Pbx setup

Fortunately, only moderately powered hardware is needed so the computing component does not have to be expensive. The preferred storage reserve for the system is GB, but this assumes use of call recordings and voicemails so it can vary quite substantially.

Ideally, this system should be the only OS that is utilized on the machine and it must always be running for your phone system to function. When installing the platform distribution, you will have the option to completely wipe the drive which is recommended or install it alongside another OS. In this guide, we will be using a FreePBX distro package that includes both a supported operating system CentOS and the main application.

This package is free and available directly from Schmooze Com. By navigating to the FreePBX download , you will find the current distro's - keep in mind, these will change over time as new versions are released. Select Full ISO from the stable 64 bit option for the best performance. You can burn the image natively with most operating systems but for Microsoft OS's prior to 7, you will need a 3rd party app like ImgBurn or UltraISO beware of packaged Adware with some hosts!

These are all good options, regardless of native burning apps since disc recording speeds may be lowered for more accurate media. Boot into the disc on the machine where you want to install the software. The first screen you see will ask you to select between an older or newer version of Asterisk and the method of installation as seen in figure 2. Assuming you chose the stable version of the distro as earlier indicated, you should choose the newest version of Asterisk and the Full Install - Advanced option.

Walk through the following steps to install the full system to your machine. It is recommended you use the whole hard drive as this is more secure and the machine must be up for the system to work. If you want to keep the existing system s on your machine, use the Shrink Current System. If a suitable chunk of unused space is already available, choose Use Free Space and select the desired space.

The system will proceed to install the software, reboot once then load up the welcome screen. Essentially everything you need to do from here will be done via a web browser so logging into the machine and poking around is not necessary, as this will likely do more harm than good, especially if unfamiliar with the command line for Linux. After the installation, you will be able to access the web management console from a browser on another machine within the LAN.

Type the IP address of the machine into your browser to get started. The first page you see should look like the one shown below in figure 4.

Enter the required information by creating an administrator name and password. For email to function properly, you will need to configure an SMTP connector.

In a Windows server environment, an admin can easily create an account for such purposes in Microsoft Exchange. Using a 3rd party email such as Google is also possible but doing it ideally requires use of Google Apps for Work. When navigating to the system via a web browser, you will land on a splash page with four main options. After logging in for the first time and each subsequent time you will land on the system status page as shown in figure 5.

Take a look at the notification area in the top left corner. One of the first messages you should see recommends changing the password for the Asterisk server as the default password is common knowledge. You can change this by going to the Advanced Settings option under the Settings tab as can be seen in figure 6 below.

This can save a lot of headaches in the future, trust me! Updating FreePBX modules i. To upgrade modules, visit the Admin tab and select Module Admin. Click the Check Online button to receive the most current information then check the box for Show only upgradeable.

Click the Upgrade all hyperlink then the Process button on the following page. A couple of things to note at this point: you can add additional modules from this point but it is not necessary as the required features for voice and video are already installed. The following steps do not necessarily require completion in the same order but if you are attempting to mimic my setup, it is best to complete in the same order.

Next we will configure the network settings inside the console as a measure to prime the system for connecting to our wholesale provider, VoIP Innovations. The first area we will configure is related to settings for the LAN.

This will provide functionality for your phones and softphones on the local network so if you are not planning on connecting a SIP account at this time, you can skip to the Configuring Extensions section. Navigate to the System Admin module found under the Admin tab at the top. On the right side of the page, you will see a box for additional sub settings. First, select the DNS and make the following changes as seen in figure 7. Order is not too important as long as a few entries are added to the box that normally do not populate automatically.

These addresses and the public DNS entries the two that begin with 8 are mostly for good measure - as long as your loopback address Make sure the rest of the information matches your local network, as viewed in figure 8. You will also need to find the external IP of your network - the easiest way to accomplish this task is to visit Google and type 'what is my ip. The codecs used for audio compression should be set based on the capabilities of your SIP or wholesale provider.

