Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the third beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at http://downloads.digium.com/pub/asterisk/
This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then.
Included in this release are the following issues reported by the community:
- Update spiral support in trunk and 1.6.x branches to match what is in 1.4 (related to issue #13630).
- Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over (issue #14815).
- Fix a bug where the codecs of the called party leg were not properly sent back to the call leg when reinvited (issue #13569).
- Fix broken attended transfers (issue #15183).
- Add flags to chanspy audiohook so that audio stays in sync (issue #13745).
- Resolve issues with choppy sound when using res_timing_pthread (issue #14412)
Additionally, an update to chan_iax2 related to issue AST-2009-001 is included in this beta release. For more information, see:
http://downloads.asterisk.org/pub/security/AST-2009-001.html
For a full list of changes in this beta, please see the ChangeLog:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/ChangeLog
You can get more information about the new features and various changes in Asterisk 1.6.2.0 at:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/CHANGES
And if you’re upgrading from previous versions of Asterisk see this file:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/UPGRADE.txt
Issues discovered in testing of this beta can be reported at http://issues.asterisk.org
Thank you for your continued support of Asterisk!
