[OpenSIPS-Users] OpenSIPS+asterisk: cannot place call

[Stefano Sasso]

2010/9/27 Anca Vamanu <anca at opensips.org>:
> Hi Stefano,

Hi,

> I suppose that you have the nat traversal handled also on opensips,
> right? ( since the host part of the clients registration in asterisk
> points to opensips). Then in this case it is normal that the flow of the

how should I handle this?
My only routing code is what i posted.

> if (is_method("INVITE")) {
>        if (!ds_is_in_list("$si", "$sp")) ) /* if it's not from asterisk */
>                if (!ds_select_dst("1", "5")) {
>            ....
>
>        } else
>                route(1); /* send it out */

I inserted what you said, but now the destination rings, but the
source has the call interrupted.
Asterisk says: "Got SIP response 482 "Merged Request" back from 192.168.6.130"

thanks,

-- 
Stefano Sasso
http://stefano.dscnet.org/