VoIP Innovations use codecs G and G G is not an option, but its variations ulaw or u-law - pronounced "moo law" - and a-law are available so these should be selected. G is a successor, in some sense, to codec G and GSM is often needed to communicate with mobile devices hence the reason both are selected. BONUS: For fun, you can turn on the video codecs located in the section just below the regular codec section. This will enable the use of video calling between properly configured softphones!

It is likely you will want to connect your PBX to the outside world. After all, it is tough to run a business purely on internal calls!

To do this you need either a wholesale or a SIP trunking provider. These providers essentially route your calls to their intended destination and in the opposite direction, assuming you own Direct Inward Dial numbers DIDs , they will also route calls to your phone system. Your newly setup on-premise PBX is then responsible for switching the traffic inside your network. We chose VoIP Innovations to handle origination and termination for this project, as they provide a great wholesale communication service, and their per minute rates are some of the lowest you can find in the market.

Make sure Save and Rebuild is the selected Action. You should now see a list of Available Firmwares. Setup may vary between phone models. Please refer to your specific phone's instructions. The following instructions apply to the Digium D Please see our End Point Manager wiki for more information. When you set up SIPStation trunks, a few basic inbound and outbound routes are automatically set up for you.

We will discuss inbound and outbound routes later. You won't need to give us a credit card, and you can get started immediately, as your trial service is provisioned instantly! With FreePBX version 12 and above, in just a few clicks we can have you up and running with your PBX connected to our world class services.

You just pick a phone number and we'll do the rest! There is no cost or obligation. You'll see just how easy it is to connect your phone system to our award-winning platform. There is no risk. After your evaluation, you can convert your account to a regular account no contract period required , or end the trial and go your own direction.

You will need an alternate means of calling in an emergency. Be sure to flip the card to add your security code. Step 4 — Read the Terms and Conditions statement and acknowledge that you agree by selecting the check box. A: This is mostly likely caused by Asterisk not reloading properly. Check your firewall for proper configuration. Many versions of Asterisk also require a restart for the new trunk hostnames to take affect. You can restart Asterisk with the bash command 'fwconsole restart' or by simply rebooting the PBX.

Simply make your selections, add the items to your shopping cart, and complete the checkout process. Detailed instructions are found below. If you are unable to find the city you are looking for, take a look at the list of rate centers to see if your city might be included under a different name. For example, Minneapolis and St. Paul, MN are both part of the "Twin Cities" rate center. You must continue to the next step and click Proceed to Checkout, or the DID s and toll-free number s in your shopping cart will not be saved to your account, meaning you will lose the ability to use those numbers if another user selects them before you check out.

If you need to remove any items from your shopping cart, you can click on the gray X next to the item. Review your Order Summary. The section "After Your Order, Your Subscription Will Include" section at the bottom of the page shows what the new monthly subscription fees will be if you make this purchase. Totals do not include taxes and fees, which vary by location. We are required to collect various taxes and fees according to federal, state, and local regulations.

Please see our wiki on Taxes and Fees for more information. Can't find what you are looking for in the store, or need some additional help with the ordering process? We may be able to help. You can use our special order form to make your request. You will use the store to find your account key.

Log into your SIPStation account. Note: SIPStation allows you to create multiple locations within the same account. Check the blue bar at the top of the page to ensure you are using the correct location. If not, click on the white triangle to bring up a list of other locations to choose from. You can set a global failover number here. Your Asterisk server would send incoming calls to the global failover number in case it is unable to connect to SIPStation. You can also see and change the status of international calling, outbound fax, and SMS services.

Notice that above your Metered Services section, you can also adjust failover settings here. Your E caller ID number is shown here. You may optionally set up e on additional numbers for an additional monthly fee. Please confirm the accuracy of the information and ensure E is working by dialing Configuring Your PBX top. When a call comes into your system from the outside, it will usually arrive along with information about the telephone number that was dialed also known as the "DID" and with the Caller ID of the person who called.

