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 20.0.0 to Asterisk 20.1.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * The AOCMessage action can now be used to generate AOC-S messages.

Add support for named capture agent.
------------------
 * A name for the capture agent can now be specified
   using the capture_name option which, if specified,
   will be sent to the HEP server.

app_if
------------------
 * Adds the If, ElseIf, Else, EndIf, and ExitIf applications
   for conditional execution of a block of code.

app_mixmonitor
------------------
 * The d option for MixMonitor now allows deleting
   the original recording when MixMonitor exits,
   which can be useful when MixMonitor copies it
   somewhere else before exiting.

 * Adds the c option to use the real Caller ID on
   the channel in voicemail recordings as opposed
   to the Connected Line.

app_voicemail
------------------
 * The voicemail user option attachextrecs can
   now be set to control whether external recordings
   trigger voicemail email notifications.

cdr
------------------
 * Two new options have been added which allow
   bridging and dial state changes to be ignored
   in CDRs, which can be useful if a single CDR
   is desired for a channel.

chan_dahdi
------------------
 * FXO channels (FXS signaled) that don't use callerid or
   distinctive ring detection can now be configured
   to enter the dialplan immediately using immediate=yes,
   instead of waiting for at least one ring.

pbx_builtins
------------------
 * It is now possible to not wait for media on
   a channel when answering it using Answer,
   by specifying the i option.

res_pjsip
------------------
 * Added options "security_negotiation" and "security_mechanisms" to pjsip
   endpoints and registrations. "security_negotiation" can be set to "no" (default)
   or "mediasec", and "security_mechanisms" can be a list of comma-separated
   security_mechanisms in the form defined by RFC 3329 section 2.2.

 * A new option named "all_codecs_on_empty_reinvite" has been added to the
   global section. When this option is enabled, on reception of a re-INVITE
   without SDP, Asterisk will send an SDP offer in the 200 OK response containing
   all configured codecs on the endpoint, instead of simply those that have
   already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
   The default value is "off".

res_pjsip_aoc
------------------
 * Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
   A new endpoint option, send_aoc, controls this.

res_pjsip_header_funcs
------------------
 * The new PJSIP_HEADER_PARAM function now fully supports both
   URI and header parameters. Both reading and writing
   parameters are supported.

res_pjsip_logger
------------------
 * SIP messages can now be filtered by SIP request method
   (INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
   SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
   allowing for more granular debugging to be done
   in the CLI. This applies to requests but not responses.

res_pjsip_notify
------------------
 * Allows using the config options in pjsip_notify.conf
   from AMI actions as with the existing CLI commands.

res_tonedetect
------------------
 * The TONE_DETECT function now supports
   detection of audible ringback tone
   using the p option.

xmldocs
------------------
 * The XML documentation can now be reloaded without restarting
   Asterisk, which makes it possible to load new modules that
   enforce documentation without restarting Asterisk.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------

New EXPORT function
------------------
 * A new function, EXPORT, allows writing variables
   and functions on other channels, the complement
   of the IMPORT function.

app_amd
------------------
 * An audio file to play during AMD processing can
   now be specified to the AMD application or configured
   in the amd.conf configuration file.

app_bridgewait
------------------
 * Adds the n option to not answer the channel when
   the BridgeWait application is called.

features
------------------
 * The Bridge application now has the n "no answer" option
   that can be used to prevent the channel from being
   automatically answered prior to bridging.

func_strings
------------------
 * Three new functions, TRIM, LTRIM, and RTRIM, are
   now available for trimming leading and trailing
   whitespace.

res_pjsip
------------------
 * A new option named "peer_supported" has been added to the endpoint option
   100rel. When set to this option, Asterisk sends provisional responses
   reliably if the peer supports it. If the peer does not support reliable
   provisional responses, Asterisk sends them normally.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------

Transfer feature
------------------
 * The following capabilities have been added to the
   transfer feature:

   - The transfer initiation announcement prompt can
   now be customized in features.conf.

   - The TRANSFER_EXTEN variable now can be set on the
   transferer's channel in order to allow the transfer
   function to automatically attempt to go to the extension
   contained in this variable, if it exists. The transfer
   context behavior is not changed (TRANSFER_CONTEXT is used
   if it exists; otherwise the default context is used).

app_confbridge
------------------
 * Adds the end_marked_any option which can be used
   to kick users from a conference after any
   marked user leaves (including marked users).

db
------------------
 * The DBPrefixGet AMI action now allows retrieving
   all of the DB keys beginning with a particular
   prefix.

locks
------------------
 * A new AMI event, DeadlockStart, is now available
   when Asterisk is compiled with DETECT_DEADLOCKS,
   and can indicate that a deadlock has occured.

res_geolocation
------------------
 * * Added processing for the 'confidence' element.