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!
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