Friends, testers,
Now is a crucial time. We're going through a lot of the patches and
branches, making decisions if they're
ready or not for 1.4. You already see [post 1.4] markers in the bug
tracker.
There's a lot of stuff to test, where we lack feedback.
Two major patches to test are:
* The SIP/RTP jitterbuffer
* The T.38 passthrough code
We also need testing of the new cool features in SIP and IAX2 - I
hope you haven't missed them.
* IAX2 native transfers of media, not signalling
IAX2 servers running the same version of Asterisk can now transfer
the media away, staying
in the signalling path to make sure CDRs are correct. This is one
step to make IAX2
more SIP-compatible. With Mark's recent love of XML, there are
propably more things to
come ;-)
* SIP direct connects :-)
If we have two devices that can speak directly without Asterisk in
the media path, we're now
setting up the call to go directly between the devices without re-
invites. Asterisk stays in the
signalling path as before.
Read the README.test-this-branch and get going. We do need test results.
http://svn.digium.com/view/asterisk/team/oej/test-this-branch/
README.test-this-branch.html
As usual, you report back in the bug tracker. The issue numbers are
listed in the
readme and should be easy to find.
It's time to give back to the community, it's time to test the new
Asterisk :-)
/Olle
PS. As an extra benefit, I've added the SSL for AMI patch to the test
branch.
Please test it too!
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