Book Image

Building Telephony Systems with OpenSIPS Second Edition - Second Edition

By : Flavio E. Goncalves, Bogdan-Andrei Iancu
Book Image

Building Telephony Systems with OpenSIPS Second Edition - Second Edition

By: Flavio E. Goncalves, Bogdan-Andrei Iancu

Overview of this book

OpenSIPS is a multifunctional, multipurpose signalling SIP server. SIP (Session Initiation Protocol) is nowadays the most important VoIP protocol and OpenSIPS is the open source leader in VoIP platforms based on SIP. OpenSIPS is used to set up SIP Proxy servers. The purpose of these servers is to receive, examine, and classify SIP requests. The whole telecommunication industry is changing to an IP environment, and telephony as we know it today will completely change in less than ten years. SIP is the protocol leading this disruptive revolution and it is one of the main protocols on next generation networks. While a VoIP provider is not the only kind of SIP infrastructure created using OpenSIPS, it is certainly one of the most difficult to implement. This book will give you a competitive edge by helping you to create a SIP infrastructure capable of handling tens of thousands of subscribers. Starting with an introduction to SIP and OpenSIPS, you will begin by installing and configuring OpenSIPS. You will be introduced to OpenSIPS Scripting language and OpenSIPS Routing concepts, followed by comprehensive coverage of Subscriber Management. Next, you will learn to install, configure, and customize the OpenSIPS control panel and explore dialplans and routing. You will discover how to manage the dialog module, accounting, NATTraversal, and other new SIP services. The final chapters of the book are dedicated to troubleshooting tools, SIP security, and advanced scenarios including TCP/TLS support, load balancing, asynchronous processing, and more. A fictional VoIP provider is used to explain OpenSIPS and by the end of the book, you will have a simple but complete system to run a VoIP provider.
Table of Contents (21 chapters)
Building Telephony Systems with OpenSIPS Second Edition
Credits
About the Authors
About the Reviewers
www.PacktPub.com
Preface
Index

Lab – testing the call forwarding feature


To create this lab, some experience with Asterisk is required for the voicemail integration. This lab is relatively hard to implement. Some IP phones hardly send the busy message because they have more than a single line. It is important to use all the lines before to get the 486 Busy Here message. I like to do this test with two softphones, Xlite and Zoiper, in the same computer. When you reject a call in Zoiper, it sends a 486 Busy Here message to the SIP server. We will reduce the INVITE timeout in order to make the tests easier and less cumbersome. We can change the SIP timers using the following parameters:

modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 10)

The test for call forward is done on two occasions:

  • Test the call forward for unanswered calls: From the extension 1001, call the extension 1002. Do not answer the call at the user 1002. The system will forward the unanswered call to the user 1000.

  • Test the call forward on...