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

[Ronald Wiplinger]

Tiago Stein D`Agostini wrote:
> Hi,
>
>   Ie been looking for some time how to use asterisk  to initiate SIP 
> connections between 2 IP phones,  but afetr initiated the 
> communication making the RTP go directly from one telephone to the 
> other, without passing by asterisk. Unfortunately I found no 
> explanations of how to do it.
>
> Does anyone care to give a pointer to any explanation about how to do it?
>
canreinvite=yes
and look at the options for dial()
>
> Thanks in advance
>


bye

Ronald Wiplinger