[asterisk-dev] Polycom SIP Issue bug id 0009335

Olle E Johansson olle at voop.com
Tue Apr 10 00:05:07 MST 2007

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


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