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

Implementing call forward on busy or unanswered


Now, we will review an important concept called failure_route. We will handle the following failure situations:

  • 408: Request Timeout (phone not registered or disconnected)

  • 480: Temporarily Unavailable (no answer)

  • 486: Busy Here

To implement call forward on busy and call forward when unanswered, we will use the concept of the failure route. In this logic, we will call the t_on_failure("missed_call") function just before sending the INVITE to the destination. This allows the handling of SIP negative replies (codes higher than 299).

We will use a logic similar to blind call forwarding; load an AVP to redirect the call to a specific URI as follows:

if(avp_db_load("$rU","$avp(fwdbusy)")) {
  $ru = $avp(fwdbusy);
  xlog("forwarded to: $avp(fwdbusy)");
}

Here is an excerpt of the route section with the changes highlighted:

####### Routing Logic ########
# main request routing logic

route{

  # apply DB based aliases (uncomment to enable)
  alias_db_lookup...