[Sipwitch-devel] Problem statement

[address@hidden]

'K, here's the deal:

I've got a small VPS sitting on a fat pipe, and I plan to start using it
as my messaging hub.

I'm looking for a lightweight SIP registration and call server, with
maybe (some very optional) IVR capability.

Requirements:

a) Phones registered to an extension can call others by their extension,
e.g., extension 201 can call extension 212.
b) Basic URL-based call routing, e.g., the URL 'sip:address@hidden'
gets routed to an appropriate extension number like 208.
c) Basic call-handling, e.g., if my handset isn't registered to my
extension, and someone tries to call me, the caller needs to get a
(temporary) redirect to my VoIP provider, or if my extension is busy, etc.

Asterisk, which I'm using now, is just too much, no way can it be called
lightwieght.

I've considered FreeSwitch, but haven't tried it yet.

I've tried Bayonne, but there are no worthwhile documentation or forums
to get the required information.

SIP Witch should be perfect.  If it can be made to work.  But the
documentation is in even worse state than Bayonne's.

So I'm listening. :)