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
