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<DIV><FONT face=Arial size=2>Hi ALL;</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have users with Sipura/Linksys phones
regsitered behind Nat( useing STUN at phone not
portforwarding ) in my Asterisk box, when I try to call them
with another phone i got:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Got SIP response 404 "Not Found" back from
217.6.190.4</FONT></DIV>
<DIV><FONT face=Arial size=2>SIP/217.6.190.4:5060-666d is
circuit-busy<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>Is above mentioned problem relates to
"Nat", Is there anybody who use sipura with STUN method and can recive
calls?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>My asterisk Sip.conf for Nat has the
following:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[sipura]</FONT></DIV>
<DIV><FONT face=Arial size=2>..</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>nat=yes</FONT></DIV>
<DIV><FONT face=Arial size=2>canreinvite=no</FONT></DIV>
<DIV><FONT face=Arial size=2>qualify=1000</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Appreciate any help</FONT></DIV>
<DIV><FONT face=Arial size=2>Mohammad</FONT></DIV></BODY></HTML>