[Asterisk-Users] SIP conections, with RTP not going trough Asterisk

[Peter Bowyer]

On 17/04/06, Tiago Stein D`Agostini <tiago at khomp.com.br> wrote:
> Hi, sorry to bother again. But I still cannot make it work. I made all
> acounts have canreinvite=yes, but found no option in Dial aplication to
> make the phones exchange RTP directly between them.  Can anyone tell me
> wich option should I look at? I am stuck with this (probably simple)
> problem for almost a whole week.

You're trying too hard - unless you tell it not to, the Dial
application will do what you're asking. As Olle said, this is the
default.

Peter

--
Peter Bowyer
Email: peter at bowyer.org