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<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2>Hi</FONT></SPAN></DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial size=2>I have managed to
set up our Asterisk server and can successfully make and receive calls via an
external Asterisk server service provider and our IAX.conf
file.</FONT></SPAN></DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial size=2>I can make SIP to
SIP calls to a remote machine on a fixed IP.</FONT></SPAN></DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial size=2>I cannot make SIP to
SIP calls from one internal phone behind our NAT firewall to another internal
phone behind our NAT firewall.</FONT></SPAN></DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial size=2>The call is received
and the recipient can answer it, but no voice nor echo can be
heard.</FONT></SPAN></DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial size=2>Any ideas would be
greatly appreciated.</FONT></SPAN></DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial
size=2>Regards</FONT></SPAN></DIV>
<DIV><SPAN class=703273811-13072004><FONT face=Arial size=2></FONT></SPAN><SPAN
style="FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><BR>James</SPAN></DIV></BODY></HTML>