sysadminman.net

I saw the first ‘externsion scan’ of my Asterisk box this week. That is, an external server tried to register as an extension, starting at extension 100 all the way up to extension 999. I’m assuming if they had found a valid extension number then this would have been been followed by a brute force password (secret) scan.

This is an interesting article explaining the problem a little more - http://michigantelephone.wordpress.com/2008/11/28/why-didnt-freepbx-developers-implement-important-security-patch/

If you’re running Asterisk (and FreePBX) then the least you need to do is make sure that you’ve got pretty strong passwords for your extensions.

If you’re based in the UK or Europe and looking for a cheap ITSP (VIOP provider) it might be worth looking at CallWithUS as they’ve recently launched a UK based SIP server.

As well as the US based servers sip.callwithus.com, east.callwithus.com and west.callwithus.com you can now use uk.callwithus.com. I now get sub 6ms pings from my Asterisk server in BlueSquare to the CallWithUs server.

I’ve been using CallWithUs for a while now and they provide very competitively priced DIDs and termination rates.

Sysadminman is now offering UK based Virtual Private Servers running Trixbox and Elastix.

Please see here for details - http://sysadminman.net/uk-voip-vps.html

Here is my CallCentric configuration for FreePBX.

If you’re thinking about signing up with CallCentric please use my referral link here. Thanks.

Please note: the above number starting 1777 is your account number and not you DID number

You also need to add 2 lines to one of the configuration files to correctly extract the DID number from incoming calls.

Edit /etc/asterisk/extensions_custom.conf

Add the following lines

Then restart Asterisk

You should now be able to create Inbound Routes based on you CallCentric DID numbers

I’ve been using CallCentric as one of my SIP trunk and DID providers for several months now and it has worked really well. If you use FreePBX (or one of the prebuilt distributions that use it such as Trixbox, Elastix or PBX-in-a-flash) and use this link to sign up for a free CallCentric account I will configure your trunk in FreePBX for free.

Please use the contact form here if you would like to take up this offer.

Some of the services offered by CallCentric are -

• US DID numbers from only $2.95/month
• DID numbers in cities all over the world
• US calls at only $0.0198 per minute
• UK landline calls only $0.0198 per minute
• Unlimited calling or pay-as-you-go plans

I’ve used Sipgate for the past few years with my Asterisk box and have been pretty impressed.

For anyone else looking to use Sipgate with Asterisk/FreePBX here is my trunk setup

If you are using NAT between your Asterisk box and Sipgate you will need “canreinvite=no” and “nat=yes” or you will probably get one way audio only on your calls

Edit: context changed to “from-trunk” to make the post clearer

Sysadminman is now offering UK based Asterisk VPS servers that have Asterisk, FreePBX and A2Billing installed and ready to use. These VPSs are hosted in one of the premier datacentres in the UK (BlueSquare) and have low pings to UK and the rest of Europe.

Asterisk and FreePBX combined make an extremely flexible and easy to manage “virtual telephone system” and A2Billing is a billing application enabling you to charge for calls made using your system if required.

Some possible uses of an Asterisk VPS are -

• Small businesses wanting to manage a collection of SIP extensions
• Reselling calling cards or VOIP termination
• People who do a lot of traveling and want to stay in touch wherever they are
• Route calls based on time of day, DDI, CLID, etc…
• Provide contact telephone numbers in cities around the world all routed to your local phones over the internet
• Voicemail to E-mail

See here for more details and have you have any questions please get in touch

Matt

If you make a lot of international phones calls, or even if you don’t make a lot of them but need to be flexible in how and when you make them, running an Asterisk server could be the way to go.

Asterisk is a OpenSource telephone system that runs on top of Linux. Asterisk can be quite difficult to manage as it involves editing text based configuration files. If you don’t want to do this you can install a web based management interface called FreePBX. This provides lots of management functionality from a nice web based GUI.

Here are just some ideas for how you could use your own phone system -

Cheap calls abroad from your cell phone

Asterisk/FreePBX provides a feature called DISA. This enables to dial into your Asterisk server (via a local access number available from an ITSP such as CallWithUs) and then dial back out again (again via an ITSP). This would enable to call anywhere in the would at a local rate, right from your cell phone.

Continue reading ‘Cheap international phone calls with Asterisk’ »

The instructions below assume that you have got Asterisk, FreePBX and A2Billing installed and working together.

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 get a lot of people asking me about running Asterisk on a VPS. It’s a great way to get started and experiment with Asterisk and find out what it can do.

If you’ve used something like Skype or Vonage before then you’ll have an idea what you can do with VOIP but running your own Asterisk server makes things so much more flexible.

If you combine Asterisk with FreePBX, a great web interface for configuring Asterisk, then you’ll have an extremely capable PBX. There are many plugin modules available for FreePBX which provide an easy way to setup advanced features such as voicemail, IVRs, follow-me, time conditions, conferencing …

It is possible to run Asterisk on pretty much any VPS but you’ll definitely have a better experince using a VPS specifically designed for running Asterisk. A couple of features that may not be available with a ‘normal’ VPS is the ztdummy driver and high quality bandwidth.

Continue reading ‘Asterisk on a VPS’ »