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Asterisk (PBX)
45,00 € *
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High Quality Content by WIKIPEDIA articles! Asterisk is a software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. Its name comes from the asterisk symbol. Asterisk is released under a dual license model, using the GNU General Public License (GPL) as a free software license and a proprietary software license to permit licensees to distribute proprietary, unpublished system components.

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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Patton SmartNode 4324
1.125,06 € *
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- 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.

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Stand: 23.01.2020
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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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Patton SmartNode 4324
1.122,24 € *
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- 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.

Anbieter: Dodax
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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Patton SN4118 Gateway/Controller
434,01 € *
ggf. zzgl. Versand

Features - Up to 8 FXS and/or FXO ports—Compact, reliable stand-alone VoIP gateway with different port options. Supports simultaneous voice or fax calls on all ports. (note: Patton does not carry an 8-FXO unit at the present time) - Advanced Local Call Switching—Virtual interfaces and routing tables provide industry leading flexibility in call handling programming. Local call switching, soft fallback to alternative routes. Simultaneously connects to multiple SIP services/IP PBXs. - Complete SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem handling, DTMF relay. Codecs G.729, G.723, etc. - Easy Management & Provisioning—Web-based management, SNMP, command line interface. Automated mass provisioning for efficient large-scale deployments. - Outstanding Interoperability—Proven integration for voice and T.38 fax with 3CX®, Asterisk™, PingTel™ and other leading IP PBX systems and soft switch vendors. - Supported by SmartNode™ Redirection Service: A free service enabling zero-touch mass deployments for Service Providers and Distributors with auto-provisioning servers (Learn More). Overview The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 Series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports, the SN4112S connects to any legacy telephone or PBX and provides dial-tone, ringing, and caller-ID. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically select the least-cost-route while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPsec VPN with AES/3DES guarantees secure voice over the public network. Patton’s SmartNode SN4110 Series delivers the legacy phone interfaces, service transparency, and flexible PSTN integration required for true converged packet voice. Applications Remote Office/Branch Office Voice Extension and Access In enterprise networks, transparent access to PBX features while using existing equipment is key to low-cost operations. Now, instead of installing a separate PBX at the remote office, the SmartNode 4110 Series is able to provide transparent extension while simultaneously connecting multiple locations. The extensions can be managed centrally and benefit from PBX services such as calling groups, least cost routing, and call forwarding. PSTN access allows local calls to be processed without using corporate remote PBX resources.  Additionally, the corporate PBX can break-out and bypass any long distance charges by using the remote office for the local gateway.

Anbieter: Dodax AT
Stand: 23.01.2020
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Patton SN4118 Gateway/Controller
429,77 € *
ggf. zzgl. Versand

Features - Up to 8 FXS and/or FXO ports—Compact, reliable stand-alone VoIP gateway with different port options. Supports simultaneous voice or fax calls on all ports. (note: Patton does not carry an 8-FXO unit at the present time) - Advanced Local Call Switching—Virtual interfaces and routing tables provide industry leading flexibility in call handling programming. Local call switching, soft fallback to alternative routes. Simultaneously connects to multiple SIP services/IP PBXs. - Complete SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem handling, DTMF relay. Codecs G.729, G.723, etc. - Easy Management & Provisioning—Web-based management, SNMP, command line interface. Automated mass provisioning for efficient large-scale deployments. - Outstanding Interoperability—Proven integration for voice and T.38 fax with 3CX®, Asterisk™, PingTel™ and other leading IP PBX systems and soft switch vendors. - Supported by SmartNode™ Redirection Service: A free service enabling zero-touch mass deployments for Service Providers and Distributors with auto-provisioning servers (Learn More). Overview The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 Series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports, the SN4112S connects to any legacy telephone or PBX and provides dial-tone, ringing, and caller-ID. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically select the least-cost-route while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPsec VPN with AES/3DES guarantees secure voice over the public network. Patton’s SmartNode SN4110 Series delivers the legacy phone interfaces, service transparency, and flexible PSTN integration required for true converged packet voice. Applications Remote Office/Branch Office Voice Extension and Access In enterprise networks, transparent access to PBX features while using existing equipment is key to low-cost operations. Now, instead of installing a separate PBX at the remote office, the SmartNode 4110 Series is able to provide transparent extension while simultaneously connecting multiple locations. The extensions can be managed centrally and benefit from PBX services such as calling groups, least cost routing, and call forwarding. PSTN access allows local calls to be processed without using corporate remote PBX resources.  Additionally, the corporate PBX can break-out and bypass any long distance charges by using the remote office for the local gateway.

Anbieter: Dodax
Stand: 23.01.2020
Zum Angebot