The Asterisk Development Team has announced the first release candidate of Asterisk 12.6.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.6.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release candidate: Bugs fixed in this release: ----------------------------------- * ASTERISK-24027 - MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up (Reported by Matt Jordan) * ASTERISK-24236 - res_hep_rtcp: Module incorrectly depends on pjsip (Reported by Matt Jordan) * ASTERISK-24032 - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined (Reported by Kilburn) * ASTERISK-24225 - Dial option z is broken (Reported by dimitripietro) * ASTERISK-24234 - app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg() (Reported by Shaun Ruffell) * ASTERISK-24043 - ARI /continue fails to actually continue into the dialplan (Reported by Krandon Bruse) * ASTERISK-24245 - gcc 4.1.2 complains of files that do not end with newlines (Reported by Shaun Ruffell) * ASTERISK-24229 - ARI: playback of sounds implicitly answers channel, preventing early media playback (Reported by Matt Jordan) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks (Reported by Walter Doekes) * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname (Reported by Private Name) * ASTERISK-24147 - ARI: channel hangup crashes asterisk process (Reported by Edvin Vidmar) * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer (Reported by Badalian Vyacheslav) * ASTERISK-24143 - pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK (Reported by Aleksei Kulakov) * ASTERISK-24019 - When a Music On Hold stream starts it restarts at beginning of file. (Reported by Jason Richards) * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying if ever not able to resolve (Reported by David Herselman) * ASTERISK-24264 - ARI: Adding a channel to a holding bridge automatically starts MOH (Reported by Samuel Galarneau) * ASTERISK-24212 - testsuite: Sporadic crash due to assert on stopping RTP engine (Reported by Matt Jordan) * ASTERISK-24241 - crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack (Reported by Deepak Singh Rawat) * ASTERISK-24254 - CDRs: Application/args/dialplan CEP updated during dial operation (Reported by Matt Jordan) * ASTERISK-24231 - crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable (Reported by Niklas Larsson) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) * ASTERISK-23577 - res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by Jay Jideliov) * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls (Reported by Roman Skvirsky) * ASTERISK-24161 - PJSIPShowEndpoint gives inaccurate count of list items (Reported by Mark Michelson) * ASTERISK-24331 - Unexpected Errors in Asterisk Manager Interface Output (Reported by xrobau) * ASTERISK-24136 - Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type (Reported by Mark Michelson) * ASTERISK-24301 - Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk (Reported by Matt Jordan) * ASTERISK-24290 - Endpoint identifier match value fails to parse when CIDR network format is specified (Reported by Ray Crumrine) * ASTERISK-24237 - CDR: FRACK With PJSIP blonde transfer. (Reported by Richard Mudgett) Improvements made in this release: ----------------------------------- * ASTERISK-24171 - [patch] Provide a manpage for the aelparse utility (Reported by Jeremy LainÃ©) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.6.0-rc1 Thank you for your continued support of Asterisk!
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