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:
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Deadlock in chan_sip handling subscribe request during res_parking reload (Reported by Giuseppe Sucameli) |
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AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps (Reported by George Joseph) |
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Opensuse Leap 15 –with-jannson-bundled will not compile (Reported by David Wilcox) |
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PJSIP realtime. getcontext not working with DUNDI (Reported by Ray) |
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codec_opus: errors setting max_playback_rate and bitrate to “sdp” (Reported by Gianluca Merlo) |
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build: Cross-compilation fails for target arm-linux-gnueabihf (Reported by Jean Aunis – Prescom) |
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res_http_websocket: PING / PONG opcodes break data reception (Reported by Jeremy Lainé) |
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HangupHandler manager events are never thrown (Reported by Gerald Schnabel) |
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res_pjsip: Threads pile up needlessly when AOR is blocked (Reported by Ross Beer) |
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res_monitor: Segfault with Monitor(wav,file,i) (Reported by Valentin Vidić) |
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stasis: Filter messages at publishing to AMI/ARI (Reported by Joshua C. Colp) |
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res_http_websocket: Not responding to Connection Close Frame (opcode 8) (Reported by Jeremy Lainé) |
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stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases (Reported by Mohit Dhiman) |
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core: RAII using clang use-after-scope issue (Reported by Diederik de Groot) |
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res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by David Kuehling) |
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need to reset DTMF last sequence number and timestamp on RTP renegotiation (Reported by Alexei Gradinari) |
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app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked “urgent” (Reported by boatright) |
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app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b) (Reported by Mark) |
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stasis: Statistics broke ABI under developer mode (Reported by Joshua C. Colp) |
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Regression: MWI polling no longer works (Reported by abelbeck) |
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Bug in ast_coredumper (Reported by Andrew Nagy) |
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app_voicemail: Leaving voicemail sometimes doesn’t trigger NOTIFYs (Reported by George Joseph) |
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Asterisk 15.4.1 h264 fmtp negotiation problem (Reported by David Kuehling) |
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confbridge: no announce to the marked users when they join an empty conference (Reported by Alexei Gradinari) |
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stasis: Add statistics for usage when in developer mode (Reported by Joshua C. Colp) |
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stasis: Filter messages at publishing based on to_* presence (Reported by Joshua C. Colp) |
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chan_sip: Leak using contact ACL (Reported by Giuseppe Sucameli) |
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chan_pjsip: When connected_line_method is set to invite, we’re not trying UPDATE (Reported by George Joseph) |
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chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE (Reported by nappsoft) |
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Asterisk crashes when the res_pjsip_* modules unload (Reported by sungtae kim) |
Improvements made in this release:
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Support skipping on the g726 format (Reported by Eyal Hasson) |
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bridge_softmix: Does not support WebRTC source with multi video tracks. (Reported by Xiemin Chen) |
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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!