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<DIV><SPAN class=375062612-07112006><FONT face=Arial size=2>I have the following
setup in my test lab (which reflects very much my production installation, just
on a smaller scale)</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=System>Asterisk server
------------- Internet -------------- Home router (Linksys) ---------------Hub
----------------> Polycom 501 (Phone A)</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT
face=System> |------------------->
Polycom 501 (Phone B)</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial size=2>All calls go through
my asterisk server, even if its from one Polycom to the other. If I dial from
phone A to phone B, audio doesnt get passed for the first 1-2 seconds. I
end up saying "hello? hello? hello?" and eventually I heard something. It
makes for a bad user experience.</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial size=2>What can be the
problem? I imagine the NAT isnt the problem, or there would be no audio at
all. My Asterisk is running 1.2.4, and my Polycom phones at running
bootrom 3.2.2 and SIP 2.0.1 (fairly recent).</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2>Mike</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
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