Server Fault is a question and answer site for system and network administrators. It's 100% free, no registration required.

Sign up
Here's how it works:
  1. Anybody can ask a question
  2. Anybody can answer
  3. The best answers are voted up and rise to the top

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!

share|improve this question
up vote 3 down vote accepted

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.

share|improve this answer

Your Answer


By posting your answer, you agree to the privacy policy and terms of service.

Not the answer you're looking for? Browse other questions tagged or ask your own question.