SYSADMINMAN

Last December the IC3 issued an alert for Asterisk users whch can be seen here.

This initially caused a panic amongst the developers as it wasn’t really clear what the alert was about. It turns out that it was for a vulnerability that was indentified and patched by Digum 9 months earlier. IC3 issued an updated buliten shortly after describing the issue a little better which can be seen here

I’m still seeing this alert being used to try and discourage people from using Asterisk but as far as I can see it’s just a normal security warning that was quickly identified and fixed by the software developer.

If you’d like to read more information there’s a good post here regarding this – http://blog.tmcnet.com/blog/tom-keating/asterisk/digium-responds-to-fbi-vhishing-security-warning-about-asterisk.asp and, as always, keep your software patched!

DISA is great!

It allows you to call in to your Asterisk server, get a dial tone, and then dial back out as if you were using a normal extension on your system. I use this lots to make cheap international calls from my mobile phone.

You may also wish to route your DISA calls via A2Billing. If you’ve integrated FreePBX and A2Billing as described here it’s a simple case of changing one setting on your DISA setup in FreePBX.

Continue reading ‘Using DISA with FreePBX and A2Billing’ »

There are several reasons you may want to integrate FreePBX and A2Billing. Whether you’re just using FreePBX to setup trunks for your a2billing calling card system or you use FreePBX and want to route the outbound calls via a2billing to do least cost routing.

There are 2 things you need to do to integrate the two. The first is to add the following to extensions_custom.conf

Continue reading ‘Integrating FreePBX with A2Billing’ »

A2Billing is a great piece of call billing software for Asterisk. It can be integrated with FreePBX and used in lots of different ways.

I run my own Asterisk system with FreePBX and use a2billing to do least cost routing. It’s possible to import rate tables from many different voip providers and then let a2billing route the call based on the cheapest route.

One of the things you need to be aware of is that a2billing will route based on the ‘best matching’ rate. So, lets say you are trying to call +17061234567 and you have rate cards for 2 providers. If provider_a has a rate for +17061 at $0.02/min and provider_b has a rate for +1706 at $0.01/min, a2billing will choose the more expensive provider_a as the rate is a better match. This could require some manipulation of the rate tables to get things to work how you want.

As I said a2billing can also integrate with FreePBX so that you can pass all outgoing calls from FreePBX to A2Billing to allow it to do the routing. You also get the benefit of better CDR reports.

So, if you’re looking for something to do least cost routing for Asterisk then it might be worth checking out a2billing!

If you’re using sip to connect to callwithus then you need to make sure that you’re using the generic address sip.callwithus.com as your proxy/registrar.

From May 1st the other, more specific addresses, that used to work will stop accepting registrations (west.callwithus.com, east.callwithus.com, uk.callwithus.com)

See here for further details – http://www.callwithus.com/configuration

I spent a little while playing with sipvicious today. This is a SIP scanner that can be used for scanning SIP servers – which obviousy includes Asterisk, Trixbox, Elastix, etc…

It’s not surpising that scanning for vulnerable SIP servers is on the increase – these sort of tools are really easy to use, and with the lure of making free phone calls at your expense it’s definitnely worth making sure that your PBX is secure.

Here’s what I did to scan one of my servers. The server is a Trixbox CE 2.6 server and I set up the following extensions for testing -

Continue reading ‘Hacking and securing your Asterisk server’ »

Our Asterisk Only VPS template has been updated to run Asterisk 1.6 and CentOS 5.3.

This VPS comes with the Dahdi dummy driver compiled meaning you can run IAX2 trunks and Conferences if required.

For more details on all Sysadminman VOIP VPS plans click here

I’ve added Future Nine to the list of servers whose ping time I monitor. Even though they are a US based company they have a sip server in Europe and incoming numbers in the UK for only $5/month.

So if you’re in the UK/Europe and looking for a voip provider with competative prices it might be worth checking them out.

You can see the current ping from my servers to theirs here – http://sysadminman.net/uk-voip-vps-pings.html

If you’re looking for a softphone to use with Asterisk X-Lite is great.  It works on both Windows and Linux, although the configuration screens are a little different on the different versions.

All you should need to get it working with Asterisk are the following settings  (screenshot from the Windows version) -

Continue reading ‘SIP soft phone – using X-Lite with Asterisk’ »