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