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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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Replacing Call Signaling with VoIP

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CHAPTER

Replacing Call Signaling with VoIP

In Chapter 6, the process of transmitting voice sounds in virtual media channels, via codecs, was presented. In order for those media channels to be set up, monitored, and destroyed when needed, a PBX calls on signaling. Different methods are used for different kinds of endpoints and trunks. On the PSTN, the SS7 network handles signaling. On a POTS voice channel, the signaling is accomplished using analog FXS signaling.

SS7, FXS, and the dozens of other signaling technologies in use on the PSTN, though all signaling protocols, are outside the realm of VoIP. They could all be considered legacy technology, since just about all of their signaling functions have been replicated using several new, modern, open TCP/IP-centric standards. Even though SS7 is a packet-based protocol and there are attempts underway to make it compatible with

VoIP softPBX systems (Asterisk included), its roots are in the PSTN, not the Internet.

This chapter describes the standards for call signaling in a softPBX-based VoIP network; it also describes the ways these standards compete with and complement one another.

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Enterprise Telephony Applications

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CHAPTER

Enterprise Telephony Applications

In the previous chapter, the technologies of legacy voice network systems were discussed. Some might find that subject fascinating enough to have spent more than a chapter on it. In fact, there are volumes on the subject, and the ITU web site (http:// www.itu.int) is filled with papers that describe it all in painfully unsparing verbosity.

But it’s telephony, the application functionality within the voice network, that is the fun part. Telephony accommodates and assists human interaction in a very real, personal way, which is why it’s such an engaging subject. Unlike written forms of communication, such as email or instant messaging, telephony’s distinguishing traits are its use of sound and its immediate, real-time nature. It’s a much more fundamental mode of interaction than the written form—because when we use telephony, we talk, the same thing we do when we’re together.

Telephony can use live, immediate speech or speech that’s recorded, stored, and played back later, depending upon the needs of the application—and it can be largely automated using well-defined standards. In fact, computer-integrated telephony applications have even been programmed to recognize and respond to human voice commands.

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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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What Can Go Wrong?

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What Can Go Wrong?

As system designers, integrators, and geeks, we do our homework to become masters in a subject fully before we go headfirst, implementing it in a production environment, right? Well, we aspire to, anyway. But experience says you can never know a hundred percent of what there is to know about a subject… particularly with Voice over IP, which is still changing and evolving. So, once you’ve equipped yourself with the book knowledge, you hit the field. It’s only then that you’ll get a chance to discover whether your VoIP project plan has any remaining flaws. But, before you take that leap of faith, there are a few things you should know about: What can go wrong when implementing VoIP in the enterprise?

Common Problem Situations

The people you call complain about echo

Generally, echo is at its worst when end-to-end latency is high. If end-to-end latency is below 150 ms, echo should be nearly imperceptible. Remove echo by removing latency. Remember that using bigger packet sizes, which are often used with lowbandwidth codecs, can increase latency. If capacity is stopping you from removing latency, increase the capacity on the links that are causing the latency. Steer away from frame-relay and VPN if the link is critical—these technologies provide the slowest links.

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