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Snom DECT M85
Snom DECT M85 is a DECT telephone that notes for its quality finish and can withstand shock, splashes and dust that make it an ideal for industrial use. In addition, it's intended for outdoor use by diversifying further possible usage scenarios.
Moreover, it is easily configurable with multicellular M700 and unicellular M300 solution.
Main Features
Range
Os telefones profissionais Snom 300 preenchem os requisitos mais importantes que são exigidos na telefonia VoIP e, além disso, oferecem todas as funções necessárias para o dia a dia de uma empresa. Eles tambén são compatíveis com a maioria dos sistemas PBX e está disponível uma ampla gama que incorpora diferentes funcionalidades dependendo das necessidades de cada professional: displays gráficos retroiluminados ou coloridos, compatibilidade bluetooth, teclas contextuais e switch BLF, PoE ou Gigabit, entre outros. Especificamos abaixo os modelos que compõem esssa faixa 300: Snom 300, Snom D320, Snom D305, Snom D315, Snom D345 e Snom D375.
Os fones de ouvido Snom são ideais para pessoas cuja ferramenta de trabalho é o telefone, porque esse acessório permite maior liberdade de movimento ao combinar chamadas com outras tarefas; Além disso, eles são projetados para tirá-los de lado (á esquerda ou à direita, alternadamente), garantindo
O klarVOICE é um aparelho 'handset' que usa banda larga o codec G.722 para que você possa ouvir sons de alta qualidade.
A fim de estender os recursos do seu telefone VoIP, a Snom oferece a possibilidade de expandir seu terminal adicionando um teclado. Atualmente, três modelos estão disponíveis: D3, D7 e Snom Vision.
Você precisa de um telefone sem fio para suas comunicações profissionais? Você precisa de uma solução sem fio em vários andares ou dentro de grandes edifícios? Dê uma olhada na série Snom DECT: M25, M65, M85, M700 Multicelda ou M215SC.
Liberdade de movimento para sua atividade profissional!
Avanzada 7 oferece uma série de produtos que complementam o uso e a experiência da gama Snom. Entre estes produtos complementares você pode encontrar:
Fonte de alimentação externa para os modelos Snom IP 7XX e 8XX
Eles permiten aumentar significativamente o alcance da série Snom DECT, proporcionando, por sua vez, melhor cobertura de rádio em ambientes com maio dificuldade. Eles são freqüentemente usados quando a faixa de telefones sem fio precisa ser ampliada, ou quando as condições de recepção em áreas remotas precisam ser reforçadas.
Ele tem duas antenas internas e, graças a isso, pode ser usado para dobrar o alcance real de uma estação base DECT. Na mesma base, até seis repetidores DECT podem ser conectados.
O adaptador Snom para fones de ouvido sem fio atua como uma ponte entre a telefonia VoIP e fones de ouvido sem fio. Este adaptador foi projetado especificamente para os terminais Snom 320, 370 e 820.
Endereço público IP
A Snom disponibiliza ao usuário o IP Paging System PA1, um equipamento utilizado em pisos de escritórios, áreas de recepção, salas de espera tanto em ambientes industriais como em pequenos escritórios.
KX-TPA65 - Wireless Desktop IP Terminal
The KX-TPA65 IP Terminal is a revolutionary VoIP desktop terminal that is also very latest wireless offering advantages over traditional desktop phones; as well as including the performance of a traditional desktop telephone (wired) incorporates DECT funcionalities avoiding the need for a LAN cable when installing the terminal.
In addition, it is compatible with the KX-TPA600 system and works on any combination of KX-TPA60 terminals, KX-TPA65, KX-UDT121 or KX-UDT131 (KX-TPA60 terminals and up to 7 more).
Products highlights
Panasonic KX - HDV130 Black
The Panasonic KX-IP Desktop Phone - HDV130 offers the ideal solution for any business environment. Fine and elegant design that adapts perfectly to different desks offering maximum comfort to the user. In addition, thanks to its 2.3'' LCD screen with 132x64px of resolution, it offers up to four lines of informacion.
