<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1264" name=GENERATOR></HEAD>
<BODY>
<DIV>
<DIV class=Section1>
<P><SPAN class=091141813-01112003><FONT face=Arial size=2>Our network is
connected to a cablemodem using a dynamic DNS service to resolve our
address. The Asterisk server has been alternately set up behind a
NAT router and without a NAT router -- that is, with two NICs, one of
which is providing NAT to the rest of the network; the office SIPs are
behind that with static private IP addresses. </FONT></SPAN></P>
<P><SPAN class=091141813-01112003><FONT face=Arial size=2>Off-premise SIPs are
all behind simple NAT routers.</FONT></SPAN></P>
<P><SPAN class=091141813-01112003><FONT face=Arial size=2>Off-premise SIPs have
been able to receive calls from and make calls through the PSTN. No problem.
Calls between on-premise SIPs, not a problem. Calls between off-premise SIPs and
any other SIPs connected to the server are a problem... they ring up but no
audio is passed in either direction.</FONT></SPAN></P>
<P><SPAN class=091141813-01112003><FONT face=Arial size=2>SIP.CONF has
NAT=YES.</FONT></SPAN></P>
<P><SPAN class=091141813-01112003><FONT face=Arial size=2>We presume that a
dedicated IP address for the Asterisk server would resolve this but we would
like to avoid the extra expense.</FONT></SPAN></P>
<P><SPAN class=091141813-01112003><FONT face=Arial size=2>What are we missing?
TIA.</FONT></SPAN></P>
<P><SPAN class=091141813-01112003><FONT face=Arial size=2>Jim
Greenfield</FONT></SPAN></P></DIV></DIV></BODY></HTML>