[asterisk-users] Asterisk 20.2.0 Now Available

[Asterisk Development Team]

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

The release of Asterisk 20.2.0 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:

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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0

Thank you for your continued support of Asterisk!
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