To deliver crip communications, the KX-HDV130 integrates a combination of HD audio features, such as full duplex, acoustic full duplex, acoustic echo cancellation and hardware and software packet loss concealment. In this way, this IP solution guarantees the highest voice quality and performance associated with broadband communications.
Key features
Wireless extension Panasonic KX-TPA60
The Panasonic KX-TPA60 terminal is a wireless extension phone compatible with the KX-TGP600 cordless phone system. It allows you to expand the capacity of its communications system and furthermore, provides great coverage and flexibility in combination with KX-A406.
It's compatible with the KX-TGP600 system and works on any combination of terminals KX-TPA60, KX-TPA65, KX-UDT121 or KX-UDT131.
Features
KX-TGP600 - Intelligent Wireless IP Phone System
El KX-TGP600 is a very simple phone to install and configure thanks to its self-provisioning, which allows remote settings and manage system terminals wireless telephone system. It also allows up to 8 handsets connected to the system allowing up to 8 different phone numbers.
Another of its most notable features is its sound quality thanks to noise reduction function and automatic volum adjustment depending on where the user is located.
The GXP1628 IP Phone, versatile, affordable and Gigabit
The GXP1628 is a powefull IP Phone recommended for Small and medium Bussinesses (SMB)
The new GXP1628 is based on Linux OS and includes 2 lines, 8 BLF keys and 3-way conferencing . A 132x48 backlit LCD sreen creates a great display for easy customer experience
Moreover in its features are characteristics as the dual switched gigabit network ports, the HD audio, the integrated PoE and more than 12 languages are supported.
All these characteristics make the GXP1628 to be a versatile, high quuality, and dependable office phone.
Os terminais IP da Grandstream Executive são a nova geração de telefones IP. Eles oferecem as mais altas funções de telefonia para empresas que precisam de recursos específicos, como várias linhas e identidades SIP, áudio de alta qualidade, PoE integrado e interoperabilidade extensiva. Eles representam a opção perfeita para empresas que precisam de um telefone IP executivo multilinha de alta qualidade a um custo acessível.
Os terminais Grandstream IP standard representam uma nova geração de telefones IP ideal como ferramenta de trabalho para pequenas e médias empresas.
Qualidade de áudio superior, serviço de aplicativos personalizável, teclas programáveis, chamada em espera, transferÇência e encaminhamento de chamadas, conferência tripartite e cancelamento de eco acústico são alguma das principais funções que tornam esses telefones da Grandstream terminais de última geração a un preço acessível.
A série Grandstream DECT é apresentada como a nova geração de telefone IP sem fio de alta qualidade e fácil de usar. Oferece uma ampla variedade de recursos e uma ampla gama de cobertura de rádio.
Suas características de tamanho compacto, excelente qualidade de voz e excelentes funcionalidades fazem dele um conjunto com excelente relação qualidade / preço.
Os adaptadores de telefone analógico (ATA) da Grandstream oferecem a possibilidade de implementar serviços comerciais de voz sobre IP através de um telefone analógico. A Grandstream oferece os seguintes modelos ao usuário.
As câmeras de vigilância por vídeo da Grandstream estão preparadas para qualquer ambiente externo ou interno, onde as condições climáticas e de iluminação mudam constantemente.
Suas lentes permitem uma adaptação de acordo com as necessidades de monitoramento dos usuários permitindo monitorar áreas próximas, entradas para edifícios, etc.
A Grandstream tem uma ampla gama de Gateways ideais para qualquer empresa que queira incorporar a nova tecnologia IP em sua atividade diária. Além disso, apresenta-se como a solução ideal para rentabilizar o investimento realizado em um equipamento de telefonia analógica, fax e sistemas tradicionais de PABX.
- Á série GXW4004/4008 representa a solução ideal para empresas que desejam conectar uma ou mais linhas de um PBX tradicional a um sistema de telefonia VoIP com 4 e 8 portas FXS, respectivamente.
- Os modelos GXW4104 / 4108 convertem chamadas IP SIP / RTP em chamadas PSTN tradicionais. Com 4 e 8 portas FXO, respectivamente, a instalação é idêntica em ambos os modelos e oferece interoperabilidade completa com sistemas IP PBX, softs whitches e servidores SIP.
