asterisk users - May 2006 - Transferring calls between two Asterisk Servers

Has anyone gotten around the general problem where you have multiple Asterisk
servers in a cluster, any of which may take a call. If you transfer a call from
one Asterisk system to another, the second has no idea of the first call, and
the first refuses to release the call and logs:

May  5 12:40:49 NOTICE[2864]: chan_sip.c:6758 get_refer_info: Supervised
transfer requested, but unable to find callid
'7df42620-54e177de-4eaf29@xxx.187.128.19'.  Both legs must reside on
Asterisk box to transfer at this time.

I know there's bugs open on this.

Doug.

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.

Then, can you please explain to me what this is all about:
http://bugs.digium.com/view.php?id=3710

This certainly appears to be a work in progress to fix this issue.
> -----Original Message-----
> From: Kevin P. Fleming [mailto:kpfleming@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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> 
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> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

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@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.
> _______________________________________________
> --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
>