Hi Ram,<br><br>Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster.<br><br>VoiceMaster only authenticates IP and cant have username password based authentication which asterisk can do. So i need to take some traffic from VoiceMaster to Asterisk and terminate it.
<br><br>Let me know <br><br>Thanks<br><br>Dan<br><br><br><div><span class="gmail_quote">On 01/01/07, <b class="gmail_sendername">ram</b> <<a href="mailto:talk2ram@gmail.com">talk2ram@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<div><span class="q"><span class="gmail_quote">On 12/31/06, <b class="gmail_sendername"><a href="mailto:sunil@koltelecom.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sunil@koltelecom.com</a>
</b> <<a href="mailto:sunil@koltelecom.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sunil@koltelecom.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">Hi Carlos,<br><br>Im interested in knowing how we can connect 2 server using SIP. Well for me both are not asterisk servers, 1 is asterisk and 2nd is an SIP based Server. i need to take multiple calls from the SIP based server and terminate it using my asterisk peers based on my dialplan. I can use SIP only as my other server doesnot support IAX2.
<br><br>How can i get that. Please let me know</blockquote>
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<div>Hi</div>
<div>what does it mean, sip based server ?</div>
<div>please do mention what is that server, most of the servers are SIP based only.</div>
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<div>ram</div></div><br>
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