[OpenSIPS-Users] Loadbalancing/Failover with 1.4

[Bogdan-Andrei Iancu]

Hi Uwe,

First of all, route via dispatcher only initial requests (without totag) 
and use record-route on the dispatcher box. This will force BYEs from 
PSTN to be routed via the same servers.

Regards,
Bogdan

Uwe Kastens wrote:
> Hello,
>
> I have the following setup:
>
> UA <> opensips <> asterik1 to asterisk3 <> pstn-gw
>
> With the dispatching module its easy to handle the calls from UA to
> pstn. The other way is the tricky one, since the pstn-gw will route the
> calls randomly to one of the asterisk-server. So I need a stable
> solution to route all sip-packets belonging to one dialog to the
> asterisk-server the communication was started.
>
> How can I solve this? route-header for the asterisk-server will cause,
> that some requests are routed directly between UA and asterisk (BYE).
>
> BR
>
> Kiste
>