Page 56 - Asterisk™: The Future of Telephony

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.

Other types of PSTN interfaces
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, Grand-
stream, 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.

Connecting Exclusively to a Packet-Based Telephone Network

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 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.

Echo Cancellation

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-

# 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.
* Technically, no timing source is needed for music on hold, but it generally works better with one.

28 | Chapter 2: Preparing a System for Asterisk