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Freeswitch 1.6 Cookbook

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Freeswitch 1.6 Cookbook

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Freeswitch 1.6 Cookbook
Jul, 2015 | 190 pages
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Table of Contents (14 chapters)
FreeSWITCH 1.6 Cookbook
FreeSWITCH 1.6 Cookbook
Credits
Credits
About the Authors
About the Authors
About the Reviewers
About the Reviewers
www.PacktPub.com
www.PacktPub.com
Preface
Preface
Free Chapter
1
Routing Calls
Routing Calls
Introduction
Internal calls
Incoming DID (also known as DDI) calls
Outgoing calls
Ringing multiple endpoints simultaneously
Ringing multiple endpoints sequentially (simple failover)
Advanced multiple endpoint calling with enterprise originate
Time-of-day routing
Manipulating SIP To: headers on registered endpoints to reflect DID numbers
2
Connecting Telephones and Service Providers
Connecting Telephones and Service Providers
Introduction
Configuring an SIP phone to register with FreeSWITCH
Connecting to Skype
Configuring an SIP gateway
Codec configuration
3
Processing Call Detail Records
Processing Call Detail Records
Introduction
Using CSV CDRs
Using XML CDRs
Inserting CDRs into a backend database
Using a web server to handle XML CDRs
Using the event socket to handle CDRs
Directly inputting CDRs into various databases in real time
4
External Control
External Control
Introduction
Getting familiar with the fs_cli interface
Setting up the event socket library
Establishing an inbound event socket connection
Establishing an outbound event socket connection
Using fs_ivrd to manage outbound connections
Filtering events
Launching a call with an inbound event socket connection
Using the ESL connection object for call control
Using the built-in web interface
5
PBX Functionality
PBX Functionality
Introduction
Creating users
Accessing voicemail
The company directory
Using phrase macros to build sound prompts
Creating XML IVR menus
Music on hold
Creating conferences
Sending faxes
Receiving faxes
Basic text-to-speech with mod_flite
Advanced text-to-speech with mod_tts_commandline
Recording calls
6
WebRTC and Mod_Verto
WebRTC and Mod_Verto
Introduction
Configuring FreeSWITCH for WebRTC
SIP signaling in JavaScript with SIP.js (WebRTC client)
Verto installation and setup
Verto signaling in JavaScript using Verto.js (Verto client)
7
Dialplan Scripting with Lua
Dialplan Scripting with Lua
Introduction
Creating a basic Lua script
User interaction and DTMF gathering
Using conditionals, loops, web calls, and regular expressions
Connecting to an external database
Index
Index

Verto signaling in JavaScript using Verto.js (Verto client)

Let's implement a click-to-call button that will allow anyone to click on a web page to join a conference in video-audio chat (more information about conferences is covered in the next chapter).

For security reasons, we need an "anonymous" user that we can allow into our system without risks, that is, a user that can do only what we have preplanned.

Create an anonymous user for click-to-call by adding to the directory and the dialplan as instructed in the SIP Signaling in JavaScript with SIP.js(WebRTC client) section. Then use reloadxml from fs_cli, or restart FreeSWITCH.

The end user does not need to log in or identify themselves; insert your name is there just for aesthetic purposes.

A minimal click-to-call Verto client looks like this:

How to do it...

In a directory served by your HTPS server (for example, Apache with an SSL certificate), put these two files:

HTML (3.html):

<!DOCTYPE html>
<html>
 <head>
  <meta...
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