- O GXW4216/4224/4232/4248 tem 16/24/32 e 48 portas FXS de telefonia analógica, repectivamente. Além disso, eles incorporam porteção de segurançza avançada, excelente qualidade de voz, provisionamento fácil e excelente desempenho no tratamento de altos volumes de chamadas de voz.
No portfólio da Grandstream, também encontramos esse tipo de solução, ideal para salas de reunião ou reunioes de qualquer escritório, graças à sua excelente mobilidade e qualidade de áudio. Possui Bluetooth, WiFi, confêrencia de 7 vias e conexão de uma segunda unidade GAC2500 no modo "Daisy-chain" (em cadeia) com um alto-falante secundário e outro microfone de expansão: por isso, sua flexibilidade e mobilidade são as principais qualidades diferenciadoras deste equipamento de audioconferência.
Além disso, com base no Android 4.4, ele oferece suporte aos aplicativos de Google App Store, como o Hangouts do Google, Skype para Empresas (Lync), navegador da Internet, Adobe Flash, Twitter, Facebook, Youtube, calendário do Google, importação /exportação de dados. Telefone celular via Bluetooth, etc. API / SDK disponível para desenvolvimento avançado de aplicativos personalizados.
Addcom USB Cord - QD to USB with DSP (PN:ADDQD-76)
This cord can be used as an adaptor for a QD connection to transfer it to a USB connection.
- Tested with Microsoft Lync: ok.
The all-in-one platform
Sentinel inherits the VoIP gateway and media adapter capabilities from the class-leading Mediatrix product portfolio and is scalable up to 240 channels to convert PSTN and Legacy PBX communications into SIP-based communications.
Sentinel is also a SBC, addressing a variety of applications including Demarcation Point, Security, SIP Normalization, Survivability, and more. With a Back-to-Back-User-Agent, Sentinel acts as the middle-man for communications between two SIP endpoints. It permits applying deep packet inspection, identifies specific properties attached to a SIP Request, and executes predefined actions to manage any particular deployment scenario. Sentinel also supports media relay for media anchoring, NAT traversal, and transcoding.
Flexible like never before
Sentinel is a multi-service business platform bundling Session Border Controller and Media Gateway capabilities into a robust, field-upgradable platform designed for medium and large enterprises. Sentinel is ideally targeted for applications ranging from 30 up to 600 simultaneous sessions.
Advantages
Sentinel permits Service Providers to deploy SIP Trunk and Hosted Services with compelling session border controller (SBC) capabilities. It delivers a flexible architecture designed to enable administrators to easily modify SIP and Media signaling to comply with any devices deployed on the customer premises.
Specifications
Sentinel enables cost-effective and profitable VoIP deployments into medium and large enterprises for both Session Border Controller (SBC) and SIP Gateway applications. It supports a large variety of telephony interfaces and is the ideal solution to deploy private and hosted toll bypass networks. It provides a simple, transparent, and cost-effective way to derive the benefits of VoIP services while maintaining a connection to the PSTN.
A série Mediatrix 4400 é a solução mais econômica da Mediatrix, que inclui modelos de gateway ISDN com portas BRI, ISDN e Ethernet, tornando-a a opção mais robusta e versátil para emrpesas. Esta série de gateways digitais para VoIP permite a conexão d eequipamentos ISDN, como PBXs, através da interface BRI para uma rede IP ou como um gateway para a rede PSTN.
A série C7 da Mediatrix oferece uma série de gateways que combinam interfaces FXS e FXO para integrar vários aplicativos em uma única plataforma. Esses Gateways VoIP apresentam uma excelente relação preço / qualidade e são ideais para conectar redes de pequenas e médias empresas a uma rede IP.
Os modelos C710, C711, C730, C731 e C733 estão atualmente disponíveis; dependendo do modelo, eles podem apresentar de 4 a 8 portas telefônicas e conectar até 8 telefones analógicos, modems, fax ou até 8 linhas RTC ou troncos de um PBX à rede IP.
