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Next, we’ll tackle a few Voice over Internet Protocol (VoIP) interfaces: a local SIP and
               IAX2 channel connected to a softphone or hardphone, along with connecting two As-
               terisk boxes via these two popular protocols.
               For SIP, we are going to cover Linksys, Polycom, Aastra, Grandstream, and Cisco sets.
               If we do not cover your phone model, we apologize, but what is important to realize is
               that while most of these devices have many different parameters that you can define,
               generally only a few parameters need to be defined in order to get the device to work.
               That will be our goal, because we figure it’s a lot less frustrating to tweak a functioning
               device than to get it perfectly set up on the first try. We won’t discuss all the features
               you may want your channel to have (such as caller ID or advanced codec and security
               settings), but you will be able to make and receive calls with your phone, which should
               put a smile on your face—a good state to be in as we dig deeper into things.
               Once you’ve worked through this chapter, you will have a basic system consisting of
               many useful interfaces, which will provide the foundation we need to explore the
               extensions.conf file (discussed in detail in Chapter 5), where the dialplan is stored
               (technically, it contains the instructions Asterisk needs to build the dialplan). If you do
               not have access to the analog hardware, some of the examples will not be available to
               you, but you will still have configured a system suitable for a pure-VoIP environment.


               What Do I Really Need?

               The asterisk character (*) is used as a wildcard in many different applications. It is a
               good name for this PBX for many reasons, one of which is the enormous number of
               interface types to which Asterisk can connect. These include:
                 • Analog interfaces, such as your telephone line and analog telephones
                 • Digital circuits, such as T1 and E1 lines
                 • VoIP protocols such as SIP and IAX ‡
               Asterisk doesn’t need any specialized hardware—not even a sound card—even though
               it is common to expect a telephone system to physically connect to a voice network.
               There are many types of channel cards that allow you to connect your Asterisk to things
               like analog phones or PSTN circuits, but they are not essential to the functioning of
               Asterisk. On the user (or station) side of the system, you can choose from all kinds of
               softphones that are available for Windows, Linux, and other operating systems—or
               use almost any physical IP phone. That handles the telephone side of the system. On
               the carrier side, if you don’t connect directly to a circuit from your central office, you
               can still route your calls over the Internet using a VoIP service provider.







               ‡ …and H.323 and SCCP and MGCP and UNISTIM

               70 | Chapter 4: Initial Configuration of Asterisk
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