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  • Book Overview & Buying Building Telephony Systems with OpenSIPS Second Edition
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Building Telephony Systems with OpenSIPS Second Edition

Building Telephony Systems with OpenSIPS Second Edition - Second Edition

By : Flavio E. Goncalves, Flavio E Goncalves, Bogdan-Andrei Iancu
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Building Telephony Systems with OpenSIPS Second Edition

Building Telephony Systems with OpenSIPS Second Edition

4.1 (8)
By: Flavio E. Goncalves, 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 (16 chapters)
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15
Index

Chapter 7. Dialplan and Routing

In the previous chapter, you learned how to use OpenSIPS to route calls between SIP users, provision such users, and authenticate them using a database backend. However, you still cannot send calls to ordinary phones because you are not connected to the Public Switched Telephone Network (PSTN). The challenge now is to route calls from and to the PSTN, as shown in the following figure:

Dialplan and Routing

This PSTN interconnection comes with special needs such as advanced number matching and manipulation (dialplans), IP-based authentication, and dynamic routing (based on the prefix, priority, location, or time of day).

Unless you have a SIP trunk to interconnect calls with the PSTN, you will need a device called SIP PSTN gateway. There are several manufacturers for these devices in the market, such as CiscoTM, AudioCodesTM, and others. You can also use open source software such as FreeSWITCH or Asterisk private branch exchange (PBX) for this task. SIP trunks are becoming...

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