<div>Hi folks,</div>
<div> </div>
<div>I have Asterisk as a PSTN termination GW, Openser will handle the user registration.</div>
<div>Openser users has name format 7xxxxxxxx ( all digistes), which can call each other peer to peer ( it is a basic openser function).</div>
<div>I want to forward the call ( 1xxxxxxxxxx or 011xxxxxxxxxxxxx ) to Asterisk when openser user try to terminate US or International calls.</div>
<div>I tried to use the setting like this:</div>
<div> </div>
<div>if(is_method("INVITE") && !has_totag() && uri=~"sip:011[0-9]+@.*") {<br> route(4);<br> exit;<br> };</div>
<div> </div>
<div> </div>
<div>
<p>route[4] {<br> strip(3);<br> # route to Asterisk Media Server<br> rewritehostport("<a href="http://222.222.222.79:5060">222.222.222.79:5060</a>");<br> if (!t_relay()) {<br>
sl_reply_error();<br> };<br> exit;<br>}</p></div>
<div> </div>
<div> </div>
<div>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.
</div>
<div> </div>
<div> </div>
<div>Steve</div>