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* ASTERISK-30367 - pbx: Fix outdated channel snapshots with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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