Notify me of new posts by email. Prepare the environment First, you should get something with Linux. Dial for Bob or for Joe. Whilst 'bob' and 'joe' could also be used here, numeric usernames are more common. Used in the Caller ID.
Asterisk™: The Definitive Guide
The password used to log in. In a secure system, you would use something other than ! Asterisk supports a few other account types, but SIP is the most widely implemented. The extensions which they can dial depend on this. Click the image below for an example: Linphone account setup. The Polycom web interface. Best of luck! Hello, This was clear, short and easy to follow.
Thank you Mike! I can only call one way, can call but not the other way. Thank you, it was clear!
I tester it with few granstream phones.. Regards, Daniel Loading Well, for starters all extension-to-extension calls are completely free and they do not use any external bandwidth, unless an extension is in a remote office. Also, depending on how many minutes you require each month you may be much cheaper with this option than using a hosted PBX, where you are typically charged per user extension.
As an example, let's assume a 5 extension hosted phone system with total long distance minutes per month from all extensions. If so read on and see if you are up for the challenge. Very little is actually required to get an Asterisk PBX up and running. An older PC running Linux, even if it is a few years old, will usually suffice as Linux can typically be run on lower performance CPUs, unlike Windows. I decided on this since I already use this for backing up computers, it runs Linux, it is always powered on and is pre-configured with the Digium Asterisk server v1. The goal of this article is not to show you how to install Linux and the Digium Asterisk PBX server, there are plenty of articles on the Internet already for this including the following useful guides and videos for installing Asterisk on CentOS and Asterisk on Ubuntu.
The only other hardware required is IP phones. Note the private IP address in the browser address window - this is my private network. We will work our way through the steps to get to this point but it is good to show what we are striving for here. It is easy to miss and can lead to frustration. The following steps are necessary in order to connect Asterisk to the outside world, using SIP trunks.
For this project I chose Flowroute simply because of the simplicity of its service and also it is pay as you go, so it is easy to load up a few dollars and start making and receiving calls. For a good list of options for trunking, visit our SIP providers section. The next step is to find out the specific credentials for your account so you can register this in Asterisk. In Figure 2 below you can see the credentials I gathered from Flowroute, with the actual authentication details blackened out.
Normally a business requires local DID phone numbers for its business. SIP providers usually offer this as a service where you can order a DID from any area code in the country.
Figure 3 highlights the DID number management section in Flowroute. This is required so it can be registered by the Asterisk PBX. The configuration is highlighted in Figure 4 below. At this point, if you followed these steps, you should see a green registered note when you click on system status. This indicates you are connected to the SIP provider's servers. If you are struggling to register, you may need to look into your firewall settings to ensure the SIP ports are being forwarded correctly between the Internet and your Asterisk PBX.
Configuration Guide for Asterisk PBX
In particular port should have a path through your network. Typically this is not an issue for most setups. Now we are registered with the SIP trunking provider, it is time to setup calling rules for incoming calls. In Figure 6 you can see how I configured outgoing calling rules for my extension.
In other words, when I use my extension and make an external call it will go through the SIP trunk with the caller ID name and number I setup. The dial plan is essentially a set of rules assigned to user extensions. In Figure 7 you can see that I have assigned this to the outgoing calling rule defined in Figure 6. All access has been permitted for directory, voicemenus, queues etc.
Time to create the users i. Figure 8 shows the settings for my first extension, which is extension Note the extension range can be altered in the global settings for Asterisk. Most of this should be self explanatory and if you hover over the question marks in each field, additional helpful information is presented.
Getting a SIP account
They present many concepts, but not with a story, an objective. I have based this book in the old training guides from Novell. So it has a start where you install Asterisk, then you create extensions, trunks, dialplan until you complete a fully functional free and open source PBX. Then we go to more advanced concepts. I hope you use this version for a long time. This book has more than 10 years, the first edition was in and since then it has been updated once each 4 or 5 years.
This book has two companions. A training on Udemy with the same name and a Lab Guide on github, more details inside the book. I sincerely hope you enjoy. Flavio E. Get A Copy. Kindle Edition , pages.
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- The Teaching Road Map: A Pocket Guide for High School and College Teachers;
- Configuration Guide for Asterisk PBX.
- The Asterisk GUI.
- Meditation For Dummies 4th Edition!
- AJS Review (The Journal of the Association for Jewish Studies), Vol 19 No. 2 1994;
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