This post is a bit late but may be relevant to someone.
I run a small LAN with a Zeroshell Release 1.0.beta12 routing SIP traffic to a Asterisk VOIP server via virtual server rules. It works very well for us, However I have not been able to update to Release 1.0.beta14. When I load beta14 I find that the incoming audio traffic on port range 10000-10100 gets dropped. I used TCPDUMP on both the beta12 and beta14 builds to confirm this but have not been able to work out why it happens.
And as for VOIP server recommendations, I spent quite a bit of time looking at alternatives to Asterisk and have concluded that Freeswitch with a FreePBX UI would be an optimal solution. I have trailed this on a test server and it works well and is easy to install and maintain on Debian.