If you are going to connect Asterisk to any traditional telecommunications equipment, you will need the correct hardware. The hardware you require will be determined by what it is you want to achieve.
Asterisk allows you to seamlessly bridge circuit-switched telecommunications networks[25] with packet-switched data networks.[26] Because of Asterisk’s open architecture (and open source code), it is ultimately possible to connect any standards-compliant interface hardware. The selection of open source telephony interface boards is currently limited, but as interest in Asterisk grows, that will rapidly change.[27] At the moment, one of the most popular and cost-effective ways to connect to the PSTN is to use the interface cards that evolved from the work of the Zapata Telephony Project (http://www.zapatatelephony.org).
Unless you need a lot of channels (or a have lot of money to spend each month on telecommunications facilities), chances are that your PSTN interface will consist of one or more analog circuits, each of which will require a Foreign eXchange Office (FXO) port.
Digium, the company that sponsors Asterisk development, produces analog interface cards for Asterisk. Check out its web site for its extensive line of analog cards, including the venerable TDM400P, the latest TDM800P, and the high-density TDM2400P. As an example, the TDM800P is an eight-port base card that allows for the insertion of up to two daughter cards, which each deliver either four FXO or four FXS ports.[28] The TDM800P can be purchased with these modules preinstalled, and a hardware echo-canceller can be added as well. Check out Digium’s web site (http://www.digium.com) for more information about these cards.
Other companies that produce Asterisk-compatible analog cards include:
Rhino (http://www.channelbanks.com)
Sangoma (http://www.sangoma.com)
Voicetronix (http://www.voicetronix.com)
Pika Technologies (http://www.pikatechnologies.com)
These are all well-established companies that produce excellent products.
If you require more than 10 circuits, or require digital connectivity, chances are you’re going to be in the market for a T1 or E1 card.[29] Bear in mind, though, that the monthly charges for a digital PSTN circuit vary widely. In some places, as few as five circuits can justify a digital circuit; in others, the technology may never be cost-justifiable. The more competition there is in your area, the better chance you have of finding a good deal. Be sure to shop around.
The Zapata Telephony Project originally produced a T1 card, the Tormenta, that is the ancestor of most Asterisk-compatible T1 cards. The original Tormenta cards are now considered obsolete, but they do still work with Asterisk.
Digium makes several different digital circuit interface cards. The features on the cards are the same; the primary differences are whether they provide T1 or E1 interfaces, and how many spans each card provides. Digium has been producing Zaptel cards for Linux longer than anyone else, as they were deeply involved with the development of Zaptel on Linux, and have been the driving force behind Zaptel development over the years.
Sangoma, which has been producing open source WAN cards for many years, added Asterisk support for its T1/E1 cards a few years ago.[30] Rhino has had T1 hardware for Asterisk for a while now, and there are many other companies that offer digital interface cards for Asterisk as well.
A channel bank is loosely defined as a device that allows a digital circuit to be de-multiplexed into several analog circuits (and vice versa). More specifically, a channel bank lets you connect analog telephones and lines into a system across a T1 line. Figure 2.2, “One way you might connect a channel bank” shows how a channel bank fits into a typical office phone system.
Although they can be expensive to purchase, many people feel very strongly that the only proper way to integrate analog circuits and devices into Asterisk is through a channel bank. Whether that is true or not depends on a lot of factors, but if you have the budget, they can be very useful.[31] You can often pick up used channel banks on eBay. Look for units from Adtran and Carrier Access Corp. (Rhino makes great channel banks, and they are very competitively priced, but they may be hard to find used on eBay.) Don’t forget that you will need a T1 card in order to connect a channel bank to Asterisk.
Many VoIP gateways exist that can be configured to provide access to PSTN circuits. Generally speaking, these will be of most use in a smaller system (one or two lines). They can also be very complicated to configure, as grasping the interaction between the various networks and devices requires a solid understanding of both telephony and VoIP fundamentals. For that reason, we will not discuss these devices in detail in this book. They are worth looking into, however; popular units are made by Sipura, Grandstream, Digium, and many other companies.
