Hi,<br><br>I have integrated set up of opensips with TLS(1.6.2) and Asterisk (1.4.3.1) running.<br><br>Now, my user is sending registration request via TLS, registration is succesful; the call fails.<br><br>Here is the setup iam trying,<br>
<br>User (UDP) and USER with TLS gets registered with Opensips.<br>Asterisk performs IVR for all incoming calls to opensips.<br><br>callee---------->Opensips--------->Asterisk-------------->opensips----------->called <br>
<br>Since Asterisk 1.4.3 doesn't support TLS; after IVR, iam tying to forward the call to opensips.<br>extension.conf: <br><br>exten => s2,n,Dial(SIP/${aaa}_${bbb}_${ccc}@OPENSIPS:5061,20,r)<br><br>At Opensips.cfg, (Note: both opensips and Asterisk running on same box. Opensips listening on port 5060 and 5061 and asterisk on 5070. Calls from asterisk will always have the user uri format has aaa_bbb_ccc<br>
<br>if (is_method("INVITE"))<br> {<br> $var(z)=$(tu{uri.user});<br> if($var(x)=~"[0-9]+_[0-9]+_[0-9]+") {<br> xlog("Call from Asterisk \n");<br>
route(1); }<br> }<br><br>Please help me, after IVR, i have to forward the call to opensips and opensips should connect the callee and called.<br><br>How should i change in opensips.cfg to handle this.<br>
<br>I appreciate the valuable input.<br><br>Thanks,<br>Prem<br><br><br>