[asterisk-dev] Asterisk Release 20.4.0-rc1

Asterisk Development Team asteriskteamsa at sangoma.com
Fri Jun 30 10:46:45 CDT 2023


The Asterisk Development Team would like to announce  
release candidate 1 of Asterisk 20.4.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.4.0-rc1
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 20.4.0-rc1
========================================

Links:
----------------------------------------

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0-rc1.md)  
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.4.0-rc1)  
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0-rc1.tar.gz)  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

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

- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.

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

- ### AMI: Add parking position parameter to Park action
  New ParkingSpace parameter has been added to AMI action Park.

- ### res_musiconhold: Add option to loop last file.
  The loop_last option in musiconhold.conf now
  allows the last file in the directory to be looped once reached.

- ### AMI: Add CoreShowChannelMap action.
  New AMI action CoreShowChannelMap has been added.

- ### sig_analog: Add fuller Caller ID support.
  Additional Caller ID properties are now supported on
  incoming calls to FXS stations, namely the
  redirecting reason and call qualifier.

- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
  When creating a bridge using the ARI the 'type' argument now
  accepts a new value 'sdp_label' which will configure the bridge to add
  labels for each stream in the SDP with the corresponding channel id.

- ### app_queue: Preserve reason for realtime queues
  Make paused reason in realtime queues persist an
  Asterisk restart. This was fixed for non-realtime
  queues in ASTERISK_25732.

- ### 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.

- ### 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'.

- ### 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.

- ### 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

- ### 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.

- ### 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.

- ### 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.

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


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

- ### app_queue: Preserve reason for realtime queues
  Add a new column to the queue_member table:
  reason_paused VARCHAR(80) so the reason can be preserved.


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

  - #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
  - #55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
  - #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
  - #65: [bug]: heap overflow by default at startup
  - #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
  - #73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
  - #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
  - #89: [improvement]:  indications: logging changes
  - #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
  - #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
  - #96: [bug]: make install-logrotate causes logrotate to fail on service restart
  - #98: [new-feature]: callerid: Allow timezone to be specified at runtime
  - #100: [bug]: sig_analog: hidecallerid setting is broken
  - #102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect.
  - #104: [improvement]: Add AMI action to get a list of connected channels
  - #108: [new-feature]: fair handling of calls in multi-queue scenarios
  - #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
  - #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
  - #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
  - #122: [new-feature]: res_musiconhold: Add looplast option
  - #133: [bug]: unlock channel after moh state access
  - #136: [bug]: Makefile downloader does not follow redirects.
  - #145: [bug]: ABI issue with pjproject and pjsip_inv_session
  - #155: [bug]: GCC 13 is catching a few new trivial issues
  - #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
  - #174: [bug]: app_voicemail imap compile errors



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