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AGI Commands

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Appendix B

APPENDIX B

AGI Commands

ANSWER

Answers the current channel if not already answered.

Returns -1 upon failure, or 0 upon success.

CHANNEL STATUS [channel]

Returns the status of the specified channel. If none is given, returns the status of the current channel. Here is what the status codes mean:

0: Channel is on hook and available.

1: Channel is on hook, but reserved.

2: Channel is off the hook but no digits have been dialed.

3: Digits have been dialed.

4: The line connected to this channel is ringing.

5: A called line connected to this channel is ringing.

6: A called line connected to this channel has a call in progress.

7: A called line connected to this channel is busy.

DATABASE DEL family key

Deletes a value in the Asterisk database for the specified family and key.

Returns 1 if successful, 0 if not.

DATABASE DELTREE family [keytree]

Deletes a family or and/or keytree within a family in the Asterisk database.

Returns 1 if successful, 0 if not.

DATABASE GET family key

Retrieves a value in the Asterisk database for the specified family and key.

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Asterisk Reference

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Chapter

17 17

CHAPTER

Asterisk Reference

Earlier in the book, we covered compiling and installing Asterisk, installing a legacy interface card and drivers, and setting up some simple PBX applications. Later on, we used Asterisk configurations to illustrate some common enterprise telephony concepts. This chapter is geared toward the person who’s comfortable with the earlier material in the book and wants a deeper understanding of Asterisk.

Asterisk is a deep subject that touches disciplines of networking, code-writing, protocols, and standards. This chapter won’t make you an Asterisk expert, but it should help you go a step or two beyond the essentials we’ve already covered. We’ll cover channel configuration—with an emphasis on Zaptel and SIP channels, dial-plan syntax elements like variables and string processing, and the commands you can use in building an Asterisk dial-plan.

How Asterisk Is Supported

Asterisk’s principal sponsor is a firm called Digium, based in the United States. The company provides development leadership and commercial support for the open source system. It also manufactures and distributes the Wildcard interface devices and iAXY ATA used in some of the projects in this book. A number of independent consultants provide commercial support for Asterisk. Subscribing to the Asterisk

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Voice and Data: Two Separate Worlds?

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Chapter 1

CHAPTER 1

Voice and Data:

Two Separate Worlds?

Telephony is the communication of spoken information between two or more participants, by means of signals carried over electric wires or radio waves. Ever since Alexander Graham Bell invented the telephone circuit and first envisioned the public telephone system, consumers and businesses have relied on telephony as a staple of human interaction.

With the advent of Internet technologies and high-speed data connectivity in the enterprise, a new family of telephony technologies began taking hold. Voice over IP, or VoIP, has significant appeal for the enterprise, for service providers, and for end users, because it allows the Internet and commonplace data networks, like those at offices, factories, and campuses, to become carriers for voice calls, video conferencing, and other real-time media applications. VoIP-savvy organizations are discovering that they can apply the paradigm of distributed, software-based networking to voice applications and enable a new generation of telecommunications features, costsavings, and productivity enhancements.

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Quality of Service

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Chapter 9

CHAPTER 9

Quality of Service

Quality of Service is a subject of crucial importance to your success with VoIP. Not surprisingly, QoS technologies aren’t often well-understood by traditional telephony people. But those with a data background have never had to use them, either. So this chapter will introduce you to QoS concepts and protocols, the problems they solve, and the complexities they introduce. Don’t let that scare you, though. There are QoS approaches for networks that have a few dozen endpoints, and there are approaches for giant, high-capcacity networks, too.

QoS Past and Present

In traditional telephony, quality of service for each and every phone call is guaranteed by the constant availability of dedicated bandwidth. Whenever a channel or

“loop” is established across the network, the bandwidth allocated to that channel is steadfast and unchanging. Most digitally encoded call paths on the PSTN use the same codec, G.711, so transcoding isn’t necessary. Almost no processing bottlenecks will be found on the PSTN, and since the system isn’t generally packet-based, there is almost never degradation in perceived call quality as a result of congestion.

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Linux as a PBX

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CHAPTER

Linux as a PBX

Evaluating VoIP for enterprise or for your home phone setup means a lot of experimentation, and you’ll need to build a test server with which to hone your VoIP skills.

That test server should be something you can get a lot out of without spending a bundle or committing to a specific vendor’s commercial VoIP platform before you’ve done your homework. Free telephony software lets you do that homework.

Free Telephony Software

If you were learning engine repair instead of VoIP, you probably wouldn’t use a Ferrari for your experiments. You would want something more forgiving and easier to work on, like a nice Dodge Omni. Luckily, there’s Asterisk PBX software—the very open, roomy-under-the-hood telephony server. Like a Dodge Omni, Asterisk is easy to work on, support is a snap to find, and experimenting is cheap. In fact, Asterisk is free

(although its development is supported by Digium, Inc., http.//www.digium.com). So is its source code.

But like a Ferrari, Asterisk is very powerful. Asterisk supports several Voice over IP communication protocols: H.323, SIP, IAX, and others (see Chapter 7 for more on these). Using these protocols, it can support just about any IP telephone, as well as traditional analog and digital telephones. Asterisk has some industrial-strength features like call-queuing, conference calling, voice mail, and caller ID.

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