Asterisk 11.13.0-rc1 Now Available

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

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!

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