The Asterisk Development Team would like to announce the first release candidate of Asterisk 16.9.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.9.0-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-28766 - PJSIP blind transfer not completed after using Proceeding() (Reported by lvl) * ASTERISK-28685 - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28742 - res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup (Reported by Kevin Harwell) * ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting to SLIN (Reported by Ross Beer) * ASTERISK-28730 - res_pjsip_session: Fix out of order session refreshes (Reported by Joshua C. Colp) * ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are depleted, should return 503 (Reported by Walter Doekes) * ASTERISK-28719 - Cannot remove defaultrule from queue using realtime queues (Reported by EDV O-TON) * ASTERISK-28713 - res_stasis_playback: Error building JSON (Reported by Sébastien Duthil) * ASTERISK-28714 - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) * ASTERISK-26082 - res_pjsip_messaging: MessageSend Content-Type can't be changed (Reported by Alex) * ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer) * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) * ASTERISK-26955 - pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected (Reported by Peter Sokolov) Improvements made in this release: ----------------------------------- * ASTERISK-28750 - TLS/SSL Key too small error (Reported by Martin Zeh) * ASTERISK-28733 - stream: Add support for adding/removing streams during SFU/calls (Reported by Joshua C. Colp) * ASTERISK-24798 - Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor (Reported by xrobau) * ASTERISK-28726 - install_prereq script uses the interactive mode when installing aptitude (Reported by Sylvain Afchain) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0-rc1 Thank you for your continued support of Asterisk!
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