Determining PSTN Needs | Asterisk

Determining Our Needs

Now that we have examined some of the options, we need to determine what our needs are. Requirements will vary quite a bit from site to site. Something to keep in mind is that, although the previous choices are distinct, they can be mixed in an Asterisk installation. We can have VoIP providers and POTS lines, as well as a PRI if we desire. It's very common to have this type of setup. For example, if we have an office in another country, we can call them using VoIP but all local calls could use POTS. It is important to understand the calls our system will be making and where they will be going, so that we can arrange for the necessary services and ensure that the calls are routed accordingly. If we have an existing telephony system, we can take a look at the calls it's making just now and our current costs so that we can determine what technologies will be of most use to our system's users.

Now is the time to begin documenting what our plan is. If Asterisk is to replace an existing system, then we should start by writing down all the current lines coming into our incumbent PBX. Once that is done, we need to look at our requirements.

First, we need to determine how many lines are needed. Telephone providers can generate a usage report that will tell us the maximum concurrent connections we have experienced in the last month. While they are able to do this, many providers are not very happy to run such a report. However, without that information we have nothing to gauge our needs other than gut feelings.

If we need more channels than we have, someone will get a busy/congested signal. Therefore, we should plan to have the maximum number of channels we have used plus a reasonable cushion. 125 percent of our current maximum is usually a reasonable cushion, this allows for instant 20-25 percent growth so that we can accommodate a sudden increase in calls without the system failing over, causing busy signals. If we do increase calling to this level for a relatively long period, we must consider an increase in lines to prevent congestion. These numbers are a guideline and they can change depending on circumstances. In a call center where the main business purpose is to make and receive calls, 150 percent may be a more satisfactory figure. We also should take into account the time it takes to get new lines set up from our local operator. If a significant event that generates a large number of calls occurs, we should have the capacity to handle this or be able to increase the capacity quickly.

Now that we have a number of lines, we need to determine the technology to use for each line. VoIP is usually the cheapest, especially for long-distance calls. PRI is usually the most reliable, and for incoming calls is often cheaper than VoIP.

While pricing the options, we need to remember that POTS lines usually have a single phone number only, while a PRI can have hundreds of phone numbers. If we are a business that receives only a few calls, but needs the calls to have different phone numbers, then a PRI probably makes the most sense. Also, with a PRI we can trunk more effectively, which may become essential.

Although a PRI can have hundreds of phone numbers, there is a charge for each number each month. Called DID (Directed Inward Dialing) numbers, these virtual numbers are usually sold in blocks of 10-20. If we do not order enough to begin with, it is usually not difficult to get new DIDs ordered. Often they can be available the same week, depending on the phone company. We assign these numbers to individual devices or groups of devices ourselves, once we have them allocated.

This means we can decommission or reallocate numbers as necessary. We may have campaign DIDs that are reassigned to different groups depending on the current campaign, personal DIDs for key staff, or our main DID, which would probably be assigned to a group of people responsible for handling these calls.

We should take this opportunity to write down what lines we want, what phone numbers we need, and get quotes if it differs from the currently installed PSTN connections.

The Public Switched Telephony Network (PSTN)

Most of the telephones in the world are connected to a vast network, enabling any telephone to reach any other. This network is called the Public Switched Telephony Network (PSTN). The phones that are on this network are reachable by dialing a number, which may include country codes, area codes, and telephone numbers.

While there are instances in which interconnection with the PSTN is inappropriate, most users of telephones have the expectation that they can reach the world at large. Therefore, we will consider interconnection to the PSTN as a requirement.

Connection Methods

There are a number of different methods to connect to the PSTN. Each has advantages and disadvantages, most of which we will touch on. As pricing varies depending on city or country, exact pricing will not be given. Pricing should be researched based upon the location of the Asterisk server.

We will handle each connection method one at a time.

Plain Old Telephone Service (POTS) Line

Probably the most common connection to the PSTN is a POTS line. This is an analog line provided by a telephone carrier. Each POTS line can carry only one conversation at a time.

For small installations, POTS lines are usually the most cost effective when connecting directly to our Local Exchange Carrier (LEC), a term used to refer to any company providing local telephone service. Eight lines is usually the point at which we should seriously look at another technology for our connection.

POTS lines from our LEC require a Foreign eXchange Office (FXO) interface to be usable in Asterisk. We will focus on Digium's offerings, namely the FXO module on a TDM410. Each TDM410 can use up to four modules. Therefore, if we have one line, we will have three empty module slots on the card.

Integrated Services Digital Network (ISDN)

ISDN is an all-digital network that has been available for over a decade. It is available in two major versions— Basic Rate Interface (BRI) and Primary Rate Interface (PRI).

