[Asterisk-Users] SIP test suite: anyone with spare time?

[John Todd]

The following URL leads to a limited (INVITE only) test suite for SIP 
protocol usage: proxies and UA's among other tests.  Since Asterisk 
implements partial proxy and full UA functionality, it may be worth 
someone's time to take a swing at getting this installed and pointed 
at an Asterisk box:

http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/index.html

If you could then record the failure modes, and document them on 
http://bugs.digium.com/ that would perhaps help the developer 
community in identifying where there are major/minor SIP flaws and in 
turn allow repairs that would make the SIP channel module in Asterisk 
a more stable and robust implementation.  The implementation for the 
test suite is in Java.

Sorry, due to time constraints, I don't even have a prayer of looking 
at this fully, so I leave it to someone else to forge ahead and do 
some testing...

JT