[asterisk-dev] TESTERS :: ARM YOUR TEST TOOLS, READY, SET .... TEST!

[Olle E Johansson]

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!