<html><div style='background-color:'><P>Hi,</P>
<P>I am new with asterisk and everything that deals with. I was wondering if there is a way to call a OH323 user or SIP user using Netmeeting/SJPhone with H323 as the default protocol without having a gatekeeper.</P>
<P>I can make a call from SIP to OH323 by specifying it in the extensions.conf file, like:</P>
<P>exten=>1001, 1, Dial(OH323/10.10.10.1)</P>
<P>so I was wondering if there was a way to call from OH323 to SIP or OH323.</P>
<P>Thanks I appreciate any thoughts and ideas,</P>
<P>azt</P></div></html>