The Asterisk Development Team would like to announce the first release candidate of Asterisk 17.6.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.6.0-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-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 (Reported by Jared Smith) * ASTERISK-28957 - chan_sip: chan_sip does not process 400 response to an INVITE. (Reported by Frederic LE FOLL) * ASTERISK-28888 - res_corosync: causes asterisk crash in huge distributed environment. (Reported by Università di Bologna - CESIA VoIP) * ASTERISK-28955 - "setvar" doesn't work properly in dahdi-channels.conf (Reported by Marin Odrljin) * ASTERISK-28954 - StreamEcho() only returns 1 active stream (Reported by Bill Kervaski) * ASTERISK-28953 - res_pjsip_session: Preserve stream label (Reported by Joshua C. Colp) * ASTERISK-28942 - res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching (Reported by Joshua C. Colp) * ASTERISK-28952 - Queue wrapuptime sometimes not respected (based on stale lastcall time) (Reported by Walter Doekes) * ASTERISK-28950 - Stale code in app_queue to check untouched channel (Reported by Walter Doekes) * ASTERISK-28644 - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) * ASTERISK-28938 - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28948 - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) * ASTERISK-28940 - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) * ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri (Reported by Walter Doekes) * ASTERISK-28900 - res_fax: Double frame free when gateway in use with off-nominal format usage (Reported by Gregory Massel) * ASTERISK-28929 - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) * ASTERISK-28932 - res_pjsip_logger writing too big packets (Reported by nappsoft) * ASTERISK-28920 - bridge show all causes crash (Reported by sungtae kim) * ASTERISK-28921 - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) Improvements made in this release: ----------------------------------- * ASTERISK-28959 - res_pjsip: Added option for disable rport parameter set (Reported by sungtae kim) * ASTERISK-28958 - Continue reading string when ping received by websocket (Reported by Nickolay V. Shmyrev) * ASTERISK-28945 - AMI SendText - add Content-Type parameter (Reported by Kevin Harwell) * ASTERISK-28949 - res_http_websocket: Add masking to websocket client (Reported by Moises Silva) * ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10 (Reported by Kevin Harwell) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0-rc1 Thank you for your continued support of Asterisk!
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