[asterisk-dev] Polycom SIP Issue bug id 0009335

[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.

Can one of the SIP gurus take a quick look at this now that the  
verbose debug data is posted?

thanks,

tim mckee



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20070409/777a37e9/attachment.htm