Asterisk 12.6.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 12.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.6.0 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-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

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.6.0

Thank you for your continued support of Asterisk!

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