[OpenSIPS-Users] Transfer issue

[Iñaki Baz Castillo]

2009/10/28 Peter den Hartog <peterdenhartog at gmail.com>:
> Oke i feel so happy right now, i fixed it! it works! i can now create dials
> over opensips, true asterisk, outside inside i can transfer, everything
> works! damn i'm happy :D!
> the answer was in my opensips.cfg and the routing back to asterisk, i've
> created a routing script that just subscribe and trows the rest in to
> asterisk.
>
> i'm thinking of creating a big straightforward blogpost about this, how you
> should do this, with what goes where and stuff like that.
>
> I would like to thank everybody who replied on this issue, thanks alot.

Congratulations ;)