Asterisk 11.20.0-rc1 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.20.0-rc1.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.20.0-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-25215] – Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember
  • [ASTERISK-25227] – No audio at in-band announcements in ooh323 channel
  • [ASTERISK-25265] – DTLS Failure when calling WebRTC-peer on Firefox 39 – add ECDH support and fallback to prime256v1
  • [ASTERISK-25299] – RTP port leaks with incoming OOH323 calls
  • [ASTERISK-25312] – res_http_websocket: Terminate connection on fatal cases
  • [ASTERISK-25315] – DAHDI channels send shortened duration DTMF tones.
  • [ASTERISK-25320] – chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite
  • [ASTERISK-25346] – chan_sip: Overwriting answered elsewhere hangup cause on call pickup
  • [ASTERISK-25394] – pbx: Incorrect device and presence state when changing hint details
  • [ASTERISK-25396] – chan_sip: Extremely long callerid name causes invalid SIP
  • [ASTERISK-25407] – Asterisk fails to log to multiple syslog destinations
  • [ASTERISK-25410] – app_record: RECORDED_FILE variable not being populated

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0-rc1

Thank you for your continued support of Asterisk!

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