Opensips configuration

Is there documentation how to setup the opensips module for ASTPP?

We have 2 servers, server 1 is the primary server where 95% of the calls are delivered and sip users are registered.
So the other 5% of the calls and sip users go to sever 2 and opensips needs to tell freeswitch where every user is registered so the calls gets delivered and not in the voicemail.

You can look at: OpenSIPS configuration for 2 or more FreeSWITCH installs - FreeSWITCH - Confluence