Um dos principais produtos da Mediatrix é o Sentinel, um produto integrador que tem a capacidade de um Controlador de Borda de Sessão (SBC) e um Gateway personalizável. É ideal para empresas de médio e grande porte que precisam de implementações de VoIP com arquitectura flexível, incluindo tolerância a falhas, normalização SIP e ponto de demarcação.
A estação base do nosso Sentinel apresenta inicialmente:
Graças a esses 8 slots, podemos personalizar nosso Sentinel da maneira que queremos, combinando os cartões e as licenás que melhor atendam às nossas necessidades.
MÓDULOS
Licencias SBC
TE436 (4E1 PCI profile) Digital Telephony Cards Quad Span (1TE436F - Four span digital T1/E1/J1/PRI PCI card)
Digium's quad span digital interface cards support 96 (T1 / J1) or 120 (E1) connections to PSTN trunks over four spans (digital circuits). Built exclusively for use with Asterisk and Asterisk-based communications systems, the quad span digital cards provide the best value in digital connectivity.
Quad span digital cards are available in PCI form factors. All quad span models can be combined with an Octasic DSP-based echo cancellation module from Digium to provide effective hardware echo cancellation across all channels.
Digium cards are compatible with all versions of Asterisk using the DAHDI driver framework. Asterisk and DAHDI are available for free from the Asterisk.org website. All cards include a five (5) year warranty and are eligible for Digium's Exceptional Satisfaction Program (ESP) risk-free Quality Guarantee.
Applications
As placas Digium da série TE são de alto desempenho e custo efetivo, com interfaces telefônicas digitais que suportam os ambientes T e E. Os ambientes são seleccionáveis em uma base por cartão ou por porta. Esta característica permite a tradução da sinalização entre os equipamentos T1 e E1, e permite conectar bancos de canais econômicos T1 com circuitos E1.
Às vezes, pode acontecer que o seu sistema de comunicações Ip seja ecoado, isso pode acontecer como resultado dos tempos de espera que um sistema VoIP geralmente tem, ao contrário de um sistema analógico. Como resultado, sua conversa pode sofrer um eco.
Embora muitos dispositivos já incorporem o cancelador de eco, a Digium oferece o hardware de cancelamento de eco de alto desempenho da Digium (HPEC). O hardware de cancelamento de eco também é vantajoso ao lidar com grandes volumes de chamadas ou um grande número de canais que, de outra forma, sobrecarregariam a CPU, o que resultaria em baixa qualidade potencial de áudio. Aqui estão alguns de seus recursos:
Os Gateways Digium VoIP oferecem o melhor valor agregado para a conexão de telefonia tradicional (T1/E1/PRI) para IP (SIP).
O software de gateway é baseado no Asterisk é gerenciado por meio de uma interface gráfica do usuário (GUI), que permite fácil navegação e configuração. Eles têm um design integrado de economia de energia com um processador de sinal digital (DSP) altamente eficiente, permitindo excelente manuseio de todas as operações relacionadas à mídia.
Digium TE235 (2E1 PCI Express-low profile) Digital Telephony Cards Dual Span (1TE235F - Two span digital T1/E1/J1/PRI PCI-Express x1 card)
Digium's dual span digital interface cards support 48 (T1 / J1) or 60 (E1) connections to PSTN trunks over two spans (digital circuits). Built exclusively for use with Asterisk and Asterisk-based communications systems, the dual span cards provide the best value in digital connectivity.
Dual span digital cards are available in PCI and PCI-Express form factors. All dual span models can be combined with an Octasic DSP-based echo cancellation module from Digium to provide effective hardware echo cancellation across all channels.
The GXP1625 is Grandstream’s standard IP phone for small businesses. This Linux-based model features 2 lines, 3 XML programable soft keys, HD audio and 3-way conferencing. A 132x48 LCD screen creates a clear display for easy viewing. Both the GXP1620 and GXP1625 include dual 10/100mb network ports and the GXP1625 includes integrated PoE. Additional features such as multi-language support, Electronic Hook Switch support for Plantronics headsets and call-waiting allow the GXP1620 and GXP1625 to be high quality, user-friendly and dependable IP phone.
