[OpenSIPS-Users] Any Help on TCP/TLS

[Premalatha Kuppan]

Hi Bogdan,

This the logic i used to handle calls from asterisk to opensips. But i thnk
since there would be via header, call is going to destination via Asterisk
(which doesn't support TCP/TLS)

Can you please help. I want the call to be handled by opensips after IVR
done by asterisk.

i.e finally callee<---------------------->opensips<------------------>called

What needs to adapted ?

       if ($rU==NULL) {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }

     if (is_method("INVITE"))
       if (method=="INVITE|BYE")
           {
                 $var(z)=$(tu{uri.user});
                if($var(z)=~"[0-9]+_[0-9]+_[0-9]+") {
                xlog("Call from Asterisk PROTOCOL=$pr PORT=$op PROTOCOL
PORT=$oP  RQT=$oU \n");
                 if (!lookup("location")) {
                sl_send_reply("404","Not here"); }
                 }
        }

Please advice.

Thanks,
Prem

On Fri, Jul 23, 2010 at 11:31 AM, Premalatha Kuppan <premalatha at ngintech.com
> wrote:

> Hi Bogdan,
>
> Thanks for your reply. Is the logic correct ?
>
> After IVR, i want asterisk to transfer call to opensips and opensips to
> handle the call further. But, i still wonder how opensips would get the call
> details of orignator, do i need to manipulate nything.
>
> Also, after googling found Opensips1.6x is compactable with 1.4x. So Iam
> using the same versions. But, yesterday i tried insalling asterisk 1.6x
> (since it has TLS/TCP) support). So far successful, i can forward calls to
> asterisk from Opensips. Will I get any other problems..i dono..Please
> advice.
>
> I appreciate your valuable comments.
>
> Thanks,
> Premalatha
>
>
>
>
>
> On Thu, Jul 22, 2010 at 8:18 PM, Bogdan-Andrei Iancu <
> bogdan at voice-system.ro> wrote:
>
>> Hi Premalatha,
>>
>> have you checked the opensips logs for error? can you post a call flow
>> showing exactly which step fails ?
>>
>> Regards,
>> Bogdan
>>
>>
>> Premalatha Kuppan wrote:
>> > Hi,
>> >
>> > I posted before my query; but no repsonse :(
>> >
>> > Can some1 helps, whether this logic works,
>> >
>> > Iam using opensips 1.6.2(TLS) and Asterisk(1.4.3.1).
>> >
>> > 1. all users registering @ opensips
>> > 2. All calls to opensips forwarded to sasterisk for IVR; thorugh IVR
>> > input destination is known.
>> > 3. Since asterisk 1.4x version doesnt support TLS/TCP. Iam forwarding
>> > call to Opensips; indeed here some messy happens. Asterisk sents
>> > invite to Opensips, opensips reach the destination via asterisk (First
>> > of all, iam not sure this logic is true). For TLS, call is successful
>> > but no audio. For TCP, call fails after IVR.
>> >
>> > Only my destination is TLS/TCP enabled.
>> >
>> > I appreicate valuable input and advice on thsi logic.
>> >
>> > Please help.
>> >
>> > Thanks,
>> > Prem
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > Users mailing list
>> > Users at lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>>
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Bootcamp
>> 20 - 24 September 2010, Frankfurt, Germany
>> www.voice-system.ro
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
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