I am having a strange problem with my asterisk server and i'm not quite sure how to solve it. We are using SER as our SIP registrar and proxy so all SIP devices are registered with the SER and any call setup/routing is handled by the SER. Asterisk is currently only used for voicemail. When I dial into one of my DID lines, it rings for a specified number of seconds and if nobody picks up hunts to voicemail (the asterisk server). This all works fine. The problem I am having is when somebody dials in and the SIP target is unavailable. When the target is unavailable the SER immediately hunts for the next target, which is usually asterisk. All of the SIP transactions are taking place and I can see the call hitting the Asterisk server from the console. Everything looks normal, but there is no audio. We have another voicemail system (pingtel eXchange) [which we are trying to get away from] and this seems to work fine with the Pingtel so I can only think that I am missing something in my Asterisk configs. This happens whether I am dialing in from the PSTN or 100% within our VoIP system. My guess is that it works with the Pingtel system because we have it set up to use the jasomi as its outbound proxy, but I'm not sure how to tell asterisk to do this. Does anybody have any suggestions? We are using: Asterisk 1.0.0 Jasomi PPT for Nat traversal SER 0.8.14 Thank you very very much for your input. Dan