[Asterisk-Users] Transferring calls between two Asterisk Servers

[Douglas Garstang]

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:kpfleming at digium.com]
> 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.
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