The Asterisk Development Team would like to announce the first release candidate of Asterisk 16.15.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.15.0-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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-29057 - pjsip: Crash on call rejection during high load (Reported by Sandro Gauci) New Features made in this release: ----------------------------------- * ASTERISK-29027 - Implement support for History-Info (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-28933 - res_pjsip.so fails to load when bundled pjproject is compiled without libssl (Reported by Walter Doekes) * ASTERISK-28825 - Any curl response checks out as valid even if 404 is returned. (Reported by dovid) * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies (Reported by Sebastian Damm) * ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed includes (Reported by Michael Newton) * ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander Traud) * ASTERISK-29146 - GCC Warnings: â%sâ directive argument is null. (Reported by Alexander Traud) * ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander Traud) * ASTERISK-29136 - config: Sample features.conf incorrectly includes " around sound files (Reported by Benjamin M.) * ASTERISK-29123 - logger.conf.sample missing comment mark on line 115 (Reported by Andrew Siplas) * ASTERISK-29108 - resource_endpoints.c : Memory leak if endpoint not found (Reported by Jean Aunis - Prescom) * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF (Reported by under) * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) * ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a single entry (Reported by lvl) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used (Reported by Sebastian Damm) * ASTERISK-29091 - Crash when ast_translator_build_path fails (Reported by Jasper van der Neut) * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format (Reported by å¨å®¶å»º) * ASTERISK-29085 - func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT (Reported by Péter Juhász) * ASTERISK-24329 - Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) * ASTERISK-29089 - RTP Ports not cleared after hangup (Reported by Ross Beer) * ASTERISK-29081 - res_stasis: Add compare function for bridges moh container (Reported by Hajek Michal) * ASTERISK-28416 - Unable to get rtp codec payload code for slin (Reported by Brian J. Murrell) * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions aren't handled correctly (Reported by George Joseph) Improvements made in this release: ----------------------------------- * ASTERISK-29054 - Logger: Add debug logging categories (Reported by Kevin Harwell) * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) * ASTERISK-29056 - Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0-rc1 Thank you for your continued support of Asterisk!
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