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

Tiago Stein D`Agostini tiago at khomp.com.br
Wed Apr 12 05:49:24 MST 2006


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





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