The Asterisk Development Team has announced the release of Asterisk 11.13.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.13.0 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:
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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0
Thank you for your continued support of Asterisk!