[OpenSIPS-Users] Loadbalancing/Failover with 1.4

Uwe Kastens kiste at kiste.org
Mon Mar 16 09:59:45 CET 2009


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
-- 

kiste lat: 54.322684, lon: 10.13586



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