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!


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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