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 station-to-station calls, line trunking, call distribution, call detail rerecords, and call recording.

Provides information on Asterisk, an open source telephony application.

Design a complete Voice over IP (VoIP) or traditional PBX system with Asterisk, even if you have only basic telecommunications knowledge. This bestselling guide makes it easy, with a detailed roadmap that shows you how to install and configure this open source software, whether you’re upgrading your existing phone system or starting from scratch. Ideal for Linux administrators, developers, and power users, this updated edition shows you how to write a basic dialplan step-by-step, and brings you up to speed on the features in Asterisk 11, the latest long-term support release from Digium. You’ll quickly gain working knowledge to build a simple yet inclusive system. Integrate Asterisk with analog, VoIP, and digital telephony systems Build an interactive dialplan, using best practices for more advanced features Delve into voicemail options, such as storing messages in a database Connect to external services including Google Talk, XMPP, and calendars Incorporate Asterisk features and functions into a relational database to facilitate information sharing Learn how to use Asterisk’s security, call routing, and faxing features Monitor and control your system with the Asterisk Manager Interface (AMI) Plan for expansion by learning tools for building distributed systems

Build high-speed and highly scalable telephony systems using OpenSIPS About This Book Install and configure OpenSIPS to authenticate, route, bill, and monitor VoIP calls Gain a competitive edge using the most scalable VoIP technology Discover the latest features of OpenSIPS with practical examples and case studies Who This Book Is For If you want to understand how to build a SIP provider from scratch using OpenSIPS, then this book is ideal for you. It is beneficial for VoIP providers, large enterprises, and universities. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS. Telephony and Linux experience will be helpful to get the most out of this book but is not essential. Prior knowledge of OpenSIPS is not assumed. What You Will Learn Learn to prepare and configure a Linux system for OpenSIPS Familiarise yourself with the installation and configuration of OpenSIPS Understand how to set a domain and create users/extensions Configure SIP endpoints and make calls between them Make calls to and from the PSTN and create access control lists to authorize calls Install a graphical user interface to simplify the task of provisioning user and system information Implement an effective billing system with OpenSIPS Monitor and troubleshoot OpenSIPS to keep it running smoothly In Detail 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. Style and approach This book is a step-by-step guide based on the example of a VoIP provider. You will start with OpenSIPS installation and gradually, your knowledge depth will increase.

Design a complete Voice over IP (VoIP) or traditional PBX system with Asterisk, even if you have only basic telecommunications knowledge. This bestselling guide makes it easy with a detailed roadmap that shows you how to install and configure this open source software, whether you’re upgrading your existing phone system or starting from scratch. Ideal for Linux administrators, developers, and power users, this updated fifth edition shows you how to set up VoIP-based private telephone switching systems within the enterprise. You’ll get up to speed on the features in Asterisk 16, the latest long-term support release from Digium. This book also includes new chapters on WebRTC and the Asterisk Real-time Interface (ARI). Discover how WebRTC provides a new direction for Asterisk Gain the knowledge to build a simple but complete phone system Build an interactive dialplan, using best practices for Asterisk’s advanced features Learn how ARI has emerged as the API of choice for interfacing web development languages with Asterisk

Asterisk has a wealth of features to help you customize your PBX to fill very specific business needs. This short cookbook offers recipes for tackling dialplan fundamentals, making and controlling calls, and monitoring channels in your PBX environment. Each recipe includes a simple code solution you can put to work immediately, along with a detailed discussion that offers insight into why and how the recipe works. This book focuses on Asterisk 1.8, although many of the conventions and information presented are version-agnostic. These recipes include solutions to help you: Authenticate callers before moving on in your dialplan Redirect calls received by your auto-attendant Create an automatic call-back service Initiate hot-desking to login to and accept calls at any office device Monitor and interrupt live calls to train new employees at a call center Record calls from your Asterisk dialplan

More and more businesses today have their receive phone service through Internet instead of local phone company lines. Many businesses are also using their internal local and wide-area network infrastructure to replace legacy enterprise telephone networks. This migration to a single network carrying voice and data is called convergence, and it's revolutionizing the world of telecommunications by slashing costs and empowering users. The technology of families driving this convergence is called VoIP, or Voice over IP. VoIP has advanced Internet-based telephony to a viable solution, piquing the interest of companies small and large. The primary reason for migrating to VoIP is cost, as it equalizes the costs of long distance calls, local calls, and e-mails to fractions of a penny per use. But the real enterprise turn-on is how VoIP empowersbusinesses to mold and customize telecom and datacom solutions using a single, cohesive networking platform. These business drivers are so compelling that legacy telephony is going the way of the dinosaur, yielding to Voice over IP as the dominant enterprise communications paradigm. Developed from real-world experience by a senior developer, O'Reilly's Switching to VoIP provides solutions for the most common VoIP migration challenges. So if you're a network professional who is migrating from a traditional telephony system to a modern, feature-rich network, this book is a must-have. You'lldiscover the strengths and weaknesses of circuit-switched and packet-switched networks, how VoIP systems impact network infrastructure, as well as solutions for common challenges involved with IP voice migrations. Among the challenges discussed and projects presented: building a softPBX configuring IP phones ensuring quality of service scalability standards-compliance topological considerations coordinating a complete system ?switchover? migrating applications like voicemail and directoryservices retro-interfacing to traditional telephony supporting mobile users security and survivability dealing with the challenges of NAT To help you grasp the core principles at work, Switching to VoIP uses a combination of strategy and hands-on how-to that introduce VoIP routers and media gateways, various makes of IP telephone equipment, legacy analog phones, IPTables and Linux firewalls, and the Asterisk open source PBX software by Digium.You'll learn how to build an IP-based or legacy-compatible phone system and voicemail system complete with e-mail integration while becoming familiar with VoIP protocols and devices. Switching to VoIP remains vendor-neutral and advocates standards, not brands. Some of the standards explored include: SIP H.323, SCCP, and IAX Voice codecs 802.3af Type of Service, IP precedence, DiffServ, and RSVP 802.1a/b/g WLAN If VoIP has your attention, like so many others, then Switching to VoIP will help you build your own system, install it, and begin making calls. It's the only thing left between you and a modern telecom network.