[OpenSIPS-Users] Any Help on TCP/TLS

[Premalatha Kuppan]

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
> > ------------------------------------------------------------------------
> >
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> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro
>
>
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>
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