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Chapter 16: Introduction to Interactive Voice Response |
near-miraculous feats of processing; something your IVR might not be able to fully |
harness. |
Text-to-Speech |
Text-to-speech (also known as speech synthesis) requires that a system be able to arti‐ |
ficially construct speech from stored data. While it would be nice if we could simply |
assign a sound to a letter and have the computer produce each sound as it reads the |
letters, written language is often not phonetic, and seldom reflects the nuances of |
speech (English is arguably one of the worst languages in this regard). |
There are now excellent APIs available from Google (and others), that will do a very |
good job of reading back what has been written. As of this writing, it’s still very obvi‐ |
ous that it’s a computer speaking, but it is nevertheless possible to generate system |
prompts on the fly from text, rather than having to record all prompts in advance. |
The usefulness of this is difficult to evaluate, since humans are still not interested in |
talking to your machines; they phoned because they want to talk to you. |
Speech Recognition |
Since we’ve managed to convince our computers to talk to us, we naturally want to be |
able to talk to them as well.3 |
Speech recognition was previously complex and expensive, but Google has recently |
released an API that allows the enormous power of their speech recognition capabili‐ |
ties to be available to external applications. |
Conclusion |
Asterisk is an excellent IVR platform. This entire book, in many ways, is teaching you |
skills that can be applied to IVR development. While the mainstream media only |
really pays attention to Asterisk as a “free PBX,” the reality is that Asterisk is at its |
most potent when used as an IVR. Within any respectable-sized organization, it is |
very likely that the Linux system administrators are using Asterisk to solve telecom |
problems that previously were either unsolvable or impossibly expensive to solve. |
This is a stealthy revolution, but no less significant for its relative obscurity. |
If you are in the IVR business, you need to get to know Asterisk. |
3 Actually, most of us talk to our computers, but this is seldom polite. |
Conclusion |
| |
297 |
CHAPTER 17 |
Asterisk Manager Interface and Call Files |
John Malkovich: I have seen a world that NO man should see! |
Craig Schwartz: Really? Because for most people it’s a rather enjoyable experience. |
—Being John Malkovich |
The Asterisk Manager Interface (AMI) is a system monitoring and management |
interface provided by Asterisk. It allows live monitoring of events that occur in the |
system, as well as enabling requests for Asterisk to perform some action. The avail‐ |
able actions are wide-ranging and include things such as returning status information |
or originating new calls. Many interesting applications have been developed on top of |
Asterisk that use the AMI as their primary interface to Asterisk. |
This chapter also includes documentation on the use of call files. Asterisk’s call files |
are an easy way to originate a few calls. Once call origination volume increases or |
your needs become otherwise more complex, you can move on to using the AMI. In |
fact, we find call files to be useful enough that we’re going to talk about them first. |
Call Files |
It is common to use AMI for originating calls, but in many situations it’s easier to use |
call files. A call file is a simple text file that describes the call that you would like |
Asterisk to originate. When a call file is placed into the /var/spool/asterisk/outgoing |
directory, Asterisk will immediately detect that a file has been placed there and pro‐ |
cess the call. |
Asterisk comes with a sample call file, which you will find at ~/src/asterisk-15.<TAB>/ |
sample.call (or wherever the root directory of your Asterisk source is located). |
299 |
Your First Call File |
For your first call file, let’s create a call between two of your telephones. Make sure |
you have at least two of your phones registered and working. For this example we’ll |
be using SOFTPHONE_A and SOFTPHONE_B. |
Create the following file in your home directory: |
$ vim ~/call-file |
Channel: PJSIP/SOFTPHONE_A |
Extension: 103 |
Context: sets |
Make a copy of this file (so that you don’t have to re-create it every time you want to |
run it): |
$ cp ~/call-file docall |
Change the ownership of the docall file to asterisk: |
$ chown asterisk:asterisk docall |
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