[asterisk-announce] Asterisk Release 20.3.0

[Asterisk Development Team]

The Asterisk Development Team would like to announce  
the release of Asterisk 20.3.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.3.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release 20.3.0
========================================

Summary:
----------------------------------------

- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.

User Notes:
----------------------------------------

- ### cel: add local optimization begin event
  The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
  by itself or in conert with the existing
  AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

- ### chan_dahdi: Add dialmode option for FXS lines.
  A "dialmode" option has been added which allows
  specifying, on a per-channel basis, what methods of
  subscriber dialing (pulse and/or tone) are permitted.
  Additionally, this can be changed on a channel
  at any point during a call using the CHANNEL
  function.

- ### pbx_dundi: Add PJSIP support.
  DUNDi now supports chan_pjsip. Outgoing calls using
  PJSIP require the pjsip_outgoing_endpoint option
  to be set in dundi.conf.

- ### cli: increase channel column width
  This change increases the display width on 'core show channels'
  amd 'core show channels verbose'
  For 'core show channels', the Channel name field is increased to
  64 characters and the Location name field is increased to 32
  characters.
  For 'core show channels verbose', the Channel name field is
  increased to 80 characters, the Context is increased to 24
  characters and the Extension is increased to 24 characters.

- ### app_senddtmf: Add SendFlash AMI action.
  The SendFlash AMI action now allows sending
  a hook flash event on a channel.

- ### res_http_media_cache: Introduce options and customize
  The res_http_media_cache module now attempts to load
  configuration from the res_http_media_cache.conf file.
  The following options were added:
    * timeout_secs
    * user_agent
    * follow_location
    * max_redirects
    * protocols
    * redirect_protocols
    * dns_cache_timeout_secs

- ### test.c: Fix counting of tests and add 2 new tests
  The "tests" attribute of the "testsuite" element in the
  output XML now reflects only the tests actually requested
  to be executed instead of all the tests registered.
  The "failures" attribute was added to the "testsuite"
  element.
  Also added two new unit tests that just pass and fail
  to be used for testing CI itself.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMonitorID when calling
  the manager action: MixMonitorMute.  This will allow an
  individual MixMonitor instance to be muted via ID.
  The MixMonitorID can be stored as a channel variable using
  the 'i' MixMonitor option and is returned upon creation if
  this option is used.
  As part of this change, if no MixMonitorID is specified in
  the manager action MixMonitorMute, Asterisk will set the mute
  flag on all MixMonitor audiohooks on the channel.  Previous
  behavior would set the flag on the first MixMonitor audiohook
  found.

- ### bridge_builtin_features: add beep via touch variable
  Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
  interval in seconds will result in a periodic beep being
  played to the monitored channel upon MixMontior/Monitor
  feature start.
  If an interval less than 5 seconds is specified, the interval
  will default to 5 seconds.  If the value is set to an invalid
  interval, the default of 15 seconds will be used.

- ### format_sln: add .slin as supported file extension
  format_sln now recognizes '.slin' as a valid
  file extension in addition to the existing
  '.sln' and '.raw'.


Upgrade Notes:
----------------------------------------

- ### cel: add local optimization begin event
  The existing AST_CEL_LOCAL_OPTIMIZE can continue
  to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
  can be ignored if desired.


Closed Issues:
----------------------------------------

  - #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
  - #39: [Bug]: Remove .gitreview from repository.
  - #43: [Bug]: Link to trademark policy is no longer correct
  - #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
  - #52: [improvement]: Add local optimization begin cel event

### For more details, see:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md