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I don't think the problem is NAT-related. Looks like "To" header in SIP INVITE message do not match to "User ID" in sipura settings.<BR>
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On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote:
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<FONT SIZE="2"><FONT COLOR="#000000">Hi ALL;</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">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></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">Got SIP response 404 "Not Found" back from 217.6.190.4</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">SIP/217.6.190.4:5060-666d is circuit-busy</FONT></FONT><BR>
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<FONT SIZE="2"><FONT COLOR="#000000">Is above mentioned problem relates to "Nat", Is there anybody who use sipura with STUN method and can recive calls?</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">My asterisk Sip.conf for Nat has the following:</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">[sipura]</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">..</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">nat=yes</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">canreinvite=no</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">qualify=1000</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">Appreciate any help</FONT></FONT>
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<FONT SIZE="2"><FONT COLOR="#000000">Mohammad</FONT></FONT>
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<PRE>
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