[asterisk-dev] Asterisk 11.20.0 Now Available
Asterisk Development Team
asteriskteam at digium.com
Fri Oct 9 18:46:45 CDT 2015
The Asterisk Development Team has announced the release of Asterisk 11.20.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.20.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25449 - main/sched: Regression introduced by
5c713fdf18f causes erroneous duplicate RTCP messages; other
potential scheduling issues in chan_sip/chan_skinny (Reported by
Matt Jordan)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
ICE is not enabled (Reported by Joshua Colp)
* ASTERISK-25427 - Callerid change does not always emit
NewCallerid AMI event (Reported by Ivan Poddubny)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog
destinations (Reported by Elazar Broad)
* ASTERISK-25410 - app_record: RECORDED_FILE variable not being
populated (Reported by Kevin Harwell)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25396 - chan_sip: Extremely long callerid name causes
invalid SIP (Reported by Walter Doekes)
* ASTERISK-25353 - [patch] Transcoding while different in Frame
size = Frames lost (Reported by Alexander Traud)
* ASTERISK-25227 - No audio at in-band announcements in ooh323
channel (Reported by Alexandr Dranchuk)
* ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
cause on call pickup (Reported by Joshua Colp)
* ASTERISK-25215 - Differences in queue.log between Set
QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
Gaetz)
* ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
for wrong or non existent peer on invite (Reported by Kevin
Harwell)
* ASTERISK-25315 - DAHDI channels send shortened duration DTMF
tones. (Reported by Richard Mudgett)
* ASTERISK-25312 - res_http_websocket: Terminate connection on
fatal cases (Reported by Joshua Colp)
* ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
Firefox 39 - add ECDH support and fallback to prime256v1
(Reported by Stefan Engström)
Improvements made in this release:
-----------------------------------
* ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
Thank you for your continued support of Asterisk!
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