I just managed to get the asterisk zimlet working on a different machine than the zimbra box and it works well.

The issue I saw was RTP streams not having any where to go... this was leading to one end of the call rining and then dying. I was seeing this line in my SIP debug SDP messages

c=IN IP4 0.0.0.0


I set canreinvite=yes on my asterisk 1.2 machine and now calls between asterisk and zimbra work with no hitches.

Next I am going to try and reverse the order the phones ring..... I want to be called before the person I am calling....

Thanks for you hard work with Zimbra guys... it is really polished. I hope I can keep helping by reporting issues....