Hi folks, I have Asterisk as a PSTN termination GW, Openser will handle the user registration. Openser users has name format 7xxxxxxxx ( all digistes), which can call each other peer to peer ( it is a basic openser function). I want to forward the call ( 1xxxxxxxxxx or 011xxxxxxxxxxxxx ) to Asterisk when openser user try to terminate US or International calls. I tried to use the setting like this: if(is_method("INVITE") && !has_totag() && uri=~"sip:011[0-9]+@.*") { route(4); exit; }; route[4] { strip(3); # route to Asterisk Media Server rewritehostport("222.222.222.79:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } OpenSER keep looking the 011xxxxxxxxx from location and got 408 always , it does not forward the call to Asterisk. I read a doc about how to share acc and voice meail with Asterisk, will it have to use failure_route to accomplish it ? I am new to Openser and dont know much about the routing logic. Anybody could point me to the right way? A sample cfg will really appreciated. Steve