<DIV>Hi all,</DIV>
<DIV>I have just setup Asterisk, but the problem is that all RTP stream pass through Asterisk, I mean all call setup and voice stream pass trough Asterisk once the call is established.</DIV>
<DIV>Is there a way that call setup is established, the RTP stream pass just between the SIP endpoints.</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>Example:</DIV>
<DIV>
<DIV>Works like this</DIV>
<DIV>SIP IP phones <-----------Asterisk RTP stream--------------> SIP IP phone</DIV></DIV>
<DIV> </DIV>
<DIV> Asterisk</DIV>
<DIV> </DIV>
<DIV>SIP IP phones <------------------RTP------------------------> SIP IP phone</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>Thanks!</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV><BR><BR><p>
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