9 apr 2007 kl. 22.37 skrev Timothy McKee:
I have collected the requested data and posted it against bug id
9335. This issue appears to manifest when asterisk is part of
the INVITE to a polycom phone and the latency exceeds the 500ms T1
default timer on the Polycom. This issue did *not* occur in a much
older (8/1/04 CVS Head) release with identical configurations.
We've added SIP timer support in Asterisk and times out in a
different way than the very old CVS versions, so we're propably
much more SIP conformant in that regards now. Try adding qualify=yes
for the phone, then we'll adjust Asterisk's timers to the actual
latency.
/Olle
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