Forgot to mention. The polycom phones in this case generate a new INVITE message with a new call id when transferring a call. As far as the SIP proxy is concerned, it's a new call.
Doug. > -----Original Message----- > From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] > Sent: Tuesday, May 09, 2006 8:40 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Transferring calls between two Asterisk > Servers > > > Douglas Garstang wrote: > > > I know there's bugs open on this. > > This is not a bug. There is no practical way to handle a SIP > client who > tries to transfer a call between Asterisk servers directly. The proper > way to handle is this to ensure that your proxy/load balancer ensures > that all SIP calls placed by a phone go to the same Asterisk server as > long as that phone has any active calls. It should only > randomly pick a > server when it is placing a call and has nothing else active. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users