The Asterisk Development Team has announced the first release candidate of Asterisk 11.13.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.13.0-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-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-24032] – Gentoo compilation emits warning: “_FORTIFY_SOURCE” redefined
- [ASTERISK-24178] – fromdomainport used even if not set
- [ASTERISK-24211] – testsuite: Fix the dial_LS_options test
- [ASTERISK-24225] – Dial option z is broken
- [ASTERISK-24249] – SIP debugs do not stop
- [ASTERISK-24301] – Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
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-11.13.0-rc1
Thank you for your continued support of Asterisk!