to" parameter of the MessageSend()
   dialplan application.  This allows a user to specify
   a complete SIP "To" header separate from the Request URI.
   We now also accept a destination in the same format
   as Dial()...  PJSIP/number@endpoint

res_rtp_asterisk
------------------
 * By default Asterisk reports the PJSIP version in all
   STUN packets it sends.

   This behaviour may not be desired in a production
   environment and can now be disabled by setting the
   stun_software_attribute option to 'no' in rtp.conf.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.3.0 to Asterisk 18.4.0 ------------
------------------------------------------------------------------------------

logger
------------------
 * The dateformat option in logger.conf will now control the remote
   console (asterisk -r -T) timestamp format.  Previously, dateformat only
   controlled the formatting of the timestamp going to log files and the
   main console (asterisk -c) but only for non-verbose messages.

   Internally, Asterisk does not send the logging timestamp with verbose
   messages to console clients. It's up to the Asterisk remote consoles
   to format verbose messages.  Asterisk remote consoles previously did
   not load dateformat from logger.conf.

   Previously there was a non-configurable and hard-coded "%b %e %T"
   dateformat that would be used no matter what on all verbose console
   messages printed on remote consoles.

   Example:
   logger.conf
    dateformat=%F %T.%3q

   # asterisk -rvvv -T
   [2021-03-19 09:54:19.760-0400]  Loading res_stasis_answer.so.
   [Mar 19 09:55:43]     -- Goto (dialExten,s,1)

   Given the following example configuration in logger.conf, Asterisk log
   files and the console, will log verbose messages using the given
   timestamp.  Now ensuring that all remote console messages are logged
   with the same dateformat as other log streams.

   ---
   [general]
   dateformat=%F %T.%3q

   [logfiles]
   console  => notice,warning,error,verbose
   full     => notice,warning,error,debug,verbose
   ---

   Now we have a globally-defined dateformat that will be used
   consistently across the Asterisk main console, remote consoles, and
   log files.

   Now we have consistent logging:

   # asterisk -rvvv -T
   [2021-03-19 09:54:19.760-0400]  Loading res_stasis_answer.so.
   [2021-03-19 09:55:43.920-0400]     -- Goto (dialExten,s,1)

res_pjsip
------------------
 * PJSIP transports can now be partially reloaded safely. This allows the
   local_net and external_* options to be updated without restarting Asterisk.

 * PJSIP endpoints can now be configured to skip authentication when
   handling OPTIONS requests by setting the allow_unauthenticated_options
   configuration property to 'yes.'

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
------------------------------------------------------------------------------

app_mixmonitor
------------------
 * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
   MixMonitorMute when the channel monitoring is started, stopped and muted (or
   unmuted) respectively.

chan_iax2
------------------
 * You can now specify a default "auth" method in the
   [general] section of iax.conf

chan_pjsip, app_transfer
------------------
 * Added TRANSFERSTATUSPROTOCOL variable.  When transfer is performed,
   transfers can pass a protocol specific error code.
   Example, in SIP 3xx-6xx represent any SIP specific error received when
   performing a REFER.

func_odbc
------------------
 * Introduce an ARGC variable for func_odbc functions, along with a minargs
   per-function configuration option.

   minargs enables enforcing of minimum count of arguments to pass to
   func_odbc, so if you're unconditionally using ARG1 through ARG4 then
   this should be set to 4.  func_odbc will generate an error in this case,
   so for example

   [FOO]
   minargs = 4

   and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
   potentially leaked ARG4 from Gosub().

   ARGC is needed if you're using optional argument, to verify whether or
   not an argument has been passed, else it's possible to use a leaked ARGn
   from Gosub (app_stack).  So now you can safely do
   ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.

res_srtp
------------------
 * SRTP replay protection has been added to res_srtp and
   a new configuration option "srtpreplayprotection" has
   been added to the rtp.conf config file.  For security
   reasons, the default setting is "yes".  Buggy clients
   may not handle this correctly which could result in
   no, or one way, audio and Asterisk error messages like
   "replay check failed".

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.1.0 to Asterisk 18.2.0 ------------
------------------------------------------------------------------------------

Core
------------------
 * The location where the media cache stores its temporary files
   is no longer hardcoded to /tmp but can now be configured separately
   via the astcachedir config variable in asterisk.conf. To retain
   backwards compatibility, the default location remains /tmp.

app_voicemail
------------------
 * The VoiceMail application can now be configured to send greetings and
   instructions via early media and only answering the channel when it is
   time for the caller to record their message. This behavior can be
   activated by passing the new 'e' option to VoiceMail.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.0.0 to Asterisk 18.1.0 ------------
------------------------------------------------------------------------------

Core
------------------
 * Added debug logging categories that allow a user to output debug information
   based on a specified category. This lets the user limit, and filter debug
   output to data relevant to a particular context, or topic. For instance the
   following categories are now available for debug logging purposes:

     dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet

   These debug categories can be enable/disable via an Asterisk CLI command:

     core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
     core set debug category off [<category> [<category>] ...]

   If no sub-level is associated all debug statements for a given category are
   output. If a sub-level is given then only those statements assigned a value
   at or below the associated sub-level are output.

app_confbridge
------------------
 * app_confbridge now has the ability to force the estimated bitrate on an SFU
   bridge.  To use it, set a bridge profile's remb_behavior to "force" and
   set remb_estimated_bitrate to a rate in bits per second.  The
   remb_estimated_bitrate parameter is ignored if remb_behavior is something
   other than "force".

