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 Vidić) * 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!