[OpenSIPS-Users] Using SST with Asterisk as a PSTN gateway

[Bogdan-Andrei Iancu]

Hi Robert,

SST is available only in trunk Asterisk.....

What you can try is to use SST from OpenSIPS - see the SST module - 
http://www.opensips.org/html/docs/modules/1.4.x/sst.html

Regards,
Bogdan

Robert Borz wrote:
> I want to use sip session timers to ensure client sip-sessions are really active for billing purposes. We use Asterisk version 1.4 for pstn connectivity which does not support SSTs.
>
> So I want to use SSTs only on the client-side of a dialog. Would this be possible? Has anybody a setup like this?
>
>
> Regards,
> Robert
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>