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
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