Angebote zu "Telephony" (31 Treffer)

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PSTN-IP Telephony Gateway for Ensuring QoS in H...
41,10 € *
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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.

Anbieter: Dodax AT
Stand: 23.01.2020
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PSTN-IP Telephony Gateway for Ensuring QoS in H...
39,90 € *
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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.

Anbieter: Dodax
Stand: 23.01.2020
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Packt Asterisk 1.6 Software-Handbuch 240 Seiten
46,14 € *
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- 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.

Anbieter: Dodax
Stand: 23.01.2020
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Packt Asterisk 1.6 Software-Handbuch 240 Seiten
46,13 € *
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- 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.

Anbieter: Dodax AT
Stand: 23.01.2020
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IP Telephony Over IUG Network
50,40 € *
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Traditional telephony last about 100 years in use as the basic voice communication because of its reliability that satisfy the normal needs. Packet switching network, especially the Internet, which rapidly spread , attract more and more applications, because of its flexibility and efficiency. One of the most important applications is VoIP, which transmit voice in the same method of transmitting data, taking into account that voice is a real time application. Once we can transmit the voice over IP network, we do not need to route calls using the expensive and large central switches of PSTN, since we can route the calls in the same manner that we route data in IP network. This is the task of the soft switch, soft switch may seem to be the counterpart of central office or PBX of PSTN network. Finally, this project outlines the overall system of VoIP communication, and then show how to implement a whole telephone system based on IP protocols. The implementation of this project will improve the telephony service and management in (IUG).

Anbieter: Dodax AT
Stand: 23.01.2020
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IP Telephony Over IUG Network
49,00 € *
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Traditional telephony last about 100 years in use as the basic voice communication because of its reliability that satisfy the normal needs. Packet switching network, especially the Internet, which rapidly spread , attract more and more applications, because of its flexibility and efficiency. One of the most important applications is VoIP, which transmit voice in the same method of transmitting data, taking into account that voice is a real time application. Once we can transmit the voice over IP network, we do not need to route calls using the expensive and large central switches of PSTN, since we can route the calls in the same manner that we route data in IP network. This is the task of the soft switch, soft switch may seem to be the counterpart of central office or PBX of PSTN network. Finally, this project outlines the overall system of VoIP communication, and then show how to implement a whole telephone system based on IP protocols. The implementation of this project will improve the telephony service and management in (IUG).

Anbieter: Dodax
Stand: 23.01.2020
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YATE Yet Another Telephony Engine
50,40 € *
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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

Anbieter: Dodax AT
Stand: 23.01.2020
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YATE Yet Another Telephony Engine
49,00 € *
ggf. zzgl. Versand

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

Anbieter: Dodax
Stand: 23.01.2020
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IP Phone
29,90 € *
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An IP phone uses Voice over IP technologies allowing telephone calls to be made over an IP network such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP Network such as that of a company. The phones use control protocols such as Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype. IP phones can be simple software-based Softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA).It may have many features an analog phone doesn't support, such as e-mail-like IDs for contacts that may be easier to remember than names or phone numbers.

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Stand: 23.01.2020
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