I have the following
setup in my test lab (which reflects very much my production installation, just
on a smaller scale)
Asterisk server
------------- Internet -------------- Home router (Linksys) ---------------Hub
----------------> Polycom 501 (Phone A)
|------------------->
Polycom 501 (Phone B)
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.
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).
Mike
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