Part 4 – Create an outbound route in Elastix

This is part 4 of a series of posts on setting up an Elastix extension with A2Billing. See here for details of the other parts – Using A2Billing to account for extension calls in Elastix

Next we need to create an Outbound Route to tell Elastix which trunk to use for our calls.

Click on Outbound Routes on the left hand menus and then create a new Outbound Route -

Continue reading ‘Part 4 – Create an outbound route in Elastix’ »

Part 3 – Create a trunk in Elastix

This is part 3 of a series of posts on setting up an Elastix extension with A2Billing. See here for details of the other parts – Using A2Billing to account for extension calls in Elastix

Now we have got an extension set up we need to be able to make calls to the regular telephone system. To do this we need an account with an ITSP, I’m going to use callwithus.

First click on Trunks and then Add SIP Trunk -

Continue reading ‘Part 3 – Create a trunk in Elastix’ »

Part 2 – Set up x-lite to use your Elastix extension

This is part 2 of a series of posts on setting up an Elastix extension with A2Billing. See here for details of the other parts – Using A2Billing to account for extension calls in Elastix

X-Lite is a free SIP client and can be downloaded here. There are many others!

Once you get x-lite installed bring up the “SIP Account Settings” page -

Continue reading ‘Part 2 – Set up x-lite to use your Elastix extension’ »

Part 1 – Create a test extension in Elastix

This is part 1 of a series of posts on setting up an Elastix extension with A2Billing. See here for details of the other parts – Using A2Billing to account for extension calls in Elastix

Log in to Elastix -

Continue reading ‘Part 1 – Create a test extension in Elastix’ »

Using iPhone Skype over 3G with Asterisk and FreePBX

Skype have recently updated their iPhone application to work over 3G. When combined with Skype For Asterisk from Digium it is now possible to call in to your Asterisk server for free over 3G. Unfortunately though this will only last for a short time as Skype are planning on charging for Skype to Skype calls over 3G (I’m not sure how the mobile operators feel about this as they will be the ones losing out on the call revenue!)

Skype on the iPhone is not really useful for receiving calls as, unless the Skype application is running, it is not logged it. Roll on iPhone 4 and multi-tasking!

Once the Skype for Asterisk software is installed and configured it is possible to create Inbound Routes in FreePBX as normal to route the incoming Skype calls to their destination -
Continue reading ‘Using iPhone Skype over 3G with Asterisk and FreePBX’ »

VOIP VPSs to customers in over 15 countries

I’m still occasionally surprised by the global nature of running a business on the Internet.

Sysadminman now has Asterisk VPS customers in over 15 countries including Argentina, Belgium, Canada, Switzerland, Spain, France, Italy, Netherlands, Poland, Portugal, Thailand and the United States, with most customers based in the UK.

Asterisk conferences with multiple DDI numbers

One of the most useful features of Asterisk is the ability to have multiple geographic or non-geographic telephone numbers route to your Asterisk server. These numbers can be regular telephone numbers, based anywhere in the world.

Another feature that many people use are conference rooms. Combining these 2 features it’s possible to host conference calls accessibly by both internal extensions and callers dialling regular telephone numbers.

To demonstrate this I have set up a conference room that is accessible by dialling one of 3 regular phone numbers.

Try for yourself by calling -

  • 0121 279 0080  (Birmingham)
  • 0161 353 0126  (Manchester)
  • 020 3298 2321  (London)

Calling any of these numbers will put you through to the same conference room. Usually there would be a user and admin pin number for the conference room but there is none for this demonstration.

The conference call is hosted on a Sysadminman VPS.

For more information on setting up a conference call in FreePBX see here.

Demo system now running FreePBX 2.7 and A2Billing 1.6

The sysadminman demo system is now running FreePBX 2.7 and A2Billing 1.6.

Please see here for access details – http://sysadminman.net/blog/2009/live-demo-freepbx-2-5-and-a2billing-1-4-vps-909

Limiting SIP/IAX connections to Asterisk with IPTables

WARNING: be very careful when editing IPTables firewall rules. It is relatively easy to completely disable access to your machine.

All Sysadminman VPSs come with IPTables enabled. However to allow for VOIP traffic both SIP and IAX ports are opened.

If you know that your VOIP providers and all extensions are on fixed IP addresses then it is possible to limit connections to just those addresses.

Continue reading ‘Limiting SIP/IAX connections to Asterisk with IPTables’ »

Namecheap SSL certificate for Sysadminman VPS

A sysadminman template VPS comes already setup to use SSL (https) for web connections to a2billing and FreePBX. However, this is using a locally signed ssl certificate so you will receive a certificate warning when accessing your VPS. This is no less secure but can create a poor impression depending who will be accessing the site.

It’s relatively straight forward and inexpensive to get yourself a valid, externally signed, certificate.

The sysadminman template uses lighttpd as the web server so you need to follow these instructions -

Log in to your VPS as root:

[root@livedemo /]#

Continue reading ‘Namecheap SSL certificate for Sysadminman VPS’ »