[Gllug] Is this a silly idea - VOIP telephony with Asterisk?

[John Winters]

On Sat, 2006-02-11 at 16:20 +0000, Matthew Thompson wrote:
> On 11 Feb 2006, at 16:05, John Winters wrote:
> 
> > The problem is that the VoIP phone adaptor wants all incoming packets
> > addressed to e.g. UDP port 7016 (I forget the number, but it's a
> > specified set of UDP port numbers) forwarded to itself.  Instructions
> > are included for setting up your router to forward the packets
> > appropriately.  AFAICT there's no other routing information, and so no
> > way for a plain routing device to decide which of two boxes each
> > individual packet should be sent to.
> 
> What manufacturer are the devices

One is Fritz and the second one I'm playing with (with a view to having
a phone down in the shed at the bottom of my garden) is a Linksys box.
AFAICT it's the same one which Vonage provide with their service, but as
I've bought it separately the configuration is unlocked.

I've just discovered that the user manual is severely incomplete.
According to the manual, all the configuration is done by way of a touch
tone phone, but pointing a web browser at the device has revealed a much
richer configuration interface.

>  and what VoIP service provider?

Currently I have numbers registered with Gradwell and one other (the
name escapes me for the moment).  I'm probably going to register further
numbers with Gradwell.


> Is this a SIP based service?

Yes.

> SIP is notoriously difficult to play with on a simple port forwarding  
> basis and you Asterisk plan is a good way around this - I've done  
> similar myself in the past.

The impression I get is that the protocol hasn't quite been thought
through from this viewpoint.  I may however be talking through an
unsuitable orifice because I'm just getting started with SIP.

Do I need the latest version of Asterisk, or will the Sarge packages
(1.0.7) be good enough to get started?

Cheers,
John

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