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...xxx.yyy.165.201 which is registered as xxx9931211 4) Call completes fine and audio works 5) PSTN Hangs up and the sends a BYE to Asterisk 6) Asterisk recieves the bye and sends BYE directly to the UA skipping the proxy 7) Results is that the proxy never recieves a BYE Complete SIP Trace http://www.routerboy.com/sip2004041902.html Example of actual Record-Route issue: -Message 9 the Proxy sends a Record-Route to Asterisk and Message 10 seems to be built are return to the Proxy but not with the infomation from the Record-Route Statement.) SIP MESSAGE 9 xxx.yyy.91.74:5060(4) -> xxx.yyy.7...

I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.