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Freeswitch 1.6 Cookbook

By : Anthony Minessale II, Giovanni Maruzzelli

5 (3)

Freeswitch 1.6 Cookbook

5 (3)

By: Anthony Minessale II, Giovanni Maruzzelli

Overview of this book

FreeSWITCH 1.6 Cookbook is written for anyone who wants to learn more about using FreeSWITCH in production. The information is presented in such a way that you can get up and running quickly. The cookbook approach eschews much of the foundational concepts, and instead focuses on discrete examples that illustrate specific features. If you need to implement a particular feature as quickly as possible, then this book is for you.

Preface

Preface

What this book covers

What you need for this book

Who this book is for

Sections

Conventions

Reader feedback

Customer support

Free Chapter

1. Routing Calls

1. 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

2. 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

3. 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

4. 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

5. 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

6. 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

7. 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

Customer Reviews

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SIP signaling in JavaScript with SIP.js (WebRTC client)

Let's carry out the most basic interaction with a web browser audio/video through WebRTC. We'll start using SIP.js, which uses a protocol very familiar to all those who are old hands at VoIP.

A web page will display a click-to-call button, and anyone can click for inquiries. That call will be answered by our company's PBX and routed to our employee extension (1010). Our employee will wait on a browser with the "answer" web page open, and will automatically be connected to the incoming call (if our employee does not answer, the call will go to their voicemail).

Getting ready

To use this example, download version 0.7.0 of the SIP.js JavaScript library from www.sipjs.com.

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 in a file named /usr/local/freeswitch/conf/directory/default/anonymous.xml :

<include>
 ...

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