• 2 dual-color line keys (with 2 SIP accounts and up to 2 call appearances), 3 XML programmable context-sensitive softkeys, 3-way conferencing, multi-language support • Automated personal information service (e.g., local weather, etc.), personalized music ring tone/ring back tone, customizable screen content/format using XML, advanced web and enterprise applications integration (pending) • Dual-switched 10/100 Mbps ports, integrated PoE on GXP1625 • Use with Grandstream’s UCM6100 series IP PBX appliance for Zero-Config provisioning, 1-touch call recording and more • HD wideband audio, superb dull-duplex hands-free speaker-phone with advanced acoustic echo cancellation and excellent double-talk performance •Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control • 132 x 48 pixel backlit graphical LCD display • Large phonebook (up to 500 contacts) and call history (up to 200 records)
Panasonic KX-A450 CE Repeater DECT Panasonic
The KX-A406 DECT repeater is an ideal solution for when you need to extend the range of your DECT CS (Cell Station), to cover areas where reception was previously not available.
The A406 DECT repeater extends the range in all directions, allowing for a wider area to be covered. A single cell station can register up to 6 A406 repeaters while each repeater can support 4 simultaneous transmissions channels.
The DECT repeater is:
The beroNet 1 PRI Gateway contains one PRI (S2M, E1) port. The port can be operated individually in NT (Network Termination) or TE (Terminal Equipment) mode. By adding a virtual CAPI a Fax Server can be also connected. The Gateway is compatible with SIP. A connection to common PBX Systems is possible via the ISDN interface.
Advantages » Connects SIP with ISDN » ISDN port can be switched in TE or NT mode » Expandable with a supplemantary module (PRI, BRI, FXO, FXS, GSM) » Via Cloud administrable » Cascadable using PCM » High quality Aluminium Housing » Virtual CAPI available
Grandstream GXP2170
The GXP2170 is a powerful enterprise-grade IP phone that is ideal for busy users who handle high call volumes. This top-of-the-line Enterprise IP Phone features up to 12 line keys/line appearances and 6 SIP accounts using 4.3 inch color display LCD and full HD audio.
The GXP2170 supports the fastest possible connection speeds with dual Gigabit network ports. It features integrated PoE and built-in Bluetooth for syncing with mobile devices and Bluetooth headsets, as well as the ability to connect/power up to 4 cascaded extension modules with LCD display to access up to 160 speed dial/BLF contact.
The GXP2170 also features up to 48 digital, on-screen speed dial/BLF keys. By adding advanced security protection and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms, the GXP2170 is the perfect choice for enterprise users looking for a top-notch executive IP phone with advanced functionality and performance.
Some features of the GXP2170
Grandstream GAC2500
Grandstream GAC2500 is a conference phone, ideal for meeting rooms and offices. As it based on Android, it provides access to all Google Play Store applications such as Skype and Google Hangouts. Among others, their features include:
Expansion microphones
Possibility to connect a second unit GAC2500 in daisy- chain mode to act as expansion microphone and speaker side ( RJ48 port ) .
Elastix High Availability is the new addon released by Elastix this year, joining the Call Center Pro addon as professional solutions with licensing deployment.
Features of the addon are the following:
Cyberdata Intercomunicador VoIP para interiores (Montaje en pared)
Intercomunicador de Cyberdata VoIP para interiores; comunicación bidireccional y control de acceso seguro para su sistema telefónico VoIP. Estos dispoisitivos son perfectos para entornos tales como instalaciones comerciales / residenciales, escuelas y universidades, establecimientos de venta, plantas de fabricación, etc.
Características destacadas
GVC3200 Videoconferencing System - 2nd Best Product at VoIP2DAY 2015
GVC3200 Videoconferencing System is an innovative solution that offers an revolutionary and flexible system. GVC3200 can support multiple protocols and videoconferencing platforms.
It is based on Android 4.4 so It provides full access to all videoconferencing applications in Google Play Store; It also supports up to 9 participants between SIP and other protocols.
The GXP1630 IP Phone, versatile, affordable and Gigabit
The GXP1630 is a powefull IP Phone recommended for Small and medium Bussinesses (SMB)
The new GXP1630 is based on Linux OS and includes 3 lines, 8 BLF keys and 3-way conferencing . A 132x48 backlit LCD sreen creates a great display for easy customer experience
All these characteristics make the GXP1630 to be a versatile, high quuality, and dependable office phone.
