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Re: [Sipwitch-devel] Problem statement
From: |
Charles N Wyble |
Subject: |
Re: [Sipwitch-devel] Problem statement |
Date: |
Mon, 25 Apr 2011 13:23:33 -0500 |
User-agent: |
Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.2.15) Gecko/20110419 Thunderbird/3.1.9 |
> '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
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