Re: [Sipwitch-devel] Problem statement

[Charles N Wyble]

> '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.

Same here.

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

Same.

> 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.

Same. I also want encryption.

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

Haha. Indeed not.

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

Freeswitch is cool. Not so lightweight though.



-- 
Charles N Wyble address@hidden @charlesnw
http://blog.knownelement.com
Building tomorrows alternate default free zone

signature.asc
Description: OpenPGP digital signature