Alphatech Video Door IP Phone BOLD T2C, 2 buttons
The IP BOLD SIP intercom combines timeless alluminium design, modern technology , easy installation and maintenance. The IP BOLD door stations, IP intercoms family, offer two relay contacts, PoE, full duplex audio, a wide angle colour camera and a optional numerical dialling keypad. IP BOLD offers two additional virtual relays, which you can use with IP web relays. A suitable web relay, an external IP relay. See settings guide here. Receive an email with a photo of the visitor when you have a missed incomming call You can use audio and video softphone apps for Apple (iOS), Windows or Android such as iBell, ZoiPer, Linphone or 3CX. The IP BOLD is a 3CX compatible doorphone. It is used for seamless operation with 3CX IP PBX and 3CX audio and video softphones. Recommended IP video phones:
Call up to 5 numbers / user accounts without using or programming any SIP server (IP PBX) We recommend using SIP based IP phone systems such as 3CX, Asterisk, Cisco Call Manager 10.x, Quadro Epygi QX IP PBX Each user can have up to 5 different numbers assigned, thus never missing an incomming call By pressing a call button, all your soft video phones or fixed IP video phones can start ringing no SIP server or IP PBX required for call group ringing your iPad/iPhone, Android smartphone/tablet, Windows PC and fixed HW IP audio/video phone can be ringing simultaneously or consecutively as you need Just answer the audio/video call by one of the soft audio/video phones or fixed HW IP phones and start talking to the visitor at the door entrance Apple iOS, Android UDV video softphone apps and Windows PC video softphone "iBell SW" available Available models:
Alphatech Video Door IP Phone BOLD T1C, 1 button
Call up to 5 numbers / user accounts without using or programming any SIP server (IP PBX) We recommend using SIP based IP phone systems such as 3CX, Asterisk, Cisco Call Manager 10.x, Quadro Epygi QX IP PBX Each user can have up to 5 different numbers assigned, thus never missing an incomming call by pressing a call button, all your soft video phones or fixed IP video phones can start ringing no SIP server or IP PBX required for call group ringing your iPad/iPhone, Android smartphone/tablet, Windows PC and fixed HW IP audio/video phone can
be ringing simultaneously or consecutively as you need just answer the audio/video call by one of the soft audio/video phones or fixed HW IP phones and start talking to the visitor at the door entrance Apple iOS, Android UDV video softphone apps and Windows PC video softphone "iBell SW" available Available models:
Alphatech Rain Hood Ref. 213206
A little aluminium rain hood for door phones Alphatech (*).
*For surface roof
The GXP1620/1625 is Grandstream’s standard IP phone for small businesses. This Linux-based model features 2 lines, 3 XML programable soft keys, HD audio and 3-way conferencing. A 132x48 LCD screen creates a clear display for easy viewing. Both the GXP1620 and GXP1625 include dual 10/100mb network ports and the GXP1625 includes integrated PoE. Additional features such as multi-language support, Electronic Hook Switch support for Plantronics headsets and call-waiting allow the GXP1620 and GXP1625 to be high quality, user-friendly and dependable IP phone.
Grandstream GXP-1405 (PoE)
GXP1400/1405 is a next generation small-to-medium business IP phone that features 2 lines with 1 SIP account, a 128x40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE (GXP1405 only), and 3-way conference. The GXP1400/1405 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP phone with affordable cost.
GXP1400/1405 Product Brochure
User Manuals
GXP1400 User Manual
Quick Installation Guides
GXP1400 Quick Installation Guide
White Papers
XML Application Guide
XML Based Customizable Screen
XML Based Downloadable Phonebook
XML Survey Application on GXP2020
Interoperability
Configuring GXP Series with Asterisk
Configuring GXP Series with 3COM
Configuring GXP Series on Quadro LAN
Configuring GXP Series with Broadworks
Broadsoft Provisioning Templates
Configuring GXP Series with 3CX
Compatible Headsets
Datasheets
Power Consumption Datasheet
Firmware
Firmware Download
Tools
Grandstream IP Phone Custom Ringtones Generation Tool
Tools, Utilities, and Configuration Templates
Grandstream Device IP Discovery Tool
Resources
GXP-Series Phone Labels
Snom M25 DECT
El teléfono Snom M25 DECT es un aparato de alta calidad, con una excelente relación calidad precio.
