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

Disconnecting calls


Once you have an established dialog, you can disconnect the call at any time by sending a BYE to both sides. There are many ways to disconnect a call using OpenSIPS. Some are initiated by the subscriber and others by the system itself. Fortunately, now there is a PV called $DLG_end_reason that is capable to display the disconnection reason. It returns the reason for the dialog termination. It can be one of the following:

  • Upstream BYE: BYE sent by the callee

  • Downstream BYE: BYE sent by the caller

  • Lifetime Timeout: Lifetime expired

  • MI Termination: Ended via the MI interface

  • Ping Timeout: Ended by no reply or 481 to in-dialog pings

  • RTPProxy Timeout: Timeout signaled by RTPProxy

Disconnecting a call using the MI interface

To disconnect a call using the MI interface, you will need to identify the hash number. To identify the hash number of a dialog, you can list the dialogs by call-id using the following command:

opensipsctl fifo dlg_list call_id

The output of this command...