The appearance of VoIP technology has now made feasible the use of data network for data, as well as voice communication. In this book, the protocol conversion between two different networks, packet scheduling and routing schemes for gateway selection has been introduced and also verified with Asterisk server. The recent trend for convergence of different access network technologies, market deregulation and emergence of dual-mode end-user devices allow the same destination to be reached by alternative paths and interfaces.Providing a good QoS in heterogeneous networks for irregular traffic flows remains a significant challenge. One of the difficult tasks is that of gateway location, also known as gateway selection, path selection, gateway discovery, and gateway routing. The main objective of this book is to design and develop a PSTN-IP Telephony Gateway (PITG) model and to assist routing packets while ensuring minimal call blocking probability with respect to call arrival rate or load, to increase the overall performance of the system. Specifically, the primary focus is on the development, analysis and optimization of the PITG model and evaluation of a gateway selection algorithm.
- Install, configure, deploy, secure, and maintain Asterisk - Build a fully-featured telephony system and create a dial plan that suits your needs - Learn from example configurations for different requirement - Implement 3rd party applications which will help you manage Asterisk through a web interface Asterisk is a powerful and flexible open source framework for building feature-rich telephony systems. As a Private Branch Exchange (PBX) which connects one or more telephones, and usually connects to one or more telephone lines, Asterisk offers very advanced features, including extension-to-extension calls, queues, ring groups, line trunking, call distribution, call detail rerecords, and call recording. This book will show you how to build a telephony system for your home or business using this open source application. 'Asterisk 1.6' takes you step-by-step through the process of installing and configuring Asterisk. It covers everything from establishing your deployment plan to creating a fully functional PBX solution. Through this book you will learn how to connect employees from all over the world as well as streamline your callers through Auto Attendants (IVR) and Ring Groups. This book is all you need to understand and use Asterisk to build the telephony system that meets your need. You will learn how to use the many features that Asterisk provides you with. It presents example configurations for using Asterisk in three different scenarios: for small and home offices, small businesses, and Hosted PBX. Over the course of ten chapters, this book introduces you to topics as diverse as Public Switched Telephony Network (PSTN), Voice over IP Connections (SIP / IAX), DAHDI, libpri, through to advanced call distribution, automated attendants, FreePBX, and asterCRM. With an engaging style and excellent way of presenting information, this book makes a complicated subject very easy to understand. An easy introduction to using and configuring Asterisk to build feature-rich telephony systems for small, medium and large businesses What you will learn from this book : - Install, configure, and deploy Asterisk to build a fully-featured telephony system - Install and use FreePBX - Connect your Asterisk server with your phone service (via PSTN, SIP, etc) as well as learn to deploy some basic PBX features such as queues, voicemail and music on hold - Determine how calls are routed through the Asterisk server by creating a dialplan - Create extensions, distribute calls in an orderly fashion using queues, and present callers with a greeting using automated attendants (IVR) - Install and learn how to monitor, record, and capture detailed call logs - Install and use asterCRM (customer relation management solution) to streamline your business operations - Gain knowledge of security precautions, network deployment recommendations as well as maintenance tips such as backups and preparing disaster recovery plans for keeping the Asterisk system running smooth and secure Who this book is written for This book is aimed at anyone who is interested in building a powerful telephony system using the free and open source application, Asterisk, without spending many thousands of dollars buying a commercial and often less flexible system. This book is suitable for the novice and those new to Asterisk and telephony. Telephony or Linux experience will be helpful, but not required.
- 16, 24 or 32 FXS or FXO—Simultaneous voice or fax calls on all ports. Advanced local call switching.- Full SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem bypass, DTMF relay. Codecs G.729, G.723 etc.- Outstanding Interoperability—Interoperable for voice and T.38 fax with leading SIP service providers, soft-switch vendors and Asterisk IP-PBXThe SmartNode 4300 VoIP Gateway provides 12 to 32 analog FXS or FXO interfaces to connect phones, fax or PSTN trunk lines to your IP-based communications (IP PBX, UC systems and SIP Trunks).Like every SmartNode, the SN4300 supports every industry-standard CODEC to deliver toll-quality voice on every call. The Unified Communications Agent™ (UCA) provides any-to-any multi-path switching (simultaneous SIP, H.323, ISDN, and POTS calls with routing and conversion between TDM/PSTN and IP/Ethernet networks—plus T.38 and SuperG3 FAX). VoIP-over-VPN with voice encryption provides secure voice and data via IPsec with AES/DES strong encryption and automated keying via Internet Key Exchange (IKE). In addition, advanced call-router functionality includes least-cost call routing with flexible dialed-number plan support.The SmartNode survivability suite provides PSTN fallback to ensure business continuity in case the IP network fails. In addition, SmartNode delivers a smooth transition to VoIP with strong number portability support accepting incoming calls from the PSTN throughout the VoIP service provider's number porting process.Preserve investments in legacy phone equipment while taking the next steps toward unified communications with the SN4300 VoIP Gateway. Providing 12 to 32 FXS or FXO interfaces and one 10/100/1000 Ethernet, the SN4300 delivers a reliable, cost-effective solution for the Enterprise.