CONFIGURATION GUIDE FOR ASTERISK PBX PDF

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After all, it is probably the easiest way to reduce your business phone bills and there is no hardware or software to maintain. Having said that, with some basic computer skills, it really isn't that difficult to setup an on-premise IP-PBX phone system in your office to handle all of your business phone calls.

It doesn't matter if you are a small business or a large business, the steps are relatively straightforward. 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. 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.

The most important aspect is setting up the MAC address for your device. We will configure the SPA later with this information. Retrieving the MAC address from your IP phone is relatively straightforward but does vary per phone so check the users guide for your phone.

For the SPA you can get the MAC address by logging into the phone using a web browser find the address in the network settings on the phone, usually over DHCP and looking at the Info tab and then the product information. Although beyond the scope for this article, take a look at the features you can set up in Asterisk.

It really is a feature-rich phone system. For example, I set up an auto attendant using voice menus and voice prompts so the caller can hear a menu such as press '1' for sales, '2' for customer service etc. This really does give your business the appearance of a larger enterprise - for more information see our article on auto attendants. Other useful features to explore include ring groups, music on hold, call queues, voicemail to email, directory, follow me and conferencing.

They are all relatively simple to set up. Interestingly, I found configuring the phone more complex than configuring the Asterisk PBX, mainly due to the lack of help for each parameter - thanks Cisco! I connected the SPA as per the users guide that came with the phone i. Once initialized, press the setup button, then the Network entry. This brought up the SPA configuration utility, and when the System tab is selected it should be configured as per Figure 9 below.

Also note below that this is in advanced admin mode. This is required because otherwise I would not have access to all the parameters needed. The settings for the extension are highlighted in Figures 10, 11 and 12 below. The important elements here are that the SIP port is , the proxy is set to the IP address of the Asterisk server and the User ID and password are set to be the same as that for the user in Asterisk i.

Figure 8. This registers all of the information and resets the SPA If you are successful then the light should turn green on the SPA and if you refresh the System Status in Asterisk, the phone s should turn green in the extensions area as per Figure 1.

If you have followed along to this point then you should be able to make and receive calls successfully. When I pick up the SPA I hear a dial tone and can dial outside lines don't forget the '1' for long distance as this is the way the system has been configured and expected by Flowroute. I set up multiple phone extensions and I was able to make and receive calls to other extensions. The beauty here is that all of these calls are free since they are all routed by the Asterisk PBX on my internal network.

No charges are made by Flowroute since the trunk is not used. Want extra credit, set up a softphone on your smartphone and use it as an extension. You have done well to reach this point so the softphone guide will be left for another day.

Configuring the Bria was relatively straightforward and worked well when in WiFi mode since it was using the same private network so there were no firewall issues.

I could perform extension dialing through Asterisk and make and receive calls to the PSTN through the Flowroute trunks. However when in cell phone mode it was a little more problematic and the actual audio packets were not getting through, though it was registering with Asterisk.

Update : You can find a brief Bria softphone configuration guide here. The Asterisk PBX is a great money saver for business phone bills and not just for large businesses. With a little computer knowledge, any sized business can get this free software installed on a computer and have it up and running with phone calls in just a few hours. There is no doubt that a hosted PBX system in the cloud is simpler to use, since the hosted providers take care of the complexities for you, but you will pay a premium for this service.

This premium is small when compared to landline costs but an on-premise Asterisk PBX can result in even more savings, if you are willing to maintain the system in your office. Select number of employees: 1 to 4 5 to 10 11 to 20 21 to 50 51 to More than Published by WhichVoIP. All comments will be moderated by WhichVoIP. We try to answer all questions within 24 hours.

By submitting you agree to our site Terms of Use. It is really a function of what else you have running on the NAS, amount of memory and CPU resources so it is tough to give a true estimation without running a load test.

If you need to know the limits you could replicate this test. I'm having trouble getting incoming and outgoing calls to work. Everything seems to be configured properly, as the trunk and Pi2 see each other. I setup my trunk and outgoing settings similar to you, but I'm not able to get calls to work. Any ideas? Is your PBX registering successfully with the trunk provider? If you set up 2 extensions inside your LAN can you make extension calls successfully? The answer to these questions should help get to the bottom of the problem.

Often these issues are related to NAT or firewall related problems especially if you can make internal extension calls successfully. Good guide, thank you. Personally, I have always found the Cisco phones to be tougher to setup. We are preparing your quotes. Get First Month Free! Save Time: Get multiple quotes to compare. Trusted: Over , quotes processed. How many employees use the system: Select number of employees: 1 to 4 5 to 10 11 to 20 21 to 50 51 to More than Just a few more questions

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