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<DIV><FONT face=Arial size=2>I have a test box setup and I can make outbound
calls on the PSTN thru the diguim card, however I can not make a sip user to sip
user call by dialing the extensions. I am getting the following
error.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-- Called cisco7960<BR> -- Got
SIP response 482 "Loop Detected" back from 208.218.14.123<BR> == No one is
available to answer at this time</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>CLI> sip show
peers<BR>Name/username
Host Dyn Nat
ACL Mask
Port Status</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>cisco7960/5052
208.218.14.123 D N
255.255.255.255 5060 OK (1 ms)</FONT></DIV>
<DIV><FONT face=Arial size=2>garycarr/5011
208.218.14.123 D N
255.255.255.255 5060 OK (1 ms)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>sip.conf statements</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>register => <A
href="mailto:garycarr@sip.carolina.net/5011">garycarr@sip.carolina.net/5011</A><BR>register
=> <A
href="mailto:cisco7960@sip.carolina.net/5052">cisco7960@sip.carolina.net/5052</A></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[cisco7960]<BR>type=friend<BR>host=dynamic<BR>nat=yes<BR>qualify=200<BR>dtmfmode=rfc2833<BR>canreinvite=no<BR>mailbox=5052<BR>callerid="Cisco
7960"<BR>context=local</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[garycarr]<BR>type=friend<BR>host=dynamic<BR>nat=yes<BR>qualify=200<BR>dtmfmode=rfc2833<BR>canreinvite=no<BR>mailbox=5011<BR>callerid="Gary
Carr"<BR>context=local</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>extensions.conf statements</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>exten =>
5011,1,dial(SIP/garycarr,20,tr)<BR>exten =>
5052,1,dial(SIP/cisco7960,20,tr)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Is this a possible nat issue? I can make a good
call from behind the firewall doing sip to pstn so it seems 2 way traffic thru
the firewall is working.</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 am still sifting thru the sip debug info but
anyone has any ideas that would be great.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Gary</FONT></DIV>
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