[asterisk-dev] Asterisk 20.2.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 20.2.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.2.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

New Features made in this release:
-----------------------------------
 * ASTERISK-29810 - app_signal: Add channel signaling
      applications
      (Reported by N A)
 * ASTERISK-30262 - res_pjsip_session: Allow a context to be
      specified for overlap dialing
      (Reported by N A)
 * ASTERISK-30319 - Add BYE Reason support for SIP
     
      (Reported by Igor Goncharovsky)
 * ASTERISK-30180 - app_broadcast: Add a channel audio
      multicasting application
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string
   
      (Reported by AvayaXAsterisk)
 * ASTERISK-30354 - chan_iax2: Lack of formats prior to
      receiving voice frames causes jitterbuffer to stall
     
      (Reported by N A)
 * ASTERISK-30162 - when chan_iax is used to relay calls, no
      ringing indication is played
      (Reported by Jaco Kroon)
 * ASTERISK-30424 - pjproject_bundled: cross-compilation broken
      when ssl autodetected
      (Reported by Nick French)
 * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when
      multi-homed
      (Reported by cmaj)
 * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP
      2.13
      (Reported by Ross Beer)
 * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember
     
      (Reported by Sean Bright)
 * ASTERISK-29604 - ari: Segfault with lots of calls
     
      (Reported by Danila Evgrafov)
 * ASTERISK-30406 - pbx_ael: Global variables are not expanded.

      (Reported by Sean Bright)
 * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding
      g722 after MES changes
      (Reported by George Joseph)
 * ASTERISK-30345 - loader.c: Modules that decline to load
      cannot be reloaded
      (Reported by N A)
 * ASTERISK-30351 - manager: Originate variables are not added
      when setvar used in manager.conf
      (Reported by Sebastian
      Gutierrez)
 * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
      when they shouldn't be
      (Reported by Joshua C. Colp)
 * ASTERISK-30379 - http: fix NULL pointer dereference while
      enable_status on TLS-only
      (Reported by Boris P. Korzun)
 * ASTERISK-30375 - res_http_media_cache: Crash when URL has no
      path component.
      (Reported by Sean Bright)
 * ASTERISK-30367 - pbx: Fix outdated channel snapshots with
      pbx_exec
      (Reported by N A)
 * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking
      for extension, callerid supplement executed too late
     
      (Reported by Oleg)
 * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not
      used when moh_passthrough has call on hold
      (Reported by
      Benjamin Keith Ford)
 * ASTERISK-30240 - app voicemail odbc build error with gcc
      11.1
      (Reported by Michael Bradeen)
 * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to
      endpoint
      (Reported by Yury Kirsanov)
 * ASTERISK-30198 - Error `Too many open files` occurs after
      about ~8000 calls when using mixmonitor
      (Reported by
      Julien Alie)
 * ASTERISK-30347 - xmldocs: Remove references to removed
      applications
      (Reported by N A)

Improvements made in this release:
-----------------------------------
 * ASTERISK-30411 - app_read: add option to include terminating
      digit on empty, terminated strings
      (Reported by Michael
      Bradeen)
 * ASTERISK-30405 - app_directory: Add 's' option to skip
      channel call
      (Reported by Michael Bradeen)
 * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to
      answer
      (Reported by Michael Bradeen)
 * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13
 
      (Reported by Stanislav Abramenkov)
 * ASTERISK-30404 - app_directory: Add reading directory
      configuration from custom file
      (Reported by Michael
      Bradeen)
 * ASTERISK-29913 - func_json: Adds multi-level and array
      parsing to JSON_DECODE
      (Reported by N A)
 * ASTERISK-30353 - func_frame_trace: Print text for text
      frames
      (Reported by N A)
 * ASTERISK-30361 - json.h: Add missing
      ast_json_object_real_get
      (Reported by N A)
 * ASTERISK-30280 - Create capability to assign a Media
      Experience Score to RTP streams
      (Reported by George
      Joseph)
 * ASTERISK-30332 - func_callerid: Warn if invalid redirecting
      reason provided
      (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0-rc1

Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20230302/7ca7007f/attachment-0001.html>