- 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.
Softswitch is an innovative concept in the field of telecommunication and ICT, which creates new challenges and opportunities for the network operators, service providers and researchers. The migration from PSTN to NGN involves the convergence between PSTN and IP network that is a single network for voice, data and video. Softswitch performs all analysis functions, routes signalling messages and handles traffic data. So it requires an extension of existing charging collection mechanisms and billing system that is functionally intelligent and flexible, and provides fair policy towards end users. The book is designed to present the design principle and implementation concept of Softswitch. It also includes different convergence billing protocols and their interface challenges. In addition, the operational results at the end demonstrate the effectiveness of the Softswitch with billing interface. On the whole, the study has sought to explicate an integrated system for Charging, Accounting and Billing, which is capable of handling all the challenges and generating fair traffic bill.
National Telecom Limited (NTC) was incorporated on June 8, 2004 for the purpose of implementing, owning and operating a fixed line network in Bangladesh. NTC was awarded a PSTN license by the Bangladesh Telecom Regulatory Commission (BTRC), and the Government of Bangladesh (GoB) to operate basic telephone services, data and broadband services in all over Bangladesh. The license was granted for a period of 20 years and is renewable prior to expiry. The authorised capital of NTC is Bangladesh Taka (BDT) 5 billion (approx US$ 73 million) divided into 50,000,000 ordinary shares of BDT 100 each. 1 US$ = 68.7 BD National Telecom, received a national license on 23 September 2007 and is one of the six PSTN operators to operate PSTN service all over Bangladesh. The license was granted for a period of 15 years and is renewable prior to expiry and NTC had to surrender all previous (four Zone) licenses. NTC also received one spot of 450 MHz and one spot of 1900 MHz frequency from BTRC.
Patton SmartNode 4120/1Bis2V/eui BRI Pstn VoIP Gateway 1 BRI/So TE 2 voi fax calls 1x 10/100 Ethern. ext. UI power 1 Jahr Garantie (Sn4120/1Bis2V/eui)
Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Yate is a telephony engine, a softswitch with PBX capabilities, originally created in 2004 by Paul Chitescu of NullTeam. It is released under a GNU General Public License with an exception for linking with OpenH323 and PWlib (licensed under MPL). Started as a free/open source softswitch for connecting PSTN with VoIP, support for other telephony technologies like conference, PBX, IVR has been added later on. During the years Yate has evolved into a Unified Communications server providing advanced integration with instant messaging, video and fax. Historically the basic function of Yate is the softswitch, connecting together call legs either to external devices or internal server resources. The logic for routing calls uses messages and can be implemented in configuration files, database queries or external programs. Call signalling is possible using PSTN protocols like ISDN or SS7, either directly attached over a TDM interface card (T1, E1, BRI), analog interface or by a remote Signaling gateway. A variety of VoIP protocols can be used without the need of special hardware: SIP, H.323, IAX2 or Jingle
Compared to traditional (PSTN) voice networks, a Voice over Internet Protocol network is a convergence of a signaling network and a data network using Internet Protocol (IP). The use of shared media by VoIP systems opens the door to some uncertainty as to the source of a call. While in the traditional voice networks one has to tap into a specific circuit to eavesdrop, in an IP network any equipment connected to the target LAN can identify, store and playback the VoIP packets that traverse that LAN. An unprotected, unauthenticated IP network makes VoIP susceptible to hostile use, such as call hijacking, connection tear down, denial of service, or sending computer viruses over the network. In this work, we perform a series of attacks against a VoIP application, and prove that they succeed with nothing more than a couple of identity tokens captured from the network traffic as prerequisites. We then design an Intrusion Detection System implementing a gradual attack-response procedure, destined to inform and protect the End-Users of the Application Under Test.
Existing wireless networks enable different important applications over IP based network such as public internet and Voice over Internet protocol is one of the important applications which have become a possible alternative to public switched telephone network (PSTN). This project investigates one of the time sensitive data applications on WiMAX i.e. VoIP. I carried on WiMAX because WiMAX networks provide advance features and protocols to support the Quality of service (QoS). This project also indicates about the impact of load and mobility on the voice calls. This book provides details about the Voice Over Internet Protocol (VoIP), Which includes types of VoIP calls, VoIP System and Components, VoIP Protocols, VoIP codecs and VoIP Quality of Service (QoS). It also provides detailed study of WiMAX including WiMAX Protocol Layer and WiMAX Quality Of Service(QoS)
Amplitude-shift keying (ASK) is a form of modulation that represents digital data as variations in the amplitude of a carrier wave. The amplitude of an analog carrier signal varies in accordance with the bit stream (modulating signal), keeping frequency and phase constant. The level of amplitude can be used to represent binary logic 0s and 1s. We can think of a carrier signal as an ON or OFF switch. In the modulated signal, logic 0 is represented by the absence of a carrier, thus giving OFF/ON keying operation and hence the name given. Like AM, ASK is also linear and sensitive to atmospheric noise, distortions, propagation conditions on different routes in PSTN, etc. Both ASK modulation and demodulation processes are relatively inexpensive. The ASK technique is also commonly used to transmit digital data over optical fiber. For LED transmitters, binary 1 is represented by a short pulse of light and binary 0 by the absence of light. Laser transmitters normally have a fixed "bias" current that causes the device to emit a low light level. This low level represents binary 0, while a higher-amplitude lightwave represents binary 1.
Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. In telephony, the local loop is the physical link or circuit that connects from the demarcation point of the customer premises to the edge of the carrier or telecommunications service provider''s network. At the edge of the carrier access network in a traditional PSTN scenario, the local loop terminates in a circuit switch housed in an ILEC CO. Traditionally, the local loop was wireline in nature from customer to central office, specifically in the form of an electrical circuit provisioned as a single twisted pair in support of voice communications. Where the number of local loops was restricted, different customers could share the same loop, known as a party line. Modern implementations may include a digital loop carrier system segment or fiber optic transmission system known as fiber-in-the-loop. The local loop may terminate at a circuit switch owned by a CLEC and housed in a POP, which typically is either an ILEC CO or a "carrier hotel". A local loop may be provisioned to support data communications applications, or combined voice and data such as digital subscriber line.
The GXW410x series of FXO IP Gateways series offers the small enterprise, SOHO, remote offices and multi-location enterprises a cost-effective, easy to deploy VoIP FXO solution. The GXW410x series allows any business to seamlessly connect devices within an office or multiple offices with up to 8 PSTN lines, to an IP PBX system, or with an existing traditional phone system.
High Quality Content by WIKIPEDIA articles! Teletraffic engineering is the application of traffic engineering theory to telecommunications. Teletraffic engineers use their basic knowledge of statistics including Queueing theory, the nature of traffic, their practical models, their measurements and simulations to make predictions and to plan telecommunication networks at minimum total cost. These tools and basic knowledge help provide reliable service at lower cost. Because the approach is so different to different networks, the networks are handled separately here: the PSTN, broadband networks, mobile networks, and networks where the possibility of traffic being heavy is more frequent than anticipated.
Die GXW FXO-Serie ist die ideale und kosteneffektive VoIP-Lösung für Unternehmen, Kleinbetriebe und Remote-Offices, welche diverse Standorte nutzen und diese nahtlos mit bis zu 8 PSTN-Leitungen, einem IP PBX-System oder mit einer bereits vorhand