[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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