ISDN divides a line into multiple channels. Each channel can contain either pay load (Bearing, or B channel) or signaling (Data, or D channel). A BRI has three channels—one D channel and two B channels. Therefore, two phone calls can be in progress at a time on a single BRI. A PRI has 24 channels—one D channel and 23 B channels, resulting in up to 23 simultaneous calls.

ISDN is not limited to voice alone. Each channel can carry 64k of data, if so configured with the LEC. This gives ISDN a lot of flexibility over POTS lines, as the channels can be reconfigured between voice and data on the fly.

With its separate D channel, ISDN is able to do things POTS cannot, such as setting custom caller ID, receiving dialed number information, on-the-fly redirection of calls, and a host of other cool features. Of course, all of these features require cooperation from the LEC, which is not always forthcoming.

BRI does not have high penetration in the United States market. Some accuse LECs of vicious pricing, while others claim consumers are to be blamed for fearing new technology. Either way, the result is the same—if we call our LEC and request a BRI, they will assume it is for data.

On the other hand, PRI is widely used in the US. It is the connection of choice for larger installations. PRIs are actually delivered over T1 connections, a proven and usually very reliable technology.

T1 or E1

Technically speaking, when ordering service from an LEC, we order a DS1, which is delivered over a line referred to as T1. However, this detail is usually overlooked. Therefore, we will refer to it in its vernacular—T1.

A T1 is a line with 24 channels. Each channel can contain a call. Therefore, a T1 can contain an additional call when compared with a PRI. In Europe, E1s are more common. In comparison to T1, they have 32 channels instead of 24. T1s signal the call through Robbed Bit Signaling, also referred to as CAS (Channel Associated Signaling) or flat T1. What this means is that a bit is robbed from time to time, as information needs to flow about the connection. While this is usually imperceptible to the human ear, it can be deleterious to data connections.

Using a T1 to deliver both data and voice is common. Some of the 24 channels are designated to be used for data and others are used for voice. There may even be unused channels. LECs are able to offer lower pricing when bundling services in this way, as a few channels may be used for voice, others for an Internet connection, and yet others could be used for a private data connection to another office.

LECs are able to send information about the number that was dialed at the beginning of the call. In this way, one advantage of the PRI has been matched by T1s. If we intend to have about 8 to 12 lines as well as a data connection, a T1 can be a good choice.

An excellent telephony interface card to connect your Asterisk to a T1/E1 connection is the Digium TE122. Today T1 connections can be split to accommodate data and voice. For example, your provider can offer 12 channels of voice as well as a data connection for your computers all on a single T1. The TE122 can support both modes and direct the voice channels to your Asterisk, while separately directing your data connection to the underlying Linux operating system, thus eliminating the need for an external router.

Voice over IP Connections

In recent years, a new way to connect to the PSTN has cropped up. Companies are using PRIs, T1, and other technologies to connect to the PSTN, and then reselling those connections to consumers. The users connect to the companies offering these connections through Voice over IP technologies. By doing so, we can skip dealing with LECs completely.

This service is called origination and termination. Through these services, we can receive a real telephone number with the area code, depending on what the provider has access to. Not all providers can offer numbers in every locality. This means that our number could be long distance from our next-door neighbor, yet local to someone in the next state. However, the advantage of this is that the provider will route most of the calls over their VoIP infrastructure and will then use the PSTN when they get to their most local point at the receiving end. This can mean that long distance charges are dramatically reduced. If we call a variety of countries, states, or cities it can be worthwhile to research a provider that offers local PSTN access to the areas we call the most.

The rates per minute are usually very attractive. Often, long distance is at the same rate as local calls. One thing to watch out for is that some providers charge for incoming minutes much like on a cellular telephone, and some providers also charge for local calls.

Today there are VoIP carriers offering unlimited US packages for those running Asterisk. However, one thing to watch out for with unlimited packages is that the carrier usually restricts the number of simultaneous calls you can make or receive. When you inquire about an unlimited package be sure to ask how many channels you are receiving for origination and termination.

Another thing to be aware of is that some providers require you to use their Analog Terminal Adapter (ATA). This means that they will send you a box that you plug into the Internet, which uses Voice over IP. Then, you have a POTS line to connect a phone (or Asterisk) to. However, today many VSPs (VoIP Service Providers) are offering BYOD (Bring Your Own Device) in which they provide you with the SIP or IAX settings. Once you have these settings you can connect them to your Asterisk deployment.

Voice over IP makes sense in many installations. But for the quality to be acceptable, a reliable Internet connection with low latency is required. Another thing to watch out for is jitter. Jitter refers to the variation in latency from packet to packet. Most protocols can handle latency a lot better if it is constant throughout the call.

A good candidate for Voice over IP is a site where interruptions in service will not endanger life and will not irreparably harm the company. While VoIP providers strive to achieve very high availability, we also have to rely on the Internet at large and our VoIP provider's ISP, as well as our own ISP.

