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First steps: Setting up your hardware infrastructure and configuration.

The center component of a hardware infrastructure designed to offer Unified Communications is the server. It is a physical machine with an Internet connection to which all computers and phone terminals are connected. Linux (CentOS and Rhel are supported by VoipNow) and your PBX should be installed on this server. Its hardware specifications rely on the network capacity and on the expected performance.

There are various methods of benefiting from a server.

  • in-house: If you already own an adequate server you may install an IP PBX and are ready to provide Unified Communications
  • hosted: Your server can be held within a data center. It is highly recommended to choose a data center that is geographically close to most of your customers as the quality of service relies on this.
  • in the cloud: There are some organizations that offer infrastructure as a service. For instance, you can rent a server from Amazon.com.

You have to choose your hardware configuration to meet your network's needs. You can decide on the performance you need your sever to be capable of by considering the maximum number of concurrent calls you expect to be made on the server. The general formula is that for every concurrent call (with a ulaw codec - G711), you server will use up

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Each concurrent call = 33MHz CPU + 8 Mb of RAM memory + 86 Kb of your network's bandwidth

How do the extensions affect the servers?

Another way of figuring out what you server will have to stand up against is the total number of extensions on it. For example, if your server will provide VoIP for residential clients, most extensions will probably be phone terminals that do not require many resources. On the other hand, if you resell towards businesses, IVRs, Queues and other costly extensions will need a more powerful hardware configuration.

How does the server handle PSTN?

Another issue that needs to be addressed is that of connecting your server to a public telephone network. This becomes of your concern especially when your server is in-house. To connect to such a network you will need a physical gateway that routes and translates calls between the two networks for the VoIP user. The telephone company provides one or more trunk lines to the customer for connection to the customer's PBX.

What about codecs?

Codecs are designed to enhance VoIP calling sound quality. However, these codecs influence your server’s or your network’s performance.
As an example, G711u and G711a do not compress packages upon sending them. This means that your call quality will improve, but they consume bandwidth. On the other hand, if you use G729 – which is one of the most used codecs – the packages will be compressed and your call quality will be amplified, but this action consumes a greater amount of CPU.
A fair conclusion is that you should always calculate your system to have at least 30% resources free most of the time, so that you won't have to worry about peaks.

Getting the new hardware to do what you need: Setting up your software.

VoipNow is a platform that includes an IP PBX, billing engine, a browser interface and multi-tenant management. At the core of the VoipNow system lays the IP PBX, or Internet Protocol Private Branch Exchange, which is a private telephony network used within a company. It is a software application that implements the basic functions of a traditional hardware PBX as well as next generation digital services. Basically, it acts like an automatic switchboard that connects users. In addition, it offers new services that traditional telephony doesn't have the technology to.

One concern: Am I safe?


A major worry for most VoIP users is whether it is prone to hacking or not. Internet telephony may seem liable to hacking at first. In fact, all internet solutions are. The risks of using VoIP do not exceed those that have already been accepted by businesses using the internet as part of their activities.

The truth is that open source PBXs tend to have a deficit in security. On the other hand, VoipNow has not been blamed for a hacking incident in the whole 5 years that it has been used. Troubles have come up that implied significant money loss for a few extensions. They were either caused by not being careful in keeping their password private or by using other PBX software besides VoipNow. The hacker was able to get inside the system through these and it affected VoipNow accounts, also.

Call your colleague: All you need is a phone.

While most features of Unified Communications solely require a computer with internet access, VoIP calls are a little more pretentious. You will surely need a phone terminal to be able to make and receive calls. There are a few possibilities to choose from.

One of them is to install a softphone on your computer. There are various softphones available online and some are free of charge. By installing this software, you can use your computer to make and receive calls as long as the program is running and provided that you have headphones or speakers and a microphone.

For a more familiar experience, you can purchase IP phones or even use regular phones. The main function of the phone terminal is to transform analog audio signal into digital and vice-versa. IP phones are built to do this, but regular phones can also be connected to a VoIP network using a VoIP adapter.

Get in touch with the outside world: Choosing SIP carriers.


After setting up your internal hardware infrastructure, you can make VoIP calls, but only inside your local network. To connect to the public telephone network, the services of one or more SIP carriers are needed.

A SIP carrier is a VoIP carrier whose equipment supports the SIP (Session Initiation Protocol) standard. SIP is currently the most popular VoIP protocol on the Internet, used to connect millions of networks and devices.

SIP carriers operate an infrastructure that can route and connect the calls originating from your system to phone numbers on other infrastructures (IP networks or PSTN). They own media gateway machines that act as an interface between different types of networks by converting between the different transmission and coding technologies.

The secret is to hear and be heard: What is DID?

With all the characteristics described above, you are able to establish VoIP calls. However, in order to receive calls you will need DID or Direct Inward Calling. This feature can be offered by your SIP provider.

The telephone company provides one or more trunk lines to the customer for connection to the customer's PBX and allocates a range of telephone numbers to this line (or group of lines) and forwards all calls to such numbers via the trunk. To reach users with VoIP phones, DID numbers are assigned to a communications gateway connected by a trunk to the public switched telephone network (PSTN) and the VoIP network.

Calls originating in the VoIP network will appear to users on the PSTN as originating from one of the assigned DID numbers.

You’re life is easier: Automation

In addition to a visual interface, VoipNow also offers billing engine operations. With a regular IP PBX you most likely would have to export call records and then process them either manually or with another software product. This is not time effective and the reports may not be precise. VoipNow’s charging plans automate this operation for you and its call history based reporting is always exact.

To sum it up...

  1. Be sure to set up an IP network and configure compatible servers suitable for the infrastructure you have planed. Choose the most suitable codecs and connect a physical gateway if your server is in-house and you wish to extend your network to PSTN.
  2. Set up your software. With VoipNow, installation is done from the browser interface and you need not worry about the IP PBX.
  3. Set up the devices: you can either install softphones, IP Phones or regular phones with *VoIP
  4. Expand your network with the help of SIP carriers and get in touch with infrastructures other than your own.
  5. Make yourself available for calling with Direct Inward Calling which can be provided by your SIP carrier.

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