[OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS Project

[ALICOMPUTECH]

Hello
      Everyone
               I want to replace the Asterisk (being used as a SIP Server for registration, authentication and call routing) with OpenSIPS in OpenBTS project, as i am planning to have an Asterisk cluster for dedicated services and OpenSIPS will be forwarding the SIP calls to the cluster.

OpenBTS implements GSM Um air interface and emulate the Mobile handsets as the SIP endpoint and these handsets can be used as SIP extensions in a SIP-capable server.

I need to know the handoff and/or handover support in OpenSIPS as i am a newbie to this wonderful open source solution.

If there is any pointer and/or previously handoff/handover work done please share, it will then ease my work

thanks in advance

Best Regards

Bye