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<DIV><FONT face=Tahoma size=2>The setup is as follows: SIP phone registers
via international link to Asterisk Box 1 and calls mean't for termination
on Asterisk Box 2 via Zaptel Channels need to be hairpinned from Box 1 to 2. How
is sip.conf configured on Box 1 and 2 so that we don't get an error: <FONT
face=Calibri><FONT color=#1f497d size=3>"Failed to authenticate user" when 1's
extensions.conf uses SIP to dial Asterisk Box 2 . How do we ensure that RTP
traffic flows from SIP phone registering at 1 directly to 2 without first
passing through 2?</FONT></FONT></FONT></DIV>
<DIV><FONT face=Calibri color=#1f497d></FONT> </DIV>
<DIV><FONT face=Calibri color=#1f497d>Tx</FONT></DIV><FONT face=Tahoma size=2>
<DIV><BR>Shaun </DIV></FONT></BODY></HTML>