Another way to connect to the PSTN is through the use of Basic Rate Interface (BRI) ISDN circuits. BRI is a digital telecom standard that specifies a two-channel circuit that can carry up to 144 Kbps of traffic. It is very rarely used in North America, but in Europe it is very widely deployed. Due to the variety of different ways this technology has been implemented, and a lack of testing equipment, we will not be discussing BRI in very much detail in this book. Please note, however, that BRI is very popular in Europe, and Digium has produced the B410P card to address this need.
If you do not need to connect to the PSTN, Asterisk requires no hardware other than a server with a Network Interface Card (NIC).
However, if you are going to be providing music on hold[32] or conferencing and you have no physical timing source, you will need the ztdummy Linux kernel module. ztdummy is a clocking mechanism designed to provide a timing source to a system where no hardware timing source exists. Think of it as a kind of metronome to allow the system to mix multiple audio streams in a properly synchronized manner.
One of the issues that can arise if you use analog interfaces on a VoIP system is echo. Echo is simply what you say being reflected back to you a short time later. The echo is caused by the far end, but you are the one that hears it. It is a little known fact that echo would be a massive problem in the PSTN were it not for the fact that the carriers employ complex (and expensive) strategies to eliminate it. We will talk about echo a bit more later on, but with respect to hardware we would suggest that you consider adding echo-cancellation hardware to any card you purchase for use as a PSTN interface. While Asterisk can do some work with echo in software, it does not provide nearly enough power to deal with the problem. Also, echo cancellation in software imposes a load on the processor; hardware echo cancellers built into the PSTN card take this burden away from the CPU.
Hardware echo cancellation can add several hundred dollars to your equipment cost, but if you are serious about having a quality system, invest the extra money now instead of suffering later. Echo problems are not pleasant at all, and your users will hate the system if they experience it.
As of this writing, several software echo cancellers have become available. We have not had a chance to evaluate any of them, but we know that they employ the same algorythems the hardware echo cancellers do. If you have a recently purchased Digium analog card, you can call Digium sales for a keycode to allow its latest software echo canceller to work with your system.[33] There are other software options available for other types of cards, but you will have to look into whether you have to purchase a license to use them.[34] Keep in mind that there is a performance cost to using software echo cancellers. They will place a measureable load on the CPU that needs to be taken into account when you design a system using these technologies.
[25] Often referred to as TDM networks, due to the Time Division Multiplexing used to carry traffic through the PSTN.
[26] Popularly called VoIP networks, although Voice over IP is not the only method of transmitting voice over packet networks (Voice over Frame Relay was very popular in the late 1990s).
[27] The evolution of inexpensive, commodity-based telephony hardware is only slightly behind the telephony software revolution. New companies spring up on a weekly basis, each one bringing new and inexpensive standards-based devices into the market.
[28] FXS and FXO refer to the opposing ends of an analog circuit. Which one you need will be determined by what you want to connect to. Chapter 7, Understanding Telephony discusses these in more detail.
[29] T1 and E1 are digital telephony circuits. We’ll discuss them further in Chapter 7, Understanding Telephony.
[30] It should be noted that a Sangoma Frame Relay card played a role in the original development of Asterisk (see http://linuxdevices.com/articles/AT8678310302.html); Sangoma has a long history of supporting open source WAN interfaces with Linux.
[31] We use channel banks to simulate a central office. One 24-port channel bank off an Asterisk system can provide up to 24 analog lines—perfect for a classroom or lab.
[32] Technically, no timing source is needed for music on hold, but it generally works better with one.
[33] This software is not part of a normal Asterisk download because Digium has to pay to license it separately. Nevertheless, it has grandfathered it into all of its cards, so it is available for free to anyone who has a Digium analog card that is still under warranty. If you are running a non-Digium analog card, you can purchase a keycode for this software echo canceller from Digium’s web site.
[34] Sangoma also offers free software echo cancellation on their analog cards (up to six channels).