[asterisk-dev] Asterisk 10.1.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Fri Dec 30 14:10:00 CST 2011
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 10.1.0. This release candidate is available for immediate download
at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 10.1.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release candidate:
* Handling of AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop
(closes issue ASTERISK-19095)
* Fix for timing source dependency issues involving Music On Hold
(closes issue ASTERISK-17474)
* Fix for queue set penalty crash when issued without a specified queue
* Fixes for several issues with audiohooks and ChanSpy
* Multiple fixes in chan_sip relating to TCP settings, bad file descriptors,
potential deadlocks, and much more
(closes issue ASTERISK-18837, ASTERISK-18799, ASTERISK-17760, and many others)
* Restore call progress code for analog ports
(closes issue ASTERISK-18841)
* Fix parsing issues in main/pbx.c
(closes issue ASTERISK-18909)
* Fix regression in res_rtp_asterisk that rtp/rtcp set debug ip only works
when port is specified
(closes issue ASTERISK-18693)
And much more! For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.1.0-rc1
Thank you for your continued support of Asterisk!
More information about the asterisk-dev
mailing list