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

chan_pjsip
------------------
 * The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and
   returns unsuccessful if it's used on a channel prior to answering.

logger
------------------
 * Added a new log formatter called "plain" that always prints
   file, function and line number if available (even for verbose
   messages) and never prints color control characters.  Most
   suitable for file output but can be used for other channels
   as well.

   You use it in logger.conf like so:
   debug => [plain]debug
   console => [plain]error,warning,debug,notice,pjsip_history
   messages => [plain]warning,error,verbose

------------------------------------------------------------------------------
--- 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 negotiation 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 respective 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
   destination list.  Dialplan can now be simplified by not having to check
   for empty positions in the destination list.

app_voicemail
------------------
 * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
   the Asterisk voicemail directory on startup. Some users that store their
   voicemails on network storage devices experienced slow startup times due to the
   relative expense of traversing the voicemail directory structure looking for
   orphaned lock files. This feature has now been removed.

   Users who require the lock files to be removed at startup should modify their
   startup scripts to do so before starting the asterisk process.

chan_pjsip
------------------
 * A new dialplan function, PJSIP_MOH_PASSTHROUGH, has been added to chan_pjsip. This
   allows the behaviour of the moh_passthrough endpoint option to be read or changed
   in the dialplan. This allows control on a per-call basis.

chan_rtp
------------------
 * The UnicastRTP channel driver provided by chan_rtp now accepts
   "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
   The first AAAA (preferred) or A record resolved will be used as the destination.
   The lookup is synchronous so beware of possible dialplan delays if you specify a
   hostname.

func_curl
------------------
 * A new parameter, httpheader, has been added to CURLOPT function. This parameter
   allows to set custom http headers for subsequent calls off CURL function.
   Any setting of headers will replace the default curl headers
   (e.g. "Content-type: application/x-www-form-urlencoded")

 * A new option, followlocation, can now be enabled with the CURLOPT()
   dialplan function. Setting this will instruct cURL to follow 3xx
   redirects, which it does not by default.

func_jitterbuffer
------------------
 * The JITTERBUFFER dialplan function now has an option to enable video synchronization
   support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
   the video is buffered according to the size of the audio jitterbuffer and is
   synchronized to the audio.

func_volume
------------------
 * Accept decimal number as argument.

http
------------------
 * You can now disable the /httpstatus page served by Asterisk's built-in
   HTTP server by setting 'enable_status' to 'no' in http.conf.

minmemfree
------------------
 * The 'minmemfree' configuration option now counts memory allocated to
   the filesystem cache as "free" because it is memory that is available
   to the process.

res_ari_channels
------------------
 * When creating a channel in ARI using the create call
   you can now specify dialplan variables to be set as part
   of the same operation.

res_musiconhold
------------------
 * This fix allows a realtime moh class to be unregistered from the command
   line. This is useful when the contents of a directory referenced by a
   realtime moh class have changed.
   The realtime moh class is then reloaded on the next request and uses the
   new directory contents.

 * A new mode - playlist - has been added to res_musiconhold. This mode allows the
   user to specify the files (or URLs) to play explicitly by putting them directly
   in musiconhold.conf.

res_pjsip
------------------
 * Added a new PJSIP system setting called disable_rport.
   Default is no to keep support working as before.

   If it is false (default) it adds the 'rport' parameter in the outgoing request message.
   If it is true it does not add the 'rport' parameter in the outgoing request message.

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

res_pjsip_endpoint_identifier_ip
------------------
 * In 'type = identify' sections, the addresses specified for the 'match'
   clause can now include a port number. For IP addresses, the port is
   provided by including a colon after the address, followed by the
   desired port number. If supplied, the netmask should follow the port
   number. To specify a port for IPv6 addresses, the address itself must
   be enclosed in brackets to be parsed correctly.

res_pjsip_logger
------------------
 * The PJSIP packet logger now has the following CLI commands:

   pjsip set logger pcap <filename>

   When used this will create a pcap file containing the incoming
   and outgoing SIP packets, in unencrypted form.

   pjsip set logger console <on / off>

   This allows you to toggle logging to console on and off.

   pjsip set logger host <IP/subnet mask> add

   This allows you to add an additional IP address or subnet
   mask to logging, allowing you to log multiple instead of
   just a single IP address or all traffic.

   The normal "pjsip set logger host" CLI command has also been
   expanded to allow subnet masks as well.

res_pjsip_session
------------------
 * When placing an outgoing call to a PJSIP endpoint the intent
   of any requested formats will now be respected. If only an audio
   format is requested (such as ulaw) but the underlying endpoint
   does not support the format the resulting SDP will still only
   contain an audio stream, and not any additional streams such as
   video.

 * 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.

res_rtp_asterisk
------------------
 * This change include a new cli command 'rtp show settings'

   The command display by general settings of rtp configuration. For this
   point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum,
   strictrtp, learning_min_sequential and icesupport.

 * The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
   an ACL mechanism.

   As such six now options are now available:

   ice_deny
   ice_permit
   ice_acl
   stun_deny
   stun_permit
   stun_acl

   These options have their obvious meanings as used elsewhere.

   Backwards compatibility was maintained by adding {stun,ice}_blacklist as
   aliases for {stun,ice}_deny.

res_sorcery_memory_cache
------------------