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

Playing announcements


To send calls to a media server is not rocket science. All you have to do is rewrite the host and port of the R-URI to the IP address of the media server.

Playing demo-thanks

The demo-thanks.gsm file is a recording present in each Asterisk server that is installed. Let's create a number such as *100 to send a call to the Asterisk server. From now on, each number prefixed by the character, *, will be sent to the media server. In the route section of the file, insert the following just after the alias processing:

if($rU=~"^\*") {
  rewritehostport("ip_address_of_the_media_server");
  route(1);
}

It is very simple; any number starting with * will be sent to the media server. Now, in the Asterisk server, we have to receive the call. To do this, we will edit the sip.conf and extensions.conf files in the /etc/asterisk directory. Usually, for the purposes of testing and education, we install both Asterisk and OpenSIPS in the same server, OpenSIPS in the port 5060, and Asterisk...