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For Dummies is a registered trademark of Wiley Publishing, Inc. in the United States and other countries. Used here by license.The sophistication of telecom hardware available to the average company has increased tremendously. It was amazing in the 1980s when you could use an automated dialer to direct calls to your long-distance carrier instead of your local carrier. In the 1990s, the technology and market had evolved to where you could identify calls based on where you were dialing and match them to the carriers you may have had at your disposal for the best rate. In 2000, all the buzz was to use Internet circuits to transmit calls. At the time, the quality was a bit suspect, but most of those issues have been worked out, and people now send Voice over Internet Protocol (VoIP) calls around the world every day.
In this article we cover the hardware necessary to make and receive calls, and then we give you the parameters for the server that you need to run your Asterisk for optimum performance.
Introducing the Supporting HardwareAsterisk is wonderful, but it is only software diligently working inside a server. It is fully capable of converting a call from VoIP to TDM, but you still need some kind of hardware interface to connect your server to the phones in your office or the outside world. Even when you install the required hardware to make these connections, you still may need additional software drivers to bring them to life. Table 1-1 lists the hardware and software you need to make connections with Asterisk.
Table 1-1 Asterisk Interface Hardware and Drivers
Application Hardware Available Software Driver Required
Analog connection Digium FXS and Zapata
FXO cards
Digital T-1/E-1 Digium T1 card for 1, Zapata
connection 2, or 4 ports
Digital T-1/E-1 Zapata single Zapata
connection T1 card
Digital T-1/E-1 Sangoma single Zapata
connection T1 card
VoIP connection Network interface ztdummy*
card (NIC)
All above No additional libpri**
applications hardware
Determining your analog hardware needsServers don’t have an unlimited supply of expansion slots available on them. Before you download your Asterisk software, you need to develop at least a two-year plan; we recommend a five-year plan. Analog cards take up the most room and are truly limiting. A four-port analog card can only handle four connections. This is in contrast to a T-1 Internet port that could previously handle 20 consecutive VoIP calls and can now manage 50 or more because of compression. If you will need more than four analog ports in the next year, make sure that you have a second four-port card available. If you need more than eight analog lines and are running out of expansion slots, we recommend purchasing the 24-port card now and leaving the ports open until you need them.
If you aren’t versed in telephony jargon, we recommend using the Foreign Exchange Station (FXS) and Foreign Exchange Office (FXO) cards for service using regular phone lines, just as you do in your home. These lines come from your carrier and terminate in a small jack on the wall that are plugged into one phone. FXS interfaces connect directly to a handheld telephone or a dialer. The FXS cards provide the dial tone, caller ID, and ring voltage to the phone so that you understand that your call is being processed.
The FXO cards connect to the phone lines from your local carrier and transmit your call to the carrier for processing. These cards detect dial tone and ringing from the far end so that your FXS card can then forward this information to you.
The Digium analog cards allow you to either use four individual phone lines through standard RJ-11 jacks (the same jacks your home phone connects to) or an interface where you can use up to 24 phone lines through a specialized plug called an amphenol connector that separates every channel into 24 pairs of wires.
Analog cards from Digium are modular and can support FXO and FXS ports in any configuration. Your four-port card can have one, two, or three FXO cards with one, two, or three FXS ports. It’s great to have options!
If you want 24 phone lines, you need special cabling with the amphenol connector and at least one more piece of hardware. You need a break-out box or punch-down rack to allow you to wire into the individual channels (in the case of the punch-down rack) or to plug your phones into one of the 24 standard RJ-11 jacks (in a break-out box).
You can purchase analog and digital cards that handle 1, 2, 4, and 24 lines as well as digital T1/E1/J1 cards for 1, 2, or 4 ports from the following companies:
Voipsupply: www.voipsupply.com
Sangoma: www.sangoma.com
Digium: www.digium.com
Using external analog or digital cards with Asterisk allows you to connect to your existing phone lines. The larger T1/E1/J1 interface cards with two and four ports are only cost-effective in specials setups where you have an internal need for these phone lines or if you are reselling the lines to your customers.
