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

[Tiago Stein D`Agostini]

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?


Thanks in advance