[asterisk-dev] Losing SIP TLS connection on spurious INVITE

[Russell Bryant]

Bruce Atherton wrote:
> I have an application that speaks SIP, and I am trying to certify that 
> it works correctly with Asterisk 1.6 beta 9.

You may want to try the latest code in the 1.6.0 branch.  A lot of 
things have changed since beta9 and have not made it into a tarball due 
to the unforeseen long time frame it has taken to complete the 
transition from Zaptel to DAHDI.

$ svn co http://svn.digium.com/svn/asterisk/branches/1.6.0 asterisk-1.6.0

Then, if you still have trouble, create a bug report on bugs.digium.com.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.