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<div class="gmail_quote">On Jan 5, 2008 4:40 AM, ameel <<a href="mailto:akamboh@connectageek.com">akamboh@connectageek.com</a>> wrote:<br>
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<div><font face="Arial" size="2">I am trying to setup asterisk as a registrar and sip server only.</font></div>
<div><font face="Arial" size="2">Currently When I make calls all my rtp traffic is going through the asterisk server as a B2BUA.</font></div>
<div><font face="Arial" size="2">Is it possible to turn off this feature and have all my calls RTP traffic going directly to the SIP UA?</font></div></div><br>______</blockquote>
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<div>Hi</div>
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<div>Try SER or OpenSEr</div>
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<div>ram</div></div><br>