[OpenSIPS-Users] Calls from asterisk to opensips

[Premalatha Kuppan]

Hi,

I have integrated set up of opensips with TLS(1.6.2) and Asterisk (1.4.3.1)
running.

Now, my user is sending registration request via TLS, registration is
succesful; the call fails.

Here is the setup iam trying,

User (UDP) and USER with TLS gets registered with Opensips.
Asterisk performs IVR for all incoming calls to opensips.

callee---------->Opensips--------->Asterisk-------------->opensips----------->called


Since Asterisk 1.4.3 doesn't support TLS; after IVR, iam tying to forward
the call to opensips.
extension.conf:

exten => s2,n,Dial(SIP/${aaa}_${bbb}_${ccc}@OPENSIPS:5061,20,r)

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

if (is_method("INVITE"))
         {
                $var(z)=$(tu{uri.user});
                 if($var(x)=~"[0-9]+_[0-9]+_[0-9]+") {
                xlog("Call from Asterisk \n");
                route(1); }
        }

Please help me, after IVR, i have to forward the call to opensips and
opensips should connect the callee and called.

How should i change in opensips.cfg to handle this.

I appreciate the valuable input.

Thanks,
Prem
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