Snom M25 utiliza pilas estándar AAA recargables fáciles de cambiar, que proporcionan hasta 75 horas de funcionamiento en espera y hasta 7 horas de llamadas. El Software de usuario de snom está optimizado para una navegación rápida e intuitiva.
Se puede utilizar tanto con soluciones DECT Snom Multicelda M700, como en la base monocelda M300.
Teléfono IP Snom 715 Simplemente Funcional
El snom 715 es un teléfono de sobremesa completo e intuitivo a prueba de futuro: estudiado para un uso intensivo, dispone de conmutador Gigabit y puerto USB para conexiones adicionales como redes WLAN.
El snom 715, con su diseño cómodo y elegante, ofrece una mayor conectividad así como una perfecta facilidad de uso y elevadas prestaciones a un precio competitivo, además de una completa gama de funciones telefónicas para la vida laboral cotidiana.
La mejor conectividad, la gran variedad de funciones y el elevado rendimiento con facilidad de uso, convierten al snom 715 en el teléfono de sobremesa ideal para cualquier compañía en busca de calidad y adaptabilidad a un precio altamente competitivo.
BeroNet offers the berofix card series, a higlhly flexible and powerful hardware solution to integrate ISDN, (BRI/PRI) , analoge (FXO/FXS) and GSM Lines to any SIP based VoIP system. berofix is not a typical SIP Media Gateway neither a standard PCI/PCIe card, where you need proprietary drivers, so we call it a "Gateway Card". Because of the special Hardware design the operating system is detecting berofix as a classical Network card. All necessary drives to work with the berofix should in general automatically loaded by the OS. Thus berofix is OS indepedent and can be used for instance on Linux/Unix Windows and MAC environments (currently tested under Linux and Windows). Berofix has a modular concept and supports the following hardware DSP based Voice Proccessing
Features:
- G.168/G.165 Echo cancellation with echo path change detection, up to 128ms
- Codec translation: G.723.1 and Annex A, G.729a, G.726, G.711u/a
- DTMF detection and generation
- T.38 fax relay (V.27,V.29 and V.17)
- SIP over TCP with SRTP and TLS (available Q2 2011)
- DSS1, EuroISDN conform
- Q.SIG Basic-Features
- PCM Bus interconnection between berofix cards to enable hardware bridging for transparent voice,data and Fax transmission via optional PCM-Bus cable. (available Q4 2010)
- The berofix product series constists of a base board and modules (LineInterfaces) which can be plugged on the base board.
Each base board can carry up to 2 line interfaces. The berofix baseboard is available as PCI / PCIe or as external box with the following channel densities:
berofix 400 (4-16 channels)
berofix 1600 (16-64 channels)
berofix 6400 (64-120 channels)
These baseboards can be equipped with the following available Lineinterfaces:
bf4S0, 4 Port BRI module
bf1E1, 1 Port PRI module
bf2E1, 2 Port PRI module
bf2S02FXS, 2 Port BRI and 2 Port FXS analog module (available Q4 2010)
bf4FXO, 4 Port FXO analog module (available Q4 2010)
bf4FXS, 4 Port FXS analog module (available Q4 2010)
bf2GSM, 2 Port GSM module (available Q1 2011)
In addition to the baseboards and the line interface, the following accessories are available:
- bf4Bridge (to use all 4 RJ45 slots on a baseboard with one bf4S0, bf2S02FXS, bf4FXO, bf4FXS)
- bnTAdapters
- bnPCM Cable (to interconnect berofix baseboards)
- bnE1Crosscable (to connect berofix E1 ports to other systems)
- bn19Bracket 19" Rackmount bracket for berofix boxes
Due to the modular concept of the berofix you can free mix PRI / BRI, analog (FXS/FXO) and GSM Lineinterfaces on one baseboard.
beroNet Small Business Line Gateway: 2S0. BFSB2S0 (2 BRI)
The beroNet Small Business Line is designed for the SOHO area, to connect ISDN(BRI) and Analog (FXO/FXS) Lines to any SoftPBX’s like 3CX, Asterisk, FreePBX, Elastix and many more. In comparison to the beroNet VoIP Gateways, the Small Business
Line is not modular and not rack mountable.
