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.
The following are the issues resolved in this release candidate:
- [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:
Thank you for your continued support of Asterisk!