Book Image

Asterisk 1.6

Book Image

Asterisk 1.6

Overview of this book

Asterisk is a powerful and flexible open source framework for building feature-rich telephony systems. As a Private Branch Exchange (PBX) which connects one or more telephones, and usually connects to one or more telephone lines, Asterisk offers very advanced features, including extension-to-extension calls, queues, ring groups, line trunking, call distribution, call detail rerecords, and call recording. This book will show you how to build a telephony system for your home or business using this open source application. 'Asterisk 1.6' takes you step-by-step through the process of installing and configuring Asterisk. It covers everything from establishing your deployment plan to creating a fully functional PBX solution. Through this book you will learn how to connect employees from all over the world as well as streamline your callers through Auto Attendants (IVR) and Ring Groups.This book is all you need to understand and use Asterisk to build the telephony system that meets your need. You will learn how to use the many features that Asterisk provides you with. It presents example configurations for using Asterisk in three different scenarios: for small and home offices, small businesses, and Hosted PBX. Over the course of ten chapters, this book introduces you to topics as diverse as Public Switched Telephony Network (PSTN), Voice over IP Connections (SIP / IAX), DAHDI, libpri, through to advanced call distribution, automated attendants, FreePBX, and asterCRM. With an engaging style and excellent way of presenting information, this book makes a complicated subject very easy to understand.
Table of Contents (15 chapters)
Asterisk 1.6
Credits
About the Authors
About the Reviewer
Preface

SIP interfaces


Session Initiation Protocol, or SIP, is a standardized Voice over IP (VoIP) protocol. This protocol relies heavily on the RTP, which uses UDP ports in the TCP/IP stack. It presents addresses in much the same format as email, as user@domain.

We configure this protocol by editing /etc/asterisk/sip.conf. This file has a number of settings in a [general] section, followed our definitions of users.

There is a whole host of options that we can set. These options include:

  • context: Sets the default context for calls coming into the server. These calls can be from our users, or if we are connected to the Internet, they can be from anywhere. Just to be on the safe side, we should not set this to be a context that can call long distance. The default is "default".

  • realm: Sets the realm of the server. As we discussed earlier, the calls are addressed like email, in that the format is user@domain. This variable is how we set the domain part. This could be your host name or a domain...