It's fun and useful to explore pure IP telephony, but most folks want PSTN (Public Switched Telephone Network) integration as well. (PSTN is also known as POTS: Plain Old Telephone Service.) An Asterisk server can interface with "legacy" systems, which means "not IP," or your ordinary old digital and analog telephone systems.
Digium Inc., the inventors and sponsors of Asterisk, make legacy interface hardware. Other popular brands are Sangoma, VoiceTronix, and Cisco. There are dozens of different vendors, as most networking hardware manufacturers have added VoIP devices to their product lines. I'll use Digium's products as examples. These require the Zaptel drivers, which you can download from asterisk.org, and the Linux operating systembecause the Zaptel drivers only work on Linux. Other operating systems can use standalone media gateways. "Media gateway" is a broad term that includes both devices for servers, and devices for individual telephones. You can use a media gateway with your Linux system, rather than installing interface cards in your Asterisk server.
An Asterisk server plugs nicely into an existing analog phone network. Just add an adapter like Digium's TDM2400P. This connects to your existing punch block with a standard 25-pair telco cable and connector. Now you have a powerhouse PBX that can do just about anything, for a fraction of the cost of a traditional PBX. The TDM2400P cards cost from around $600 to $1,700, depending on how they are configured. You may use the TDM2400P to connect your existing analog phones to your VoIP network, which allows you to replace them with IP phones on a timetable that suits you. Or never replace them, whatever fits into your master plan.
For smaller systems use the TDM400P. The telephones plug directly into the card, so you don't need a punch block. These cost from around $150 to $400.
The TDM cards are configurable by selecting the number of FXS or FXO modules that you want. FXS modules are green, and FXO modules are red. FXS means "foreign exchange station." FXO means "foreign exchange office." Without going into tedious detail about signaling and other arcane terminology, just remember that the red FXO modules connect to phone lines, and the green FXS modules connect to telephones.
The TDM400P holds up to four modules that control a single line each, and the TDM2400P holds up to 6 4-line modules. Note that when you order these, you need to select the number and combination of modules that you want.
Digium has a bale of different digital interface cards, the Wildcard line. These support T1, E1, and J1. One gotcha is the PCI voltage; in this here era of two different PCI bus voltages, 3.3 and 5.0, you must be careful to select whichever one is supported by your motherboard. Older systems support 5.0; 3.3 is the newfangled voltage. Some boards support both.
These are real, genuine T1/E1/J1 cards that can carry both voice and data, so they are a good value. T1 is the United States standard, and it carries 24 channels. E1 is the European standard and carries 30 channels. J1 is the Japanese standard, which is similar to T1. For the best quality and performance T1/E1/J1 is the way to go. Each channel is dedicated bandwidth, which improves call quality, plus you don't have to play any complicated traffic-control games like you do with cable and DSL. You may keep your existing DID (direct inward dial) lines if you already have a digital phone system, and add VoIP channels at your own pace. It's the easiest to manage, is usually very reliable, and allows you to mix digital/analog/VoIP however you want to.
Digium's cards now all have an echo cancellation hardware module option. Whatever brand you go with, this is a good thing to spend a few extra dollars for, as it will improve call quality and reduce the load on the server's CPU.
Some Asterisk scenarios
There are a number of different ways you can set up your Asterisk voice network. Suppose you have four POTS lines and broadband Internet. Get a TDM400P with four FXS modules, populate your voice network with IP phones, and you have both PSTN and VoIP.
Remote users such as salespeople or branch offices can connect to your Asterisk server over the Internet, avoid long-distance charges to the mother ship, and get free local calling.
Add Asterisk servers to your branch offices to expand your local calling range. As long as someone on the other end has local PSTN service, Asterisk can route your calls to it toll-free.
Asterisk can make WAN calls appear as local PBX extensions, so your users can call each other directly with their four-number extensions.
Come back next week to learn how to build a small POTS-Asterisk PBX.
Thank you to James Lopeman, CTO supreme, for his endless patience and help. There will be cheesecake.