[Asterisk-Users] Transferring calls between two Asterisk Servers
Douglas Garstang
dgarstang at oneeighty.com
Tue May 9 08:26:27 MST 2006
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 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.
> _______________________________________________
> --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
>
More information about the asterisk-users
mailing list