Below are the first steps in setting up a callwithis DID number and passing the call through to A2Billing. This number can then be used as an access number for your calling card clients.

Once you’ve signed up for your callwithus account and purchased your DID number the next thing you want to do is modify how your DID number gets presented to your Asterisk box so you can route it to a2billing.

On the callwithus website, click on DID on the left hand menu and then locate the DID number you want to use and click ‘edit’. Now under ‘DESTINATION’ add ‘/yourdidnumber’ the the end of ‘SIP/youraccountnumber’

This will cause callwithus to append your DID to the call details when it is passed through to your Asterisk server. See the image to the right for details.  

Next we want to go into FreePBX and register A2Billing as a ‘Custom Destination’.

If you don’t have the ‘Custom Destination’ menu in FreePBX you will need to go to the modules menu option and install the ‘Custom Applications’ module.
Enter ‘a2billing’ as the Description and ‘custom-a2billing,${EXTEN},1′ as the custom destination

Now we want to configure our callwithus trunk.

Configure your trunk as in the images here, obviously using your username and password. More information on how to setup your trunk can be found on the callwithus website here.

Now we just need to configure our inbound route. This inbound route decides what happens to a call when it is presented to your Asterisk server. Give it a meaningful description and then in the ‘DID Number’ box enter your callwithus DID in the same format you used in step one above.

Scroll down to the bottom of the configuration page and you should see the ‘a2billing’ ‘Custom Application’ that you setup earlier. Select this and then click on Submit. Don’t forget to click on ‘Apply Configuration’ at the top of the FreePBX page to get Asterisk to read your changes.

And that’s it for now. If you call your callwithus DID it should come to your Asterisk box and then be passed to a2billing. Depending on how you’ve got a2billing configured will determine what happens to that call now.

I’ll discuss in a later post some ideas about how to configure a2billing

I thought I’d post my firewall script to help get you started. I save this script as /usr/local/bin/firewall.sh and then add a line to run it from /etc/rc.local

It allows SSH’ing to the machine plus rules required for SIP connections (you will need other rules if you use IAX) plus some basic “bad stuff” filtering.

I’ve commented it so, hopefully, you’ll be able to figure out and chages you need.

Redhat5/CentOS5 cetainly has a firewall enabled as default. Try ‘iptables –list’ to see if you’ve got iptables running

The best way to do this is use an external monitoring company. I’ve been using Hyperspin for a couple of months now and it’s been working great.

You can pick what services/ports on your server to monitor (SMTP, HTTP, HTTPS, etc.) and you can chose how to be alerted when they are not available. Obviously you need to make sure that you’re alerted via an out of band method (in other words don’t rely on an e-mail to an e-mail account on your VPS to know that your VPS is down!)

The Hyperspin website isn’t very pretty but overall their system is reliable and easy to use.

This might not be noticeable for web browsing or e-mail but if you’re doing any sort or VOIP or gaming it can be critical to know the quality of the connection between 2 points.

A simple tool for this is mtr. This is normally included in modern Linux ditros – if you’re a RedHat/CentOS/Fedora person just run -

Now you can fire it up by running -

So, we can see to www.digg.com from one of my servers I’m getting a small amount of packet loss and a latency of 82ms.

And below is an mtr to voicestick.com. If this connection was being used for voip, even though the ping time is better than the one to digg, the quality would probably be awful due to the packet loss.