asterisk announce - Oct 2023 - asterisk release 20.5.0

The Asterisk Development Team would like to announce  
the release of asterisk-20.5.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.5.0
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 asterisk-20.5.0
=======================================
Links:
----------------------------------------

 - [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)  
 -
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

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

- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
particular mailbox folder. The forward command can be used to copy a message
within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using
`rtp->themssrc_valid` into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.

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

- ### sig_analog: Add Called Subscriber Held capability.
  Called Subscriber Held is now supported for analog
  FXS channels, using the calledsubscriberheld option. This allows
  a station  user to go on hook when receiving an incoming call
  and resume from another phone on the same line by going on hook,
  without disconnecting the call.

- ### res_pjsip_header_funcs: Make prefix argument optional.
  The prefix argument to PJSIP_HEADERS is now
  optional. If not specified, all header names will be
  returned.

- ### core/ari/pjsip: Add refer mechanism
  There is a new ARI endpoint `/endpoints/refer` for referring
  an endpoint to some URI or endpoint.

- ### chan_dahdi: Allow autoreoriginating after hangup.
  The autoreoriginate setting now allows for kewlstart FXS
  channels to automatically reoriginate and provide dial tone to the
  user again after all calls on the line have cleared. This saves users
  from having to manually hang up and pick up the receiver again before
  making another call.

- ### sig_analog: Allow three-way flash to time out to silence.
  The threewaysilenthold option now allows the three-way
  dial tone to time out to silence, rather than continuing forever.

- ### res_pjsip: Enable TLS v1.3 if present.
  res_pjsip now allows TLS v1.3 to be enabled if supported by
  the underlying PJSIP library. The bundled version of PJSIP supports
  TLS v1.3.

- ### app_queue: Add support for applying caller priority change immediately.
  The 'queue priority caller' CLI command and
  'QueueChangePriorityCaller' AMI action now have an 'immediate'
  argument which allows the caller priority change to be reflected
  immediately, causing the position of a caller to move within the
  queue depending on the priorities of the other callers.

- ### Adds manager actions to allow move/remove/forward individual messages in a
particular mailbox folder. The forward command can be used to copy a message
within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
required to retrieve message ID's.
  The following manager actions have been added
  VoicemailBoxSummary - Generate message list for a given mailbox
  VoicemailRemove - Remove a message from a mailbox folder
  VoicemailMove - Move a message from one folder to another within a mailbox
  VoicemailForward - Copy a message from one folder in one mailbox
  to another folder in another or the same mailbox.

- ### app_voicemail: add CLI commands for message manipulation
  The following CLI commands have been added to app_voicemail
  voicemail show mailbox <mailbox> <context>
  Show contents of mailbox <mailbox>@<context>
  voicemail remove <mailbox> <context> <from_folder>
<messageid>
  Remove message <messageid> from <from_folder> in mailbox
<mailbox>@<context>
  voicemail move <mailbox> <context> <from_folder>
<messageid> <to_folder>
  Move message <messageid> in mailbox <mailbox>&<context>
from <from_folder> to <to_folder>
  voicemail forward <from_mailbox> <from_context>
<from_folder> <messageid> <to_mailbox> <to_context>
<to_folder>
  Forward message <messageid> in mailbox <mailbox>@<context>
<from_folder> to
  mailbox <mailbox>@<context> <to_folder>

- ### sig_analog: Allow immediate fake ring to be suppressed.
  The immediatering option can now be set to no to suppress
  the fake audible ringback provided when immediate=yes on FXS channels.


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


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

  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages
on aarch64 Debian platforms
  - #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints
to some resource
  - #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
immediate=yes
  - #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
  - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
  - #181: [improvement]: app_voicemail - add manager actions to display and
manipulate messages
  - #202: [improvement]: app_queue: Add support for immediately applying queue
caller priority change
  - #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
  - #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
  - #226: [improvement]: Apply contact_user to incoming calls
  - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
  - #233: [bug]: Deadlock with MixMonitorMute AMI action
  - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
  - #253: app_gosub patch appear to have broken predial handlers that utilize
macros that call gosubs
  - #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
registered"
  - #263: [bug]: download_externals doesn't always handle versions correctly
  - #265: [bug]: app_macro isn't locking around channel datastore access
  - #267: [bug]: ari: refer with display_name key in request body leads to crash
  - #274: [bug]: Syntax Error in SQL Code
  - #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-Wdeclaration-after-statement'
  - #277: [bug]: pbx.c: Compiler error with gcc 12.2
  - #281: [bug]: app_dial: Infinite loop if called channel hangs up while
receiving digits