If our telecommunication needs are such that periodic downtime is tolerable, VoIP will probably be our least expensive option. It requires less hardware in our Asterisk system as well, increasing the savings. In order to use VoIP with Asterisk, all we need is a system capable of Internet access. We don't require any specialized telephony hardware.

Return on Investment

The cost of owning a phone system is only one piece of the Return on Investment (ROI) puzzle. ROI attempts to quantify an expenditure's effect on the bottom line, usually used to justify a large capital outlay.

Just as an example, one phone system that I installed went into an existing business. Its existing phone system had an automated attendant that had the unfortunate habit of hanging up on customers if they pressed the 0 key, or if they didn't press any key for 5 seconds.

What was the ROI for moving to a new phone system? Not having angry customers who got hung up is a hard value to calculate. According to one of the owners of the business, that value was infinite. That made the cost of Asterisk very easy to justify!

ROI is basically the TCO subtracted from the quantification of the benefit (in money) to the business. Therefore, if we calculated that a new phone system would save $5000 and cost $4000, the ROI would be $1000.

Another interesting calculation to make, which is also categorized as ROI, is the time for the cost to be recouped. This calculation is the one that I find helpful in making a business case for Asterisk.

Suppose a phone system costs $5000 to install. Using toll bypass, you can save a net $500 per month. In 10 months, the cost of installing the system will be swallowed up in the savings.

These are simple examples, but ROI can help to justify replacing an existing phone system. By having these numbers prepared before proposing to replace the phone system, we can have a more professional appearance and be more likely to succeed in starting our Asterisk project.

Calculating Total Cost of Asterisk Ownership

Asterisk is distributed as free, open source software. The only costs involved with Asterisk are hardware, right? Well, maybe not.

As we have been discussing, Asterisk is very flexible. Determining how to use the flexibility in the best way can quickly become a huge time sink. Compatible handsets are also not free. If we are going to use the G.729 protocol, which compresses VoIP traffic by a factor of eight while maintaining excellent voice quality, we will also have to pay licensing fees.

With commercial phone systems, the costs are typically higher than with Asterisk. However, they are a fixed, known constant. Depending on the way we use Asterisk, costs can vary greatly.

The total cost of owning Asterisk can also include downtime. If we choose to support Asterisk on our own, and have to work to try to get Asterisk back up after a failure, there is an opportunity cost involved in the calls we should have received. This is why we should choose to support our phone system internally only if we have the appropriate resources to back that up.

Total Cost of Ownership (TCO) is not an easy calculation to make. It involves assumptions of how many times it will break, how long it will take us to get it up and running, and how much the consultants will charge us if we hire their services.

TCO is useful only when comparing phone systems to each other. The following elements should be included when comparing TCO of multiple phone systems:

• Procurement cost: This is the cost to buy the PBX. In the case of Asterisk, it is only the cost of the hardware; other systems will include an element of licensing.
• Installation cost: This is the cost to configure and deploy the PBX. Some companies choose to do the deployment in-house. In such instances, there is still a cost, and to enable fair comparisons it should be included.
• Licensing cost (one-time): This is the cost of any one-time licensing fees. Some PBX systems will require a license to perform administration, maintenance, connection to a Primary Rate ISDN line (PRI), and so on. In Asterisk, this would include the G.729 licensing cost, if required.
• Annual support cost: This is the estimated cost of ongoing maintenance. Of course some assumptions will have to be made. In order to keep the comparison fair, the same assumptions should be carried over between vendors.
• Annual licensing cost: Some phone systems will have an annual cost to license the software on the handsets as well as a license to be able to connect those handsets to the PBX.

When we have created the table, we can calculate the TCO for one year, two years, and so on. We can then evaluate our business and decide what costs we're willing to incur for our phone system.

Trade-Offs | Asterisk

Is Asterisk a Good Fit for Me?

Looking at what Asterisk is and is not, the natural question follows—is Asterisk right for me? This is a vitally important question that should be given serious consideration. Let's take a moment and look at some of the considerations we must explore before we commit to using Asterisk.

Trade-Offs

There are a series of trade-offs we must consider with Asterisk. Choosing Asterisk will lock us into certain choices, while others will be available whether we install an Asterisk server or not. We will now examine some of these trade-offs so that we can gauge the impact they have on us.

Flexibility versus Ease of Use

Asterisk is a very powerful framework into which we can install almost anything. We can configure each piece of Asterisk to the minutest detail. This gives us an amazing amount of flexibility.

This flexibility comes with a price. Each of these details must be researched, understood, and tried. Each change we make affects other parts of the phone system, whether for good or bad. Asterisk is not an easy-to-use platform, especially for a beginner.

