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:
Bug
- [ASTERISK-22252] – res_musiconhold cleanup – REF_DEBUG reload warnings and ref leaks
- [ASTERISK-23577] – res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL
- [ASTERISK-23634] – With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls
- [ASTERISK-23767] – Dynamic IAX2 registration stops trying if ever not able to resolve
- [ASTERISK-23994] – res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname
- [ASTERISK-23997] – chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer
- [ASTERISK-24019] – When a Music On Hold stream starts it restarts at beginning of file.
- [ASTERISK-24027] – MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
- [ASTERISK-24032] – Gentoo compilation emits warning: “_FORTIFY_SOURCE” redefined
- [ASTERISK-24043] – ARI /continue fails to actually continue into the dialplan
- [ASTERISK-24136] – Security: Crash in Asterisk’s PJSIP code when subscribing to an event with an unexpected body type
- [ASTERISK-24143] – pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
- [ASTERISK-24147] – ARI: channel hangup crashes asterisk process
- [ASTERISK-24161] – PJSIPShowEndpoint gives inaccurate count of list items
- [ASTERISK-24178] – fromdomainport used even if not set
- [ASTERISK-24212] – testsuite: Sporadic crash due to assert on stopping RTP engine
- [ASTERISK-24225] – Dial option z is broken
- [ASTERISK-24229] – ARI: playback of sounds implicitly answers channel, preventing early media playback
- [ASTERISK-24231] – crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable
- [ASTERISK-24234] – app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg()
- [ASTERISK-24236] – res_hep_rtcp: Module incorrectly depends on pjsip
- [ASTERISK-24237] – CDR: FRACK With PJSIP blonde transfer.
- [ASTERISK-24241] – crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack
- [ASTERISK-24245] – gcc 4.1.2 complains of files that do not end with newlines
- [ASTERISK-24249] – SIP debugs do not stop
- [ASTERISK-24254] – CDRs: Application/args/dialplan CEP updated during dial operation
- [ASTERISK-24264] – ARI: Adding a channel to a holding bridge automatically starts MOH
- [ASTERISK-24290] – Endpoint identifier match value fails to parse when CIDR network format is specified
- [ASTERISK-24301] – Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
- [ASTERISK-24331] – Unexpected Errors in Asterisk Manager Interface Output
Improvement
- [ASTERISK-24171] – Provide a manpage for the aelparse utility
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!