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Re: [LUG] Asterisk and getting rid of unwanted calls

 

On Fri, 7 May 2010, Charlie Pearce wrote:

Hi Paul,

I have some experience running an asterisk box (specifically TrixBox), so
although I'm no expert by my means, I'll offer my 2 pennies:

Given your hardware I would recommend an Ethernet SIP device like one of
these: http://www.voipon.co.uk/linksys-analog-adapters-c-2_29.html - this will
give you flexibility, both in future use - and actual positioning of hardware
- you can place this adaptor near the phone line and have your PC anywhere on
the network.

The SPA 3102 is the one in that range:

http://www.voipon.co.uk/linksys-spa3102-p-301.html

Similar to the Grandsteam I posted earlier - one FXO and 1 FXS port. I've found the Linksys boxes to be OK, but harder to setup than the Grandsteam boxes...

With regards to Sky - I believe the Sky boxes use a dial-up connection for
their communication. This would be borne out by the fact that they are placed
after the ADSL microfilter -like a normal telephone. In theory you can route
Dial-UP through VOIP (as you can fax) but in practice the success of this
varies best on hardware/network conditions etc (there is a long explanation of
this, which I'll spare you.) However this is a moot point - a Sky box only
ever dials out, and only if the line is not busy. As the above mentioned
adaptors are generally configured to only pick up the line for incoming calls,
and only can only dial out if the line is free, there should be no line
collisions. Therefore you can plug your sky box straight into the wall/ADSL
Microfilter and bypass your Asterisk server. It's worth noting that Sky
generally only require you to keep the box connected to the line for the
initial 12 month contract (and you can pay a one-off fee to opt-out of this) -
but if you upgrade to a subsidised Sky+ or HD box they make you renew this
clause. Also if it matters to you - (eventually) you won't get box office
access if you disconnect it from the phone line.

Indeed - there's really no point routing the sky via the asterisk box.

With regards to using the SIP client on your mobile - if you have a strong
wifi connection it should be possible to achieve good quality calls - however
if you have an "iffy" connection you may want to experiment with using
different codecs on the server for the mobile simple client to reduce bandwith
- unfortunately the less bandwidth heavy codecs tend to eat up more CPU usage
on the server - so YMMV. I'd have though if you are only doing one call at a
time you'd be fine.

Odd as this may seem, using a compressing codec can be counter productive over Wi-Fi. Wi-Fi is half duplex - ie. only one way at a time. VoIP is full duplex by it's very nature. So the way Wi-Fi boxes pretend to be full duplex is by switching the tx off and the rx on - in negotiation with the station they're talking to - this turn-around time is finite, and in tests I've done, it takes about the same time as about a 100 byte packet of data. A standard 50ms voip packet at G711 codecs is 160 bytes (plus IP overhead) as the packets get smaller the load on the Wi-Fi increases...

So while I use Wi-Fi at home, I always use G711 codecs, but I'll never sell a Wi-Fi solution to a client as it's not at all controllable. DECT is much better.

If you want to experiment without buying any hardware I would suggest you
install TrixBox CE (free) on your box and get a temporary incoming sip trunk
from a company such as Gradwell or AQL (and there are plenty of others out
there). This will give you a local or 0845 etc, number which comes across the
Internet via SIP or IAX which you can then route to your mobile using SIP over
wifi - this will let you determine if your hardware is viable before investing
too much.

Even easier might be pbxinaflash. And for cheap VoIP connections, look at sipgate.co.uk - they'll give you a free geographic number.

However, I think Paul was really looking to use asterisk as a fancy call screening mechanism... If you really want cheap VoIP (or free) look at one of the Betamax providers - see http://backsla.sh/betamax for a list.

In fact you may find this route is all you need - I'm planning to run TrixBox
in a Virtual Machine on my modest server

I'm now running many asterisk instances in virtual servers (the one I posted earlier is one such instance) although I'm using LXC containers - they seem to work extremely well.

Cheers,

Gordon

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