[OpenSIPS-Users] Transfer issue

[Khan]

Please let me know when you setup the blogspot i'm very interested in seeing
how you did it, or if you could provide me your configuration that will be
really great.

Thanks in advance...

Khan

On Wed, Oct 28, 2009 at 7:34 AM, Iñaki Baz Castillo <ibc at aliax.net> wrote:

> 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 ;)
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Khan


VoIP Rookie
Every beginning has an end regardless we believe it or not...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20091028/8d67f838/attachment.htm