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I want to use FreeSWITCH instead of Asterisk because of it's performance compared to Asterisk. I know that FreeSWITCH can be a full PBX or just run parts (modules) to do only the things I want it to.. But I am not sure where OpenSIPS fits into the equation. Lets say I had 5 FreeSWITCH servers to handle voice calls (inbound and outbound) and voicemail for my users. Could I have all of the extensions in the OpenSIPS router and use it to authenticate calls, then hand them off to FreeSWITCH?

If so, do I have to put any Extension information in FreeSWITCH at all for my users? I am trying to avoid having 5 FreeSWITCH servers with duplicate extensions in each!

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Opensips is used for creating highly scalable SIP signaling routers. So yes, use OpenSIPS with the Carrier Route module to authenticate peers and route calls to the FreeSWITCH boxes.

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