Calls come into your system on trunks that are configured in the Trunks module. The next steps are for SIPStation users who would like to create an inbound route to an extension. If you are not a SIPStation user, you can skip this section and go to the instructions for setting up an inbound route to an extension manually. Your new route should now show up in the Inbound Routes module.

If you are not using SIPStation, you will need to set up inbound routes manually. Please see our Inbound Routes Module wiki for more information. Without setting up outbound routes, you will not be able to make calls outside your system. You should see these defaults:.

You can modify the settings as needed or create new routes. See the examples below for typical configurations. If you are not a SIPStation user, you can set up outbound routes manually as described in the instructions below.

A typical configuration will include an emergency route for calls and another route for ordinary calls. You might also wish to configure routes for interoffice calls, international calls, and other special circumstances. Creating an Emergency Outbound Route Manually. In our example we are going to create a single route for local and toll-free numbers, combined.

When making an outbound call, the system will for the first matching dial pattern by working down the list of routes. The first item on the list will be checked first, then the next, and so on.

Keep this in mind when deciding the order of your routes. Please see our Outbound Routes Module wiki for more information. A ring group is a list of multiple numbers that you would like to ring when a call is received. This gives several phones the opportunity to answer a call. You can choose from various ring strategies to control the order in which the system rings different phones. Please see our Ring Groups Module wiki for more information.

You can set up automatic call distribution with the Queues module. A queue differs from a ring group because it allows advanced call routing options and escalation rules. This wiki gives you a brief overview of some basic settings, but there are many more. To learn more about queue settings, you can use the pop-up tooltips in the module.

Please see our Queues Module wiki for more information. It is most commonly used as an IVR option. Your directory is not limited to internal extensions. You can add custom entries such as remote extensions, ring groups, queues, and outside numbers. You can add as many directories as you would like. In order for the directory to work, you will need to give callers the option to dial it i. These options are controlled within the Inbound Routes module and elsewhere in other modules. Please see our Directory Module wiki for more information.

Modules such as Interactive Voice Response IVR are able to use custom system recordings in addition to default recordings. You can use the System Recordings module to create and save custom system recordings.

There are two ways you can create a system recording: by speaking over the phone, or by uploading an audio file from your computer. After you have created a recording, you can update it by uploading a replacement audio file or by making a new recording over the phone. You can enable a link to a feature code that will allow users to re-record a system recording over the phone.

The default password for these extensions are , and Holly use miniSipPhone as her softphone. Please click 'OK' button to complete miniSipPhone configuration. It will try to register to miniSIPServer. If it successes, miniSipPhone should display telephone number and be ready to make calls. We can follow the same step to configure G. T' extension. Both Holly and G. T' extensions have been connected to miniSIPServer.

We can show miniSIPServer' local user information to check their status. Their icons should be blue. After we finish this step, the basic VOIP network is established. Holly and G. T can call each other. Holly can dial '' to call G. T, and G. T can also dial '' to call Holly. In above configuration, we use default extensions '' and ''. In future, with the growth of company, more and more people will join with us, we need add more extensions to support them.

So we can do it as following:. In normal, we can just assign extension number and password to a new extension. T enjoy it. It is time to establish connection with customers now. In the miniSIPServer main window, please click button 'External lines' to add an external line information. In the pop up window, please click button 'Add' to add an external line with CallCentric account information.

The key items are described in below table. Of course, you can update it according to your own configuration. Some VoIP providers require different authorization numbers with their accounts.

In this scenario, we must configure "authorization ID" with such numbers. By default, authorization number is same with voip account, then it is unnecessary to configure "authorization ID" item or configure it as same as voip account.

Because we hope both Holly and G. T can make outgoing call, we select 'All local users can use this external line to make outgoing calls'. If the external line success to connect to peer server VOIP provider's network or VOIP gateway , the icon of the external line should be gray and without cross flag.