The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.2.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.2.0-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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28173 - Deadlock in chan_sip handling subscribe
request during res_parking reload
(Reported by Giuseppe
Sucameli)
* ASTERISK-28104 - AstriCon Feedback: Automatically create a 1
line dialplan context for stasis apps
(Reported by George
Joseph)
* ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will
not compile
(Reported by David Wilcox)
* ASTERISK-28238 - PJSIP realtime. getcontext not working with
DUNDI
(Reported by Ray)
* ASTERISK-28263 - codec_opus: errors setting max_playback_rate
and bitrate to "sdp"
(Reported by Gianluca Merlo)
* ASTERISK-28250 - build: Cross-compilation fails for target
arm-linux-gnueabihf
(Reported by Jean Aunis - Prescom)
* ASTERISK-28257 - res_http_websocket: PING / PONG opcodes
break data reception
(Reported by Jeremy Lainé)
* ASTERISK-28252 - HangupHandler manager events are never
thrown
(Reported by Gerald Schnabel)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28249 - res_monitor: Segfault with
Monitor(wav,file,i)
(Reported by Valentin VidiÄ)
* ASTERISK-28244 - stasis: Filter messages at publishing to
AMI/ARI
(Reported by Joshua C. Colp)
* ASTERISK-28231 - res_http_websocket: Not responding to
Connection Close Frame (opcode 8)
(Reported by Jeremy
Lainé)
* ASTERISK-28197 - stasis: ast_endpoint struct holds the
channel_ids of channels past destruction in certain cases
(Reported by Mohit Dhiman)
* ASTERISK-28232 - core: RAII using clang use-after-scope
issue
(Reported by Diederik de Groot)
* ASTERISK-28230 - res_rtp_asterisk: abs-send-time extension
added with Asterisk 15.5.0 breaks GXV3140 video telephony
(Reported by David Kuehling)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on RTP renegotiation
(Reported by
Alexei Gradinari)
* ASTERISK-28225 - app_voicemail: Channel variable
VM_MESSAGEFILE not updated correctly if message marked "urgent"
(Reported by boatright)
* ASTERISK-28218 - app_queue: Asterisk crashes when using Queue
with a pre-dial handler (option b)
(Reported by Mark)
* ASTERISK-28212 - stasis: Statistics broke ABI under developer
mode
(Reported by Joshua C. Colp)
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
* ASTERISK-28221 - Bug in ast_coredumper
(Reported by
Andrew Nagy)
* ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes
doesn't trigger NOTIFYs
(Reported by George Joseph)
* ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp
negotiation problem
(Reported by David Kuehling)
* ASTERISK-28201 - [patch] confbridge: no announce to the
marked users when they join an empty conference
(Reported
by Alexei Gradinari)
* ASTERISK-28117 - stasis: Add statistics for usage when in
developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28186 - stasis: Filter messages at publishing based
on to_* presence
(Reported by Joshua C. Colp)
* ASTERISK-28194 - chan_sip: Leak using contact ACL
(Reported by Giuseppe Sucameli)
* ASTERISK-27095 - chan_pjsip: When connected_line_method is
set to invite, we're not trying UPDATE
(Reported by George
Joseph)
* ASTERISK-28182 - chan_pjsip: When connected_line_method is
set to invite, asterisk is not trying UPDATE
(Reported by
nappsoft)
* ASTERISK-28157 - Asterisk crashes when the res_pjsip_*
modules unload
(Reported by sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-28246 - Support skipping on the g726 format
(Reported by Eyal Hasson)
* ASTERISK-28196 - bridge_softmix: Does not support WebRTC
source with multi video tracks.
(Reported by Xiemin Chen)
* ASTERISK-28198 - res_ari: Add new hangup causes for ARI
Channel DELETE command
(Reported by Sebastian Damm)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0-rc1
Thank you for your continued support of Asterisk!
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