There is a learning curve, but it is one that can be surmounted. Many developers have become experts in telephony and many telephony experts have mastered server administration. But each of us must decide what we expect from our phone system. I like to think of it in three major categories, as outlined in the following table:

Description	Solution
I want to plug in the telephone system and never think about it again. I want to call someone when things are not working. I do not plan to add anything to the system once it is set up.	A proprietary phone system is probably your best bet. Many offer a pre-configured system, and when changes are made, a certified consultant will be required.
I don't know much about phone systems, but I want to learn. I need a phone system soon. I'd like to have flexibility and additional features, and may change the configuration of my phone system from time to time.	Either use a packaged version of Asterisk or have a consultant build a customized Asterisk server. Learn to use Asterisk. Build a couple of Asterisk servers just to explore. Add features as necessary.
I want to learn and build my own phone system. I am interested in creating a custom solution for my problems. I am willing to accept the responsibility if something doesn't work, and take the time to figure out why.	Build an Asterisk server from the ground up. Much will be learned in the process, and the result will be an extremely powerful business tool.

Of course, these are not distinct categories. We each fall into a continuum. It is important to realize that Asterisk, as great as it is, is not the right solution for everybody. Like any technology we implement, we must consider its impact on the business. We must also decide whether it will become something useful that enables us to work better, or whether it will require too much maintenance and other work to make it an efficient addition. This depends entirely on our purposes and the other technology we have that requires our attention.

Graphical versus Configuration File Management

Asterisk currently uses plain text files to configure most options. This is a very simple way to create, back up, and modify configurations for those who are comfortable with command-line tools.

Some PBX systems offer a GUI to update the configurations. Others don't allow the configuration to be changed except by dialing cryptic code on telephone handsets. Still others cannot be configured at all, except by certified technicians who receive the required software and cables from the phone system manufacturer.

A few good open source tools are being created to ease the management of Asterisk. However, to get the full ability to customize Asterisk, editing of text files is still required. To help get used to this method of configuration, this book focuses on the text files without relying on any GUI package.

Asterisk 1-2-3

Setting up Asterisk and working with configuration files without a database is not intended for a beginner. Originally, Asterisk was not considered an off-the-shelf PBX. However, in recent years all of this has changed.

For those who are looking for an off-the-shelf Asterisk PBX system, Digium created the Asterisk Appliance, a feature-rich PBX solution that's easy to install and manage. The Asterisk Appliance allows users to use traditional analog lines as well as a VoIP service provider.

For those who are just beginners, there is a packaged solution called Trixbox CE (www.trixbox.org). Trixbox CE offers a free single CD installation that installs Linux, Asterisk, a database (MySQL), as well as an easy-to-use web-based interface to create and manage your PBX settings. The installation takes approximately 30-60 minutes and once complete, you have a VoIP server ready to go. However, if you want to connect traditional analog lines to your server, you will need to purchase an FXS/FXO card. Please note that for connecting standard POTS (Plain Old Telephone Service) lines to your Asterisk PBX, you will need to purchase an FXO expansion card called Fonality.

For those of you who are a bit more technically inclined and desire to install each piece of Asterisk individually, you may still want an easy-to-manage interface for your deployment. FreePBX is an easy-to-use GUI (graphical user interface) that controls and manages Asterisk (www.freepbx.org).

Another great resource for those interested in FreePBX is the book called FreePBX 2.5 Powerful Telephony Solutions. You may also visit:

http://www.packtpub.com/freepbx-2-5-powerful-telephony-solutions

Asterisk Scalability

In the past, Asterisk was not a solution for those requiring 100 SIP devices or more. However, in recent years major releases have dramatically increased reliability, scalability, and capacity. Today Asterisk servers can support hundreds of extensions and up to 240 simultaneous calls. For example, Asterisk Business Edition has been tested to handle up to 240 simultaneous calls without any issues. However, it being computerized, the speed, capacity, and reliability is fully dependent on the parts that make up the system. For this reason, ensure you have enough hard drive space, RAM, and CPU power to run your Asterisk server. Those of you who will be using a VoIP service provider for origination (receiving incoming calls) and also termination (outgoing calls) supporting SIP/IAX devices on remote networks, please ensure you have enough bandwidth from your ISP.

Asterisk does not Run on Windows

At one point, Asterisk had a demonstration CD that worked with Windows. However, Asterisk offered direct from Digium does not run on the Microsoft platform. Asterisk requires near real-time access to system resources. It also requires hooks into certain resources. Actually, Asterisk is built to use Linux, the open source *NIX operating system.

AsteriskWin32 (http://www.asteriskwin32.com) is an open source project that has managed to get Asterisk 1.2.26.2 compiled for Windows. However, it is highly recommended that you stick with Linux as you will find more support for it in the Asterisk community.