We suggest that you expand your service with VoIP. Everyone in your company can access phone lines, without requiring you to purchase a single phone line for each employee. One dedicated Internet circuit can easily handle the voice requirements for 10 or 20 people with the correct configuration. Turn to the section “Sending calls out VoIP” later in this article if you decide to take your phone service to the next level.
Going digital and dedicated
Digium, Sangoma, and Zapata manufacture digital cards for Asterisk. Digium has the widest selection of digital cards that offer a two- and four-port model. Sangoma has a similar offering; its four-port cards are currently less expensive than the four-port Digium cards.
Sangoma offers a clear channel DS-3 card (capable of handling IP bandwidth equal to 28 T-1 lines) at this time. The company is also planning to release a channelized version of the card that can handle 672 voice channels (28 T-1s of 24 channels each).
The Zapata drivers are necessary and work well with any of the cards you choose. You have to love software that works with everything.
T-1 is an industry term for a circuit with 1.544 Mbps of bandwidth that is broken into 24 individual channels that can process one call each. This is the standard building block of dedicated digital telephony in the United States and Canada. Europe and much of the rest of the world use circuits called E-1s that are broken into 32 individual channels, giving you the capacity to handle eight more calls than the U.S. T-1 circuits.
All the cards are solid performers, but we have always experienced very good performance with the Digium cards. We also try to support our local Asterisk vendors, and because Digium resides in Huntsville, AL, where one of the authors runs his company, it seems the neighborly thing to do. If we take out the geographic bias, we recommend either the Digium or Sangoma T1/E1 cards.
Sending calls out VoIP
Asterisk comes standard with all the drivers you need to make VoIP calls. As long as you have a dedicated Internet connection and a NIC (network interface card) in your computer, you are ready to go. The only remaining piece of the puzzle you need is the ztdummy driver to help maintain the clocking on the calls, and that is only if you are transferring the calls internally within the Asterisk (or across to another Asterisk server) or if you are setting up conference calls. If you are dialing from a VoIP phone to a VoIP provider, you don’t need the ztdummy driver.
Using VoIP to its fullest requires some research on your part. If you want to send and receive VoIP calls, you may need to have two VoIP carriers. Some carriers only provide inbound service (where people call you). This restriction allows the VoIP carrier to dodge the current federal requirement of providing 911 service. Other VoIP carriers specialize in outbound service (where you dial out) but may have limitations on some of the services provided. You may incur an additional charge for 411 and 911 services, as well as for a directory listing in the white pages. Even if you do get your name in the white pages, you may have to jump another hurdle trying to get your name in the 411 directories. Research these features with your carriers to be sure that you choose the correct carrier.
Communicating with your phones or dialers
So far, we have been speaking about hardware required to connect your Asterisk to a telecom carrier. The connection may be to a single analog telephone line or as many as four individual dedicated digital T-1 or E-1 lines. This is a vital link in sending calls to, or receiving calls from, the outside world, but it isn’t everything you need. Unless you have a recorded message to a list of phone numbers in the same server that is running your Asterisk software, you need some type of interface to individual phones or another server that has a dialing program built on it (if you are a telemarketer).
The telephones you connect to your Asterisk server must be compatible with the type of telephony you are using. You can’t use an Integrated Services Digital Network (ISDN) phone on a connection sending VoIP, just as you can’t use a TDM phone on a digital circuit.
Several manufacturers produce VoIP phones that range in price from $50 to $600 per phone. Your specific application and budget dictate which phone is the best for you. Check out the following Web sites for VoIP phone manufacturers:
VoIP Supply: www.voipsupply.com
Cisco: www.cisco.com
Linksys: www.linksys.com
Polycom: www.polycom.com
D-link: www.dlink.com
Grandstream: www.grandstream.com
Sipura: www.sipura.com
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For Dummies is a registered trademark of Wiley Publishing, Inc. in the United States and other countries. Used here by license.