Category Archives: FreePBX

Asterisk web management GUI

FreePBX 2.11 Guide: Local number dialling

When using VOIP to place telephone calls our call provider has no real concept of our exact geographical location, we could be anywhere. So there’s no ‘local’ area code by default.

In this set up guide we are using Voiptalk to place our external calls. As they are a UK provider they will accept UK numbers dialled in the regular national format. So if I want to dial a Northampton number (with the area code 01604) I can call “01604 838383”. But if my telephone users are based in Northampton they may be used to just dialling “838383”

We can implement this by manipulating a number dialled in to the format expected by the call provider. We are going to do this by setting up “Dialed Number Manipulation Rules” on the “voiptalk” trunk we created earlier.

The rule we want to create is –

  • Any 6 digit number dialled that starts with a 7 or an 8 gets 01604 added to the front before being sent to the call provider

Adding the manipulation rules

Dialed number manipulation rules

Dialed number manipulation rules

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FreePBX 2.11 user guide

This guide will be added to over time and is designed for new users to FreePBX.

System access, security and maintenance

Basic Calling

Advanced features

more guides to follow ….

FreePBX 2.11 Guide: Outbound calling

To make calls to a regular telephone we need an account with a call provider. Our FreePBX system will pass the call to the call provider (using SIP) and the call provider will pass the call to the regular telephone system. This is sometimes referred to as ‘call termination’

There are literally hundreds of SIP call providers to choose from. This guide uses voiptalk.org who are a large provider in the UK. First you need to sign up with the call provider and deposit some call credit with them. You will then receive SIP credentials which you will use to configure your connection (SIP trunk) to them.

They should also provide you with some configuration settings that you need to enter. Because FreePBX/Asterisk is so popular many providers will be able to let you know what settings to use. The settings can vary depending on the call provider.

There are 2 steps to this. First we need to set up the connection (SIP trunk) then we need to tell FreePBX what calls to send via that trunk. Don’t forget we could use many different call providers (possibly for different call destinations). We need to let FreePBX know what trunk to use for what calls.

Set up SIP trunk

Select Connectivity / Trunks form the menu

FreePBX SIP trunk

FreePBX SIP trunk

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FreePBX 2.11 Guide: SIP phone setup – Yealink T22P

Setting up a SIP phone to connect to our FreePBX extension is fairly straight forward. All phones set up are a little different, and you can normally do the config through the phone itself, or a web GUI. Normally the web GUI is easier but below is a guide to basic configuration of a Yealink T22P handset. This is a good choice for a phone, and relatively inexpensive (although it seems no SIP phones are really inexpensive!) Setting up a softphone would be similar.

There are lots of screenshots below, but we only really need 3 bits of information –

  • Server IP Address
  • Extension Number
  • Extension Password

Note: SysAdminMan servers run a security application called fail2ban. This monitors for incorrect registration attempts and blocks the IP if it sees too many incorrect attempts. While setting up your extensions it’s possible (likely!) that you will not get it correct the first time, so open a support ticket asking for your home/business IP to be whitelisted, so it doesn’t accidentally get blocked.

Here’s what I did to set up my phone and make a test call …

Enter the phones settings menu

Yealink T22P

Yealink T22P

Yealink T22P

Yealink T22P

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FreePBX 2.11 Guide: Logging in and password change

After you receive the login details for your new SysAdminMan FreePBX virtual server the first thing you are going to want to do is log in and change the administrator password.

For extra security SysAdminMan servers use both Apache and FreePBX security to restrict access. Both these access methods use the same account details.

Also all SysAdminMan server use HTTPS to access FreePBX. Because this uses a locally signed certificate you will get a warning when you first connect. This is normal and you should only see it once. Once the certificate has been accepted you should not see the warning. If you’d like more information on what a locally signed certificate is please see here – http://en.wikipedia.org/wiki/Self-signed_certificate

Browse to the URL in your welcome e-mail

Accept the web server certificate

Accept the web server certificate

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FreePBX exploit (ticket 7123 – 6/2/14)

Earlier today Schmooze announced a critical exploit in FreePBX – http://www.freepbx.org/news/2014-02-06/security-vulnerability-notice

SysAdminMan systems come deployed with Apache security in place to prevent direct access to the FreePBX web pages and are not directly affected by this. However, it is recommended that all customers update to a non vulnerable version of the FreePBX framework module.

If you have a FreePBX system that is generally accessible from the internet with no additional security (not recommended) you should check the details of the exploit and update now.

Backing up FreePBX

This post was written by a SysAdminMan customer. If you’ve got some FreePBX or A2Billing tips you would like to share please get in touch.

We are a small company and like everyone else busy.  So far we have not backed up our FreePBX settings, there always seems something more urgent to do.  However I am conscious that we backup every other piece of information we hold so we should backup our phone settings and voicemail. This is what we did:

1. Go to top left menu option “Admin”

2. Select “Backup & Restore”

backupmenu

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Asterisk and A2Billing software update

The software versions on the SysAdminMan templates have been updated as follows …

SysAdminMan FreePBX

  • CentOS 5.10
  • Asterisk 11.6
  • FreePBX 2.11

SysAdminMan FreePBX & A2Billing

  • CentOS 5.10
  • Asterisk 11.6
  • FreePBX 2.11
  • A2Billing 2.0.7

SysAdminMan FreePBX & A2Billing & OpenVPN Server

  • CentOS 5.10
  • Asterisk 11.6
  • FreePBX 2.11
  • A2Billing 2.0.7
  • OpenVPN Server

All new servers will be deployed with the software versions above. See here to order – https://sysadminman.net/uk-voip-vps-order.php

Using iPhone Media 5 as a FreePBX extension

I wanted to setup extensions on my Free PBX system. This will allow me to transfer calls to different users, fantastic. I am not keen on spending extra cash in extra handsets, I am keen to use mobile phones we already have as our extensions, I can only talk to one person at a time. The technical word I need is soft phone. I can also use these types of phone through a laptop too.

Before I used the soft phone, I found the blog post Using Android with FreePBX – a SIP extension for free really helpful

We have iPhones I am advised that the. X-Lite is great on Android phone and PC but it won’t work in background on iPhone.  By background, I mean when the app isn’t active – so potentially no incoming calls.

Instead we are using Media5-fone Pro VoIP SIP Mobile Softphone, it is the unique SIP Softclient with Wideband and HD Voice Audio Quality for the iPhone/iPad/iPod Touch, it requires Requires iOS 4.3 or later. The Media5 Facebook page has up to date information about the product too.

You install the Media5 fone on your iPhone like any other app. When you open the app it looks like you are using a normal handset, except the Media5 phone is linked to your FreePBX. You can still come away from the app and use your mobile phone as usual.

Linking the Media5 fone to the FreePBX was easy. I had to enter my username (extension) and password / secret and was away.