[OpenSIPS-Users] Incoming Calls to Opensips from asterisk

[Premalatha Kuppan]

Any insight ? Can any one help me and give me some idea please ?

On Mon, Jul 19, 2010 at 12:21 PM, Premalatha Kuppan <premalatha at ngintech.com
> wrote:

> 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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