BFSB2S0 (2 BRI) Specifications:
*********** ATTENTION *********** SmallBusiness Line is not modular and not rack mountable.
Telefone IP Aastra 6869i
O telefone 6869i pertence à série 6800i de produtos Aastra SIP. Eles apresentam um design moderno e elegante, áudio de banda larga em HD, função viva-voz e um processador de áudio ideal para conversas claras.
Adaptador Plantronics para APD-80 para Grandstream
QXFXS24 Gateway
The ability to use a large existing analog telephone base with a new IP telephony network is a very important requirement for many companies who are adapting to the Digital Age. The QXFXS24 Gateway is the perfect solution for your business because it can be added to an IP network, therefore allowing your existing analog phones to join the new VoIP network. The QXFXS24 can be stacked for additional capacity. The QXFXS24 provides a number of powerful features not found on standard FXS gateways, including a detailed call routing table with digit manipulation options.
This Gateway can be installed with any SIP-compliant IP PBX on the market or plug-and-play with an Epygi IP PBX. The modular approach of Epygi Gateways allows you to retain an investment from the past analog phones, while adopting the latest technology with ease.
snom Vision - the expansion module for the snom 8XX & snom 7XX VoIP telephones.
Expands the functional capability of your next generation IP phone to a whole new level. You now have the ability to get more out of your favourite phone when used with this brand new offering from snom. The 16 programmable keys can be used for speed dial.
The snom Vision can be powered by the conventional power supply (5V adapter included in delivery). Alternatively, the snom Vision can also be powered through Power over Ethernet (PoE IEEE 802.3af).
In addition to being used as an expansion module with a snom 8xx IP phone, the snom Vision can also be used as a stand-alone device; thanks to its independent Linux OS that connects to your network via Ethernet. The snom Vision has a built-in web server that allows you to configure its features through an elegant web interface at your convenience. Moreover, the configuration settings can also be provisioned.
The design of this brand new product is in line with the outstanding features of the snom 8xx that our discerning customers have come to expect. The snom Vision and the next-generation snom 8xx IP phone are indeed an exquisite match that combines functionality with grace.
Main features
Epygi QX DC power cable
Cable to offer Redundant Power solution between two modules Epygi QX. With this cable, a module QX can offer power to another module, in case one of them loses AC power. It's clever way to provide redundant power at combine two or more modules Epygi QX.
Rack Mount Kit
Epygi's Rack Mount Kit is designed with you in mind. The smaller, more modular QX products integrate seamlessly with the Rack Mount Kit, ultimately reducing clutter and maximizing space. With the exception of the 19-inch QX2000, Epygi's line of QX products simply slide into the rack mount and are secured with a thumb screw.
The kit, purchased separately from the IP PBXs and Gateways, includes two DC power cables for power redundancy. Power redundancy can be used on all QX products with the exception of the QXFXS24.
Please refer to the technical data sheets for our IP PBXs to see the maximum number of Gateways that can be interconnected.
Epygi QXISDN4 gateway (4 BRI)
The Epygi QXISDN4 Gateway performs similarly to the FXO version but connects using ISDN ports. This Gateway includes four ISDN BRI connections for phones and analog devices that can connect to the telephone company's central office or to a local PBX. The QXISDN4 can be used to add inbound lines and balance outbound call volumes from a combination of analog devices and IP phones. The lines can be centrally located within the office or used like an "extension cord" to remote branches. Each QXISDN4 is a stand-alone, SIP gateway device that includes a VPN-router, firewall, HTTP server and call processing software. Installation requires very little configuration. For example, a QX IP PBX will automatically present the new ISDN ports within its management system.
Plantronics' legendary CS family is setting a new wireless standard for desk phone communication with the CS500™ Series, which features the lightest DECT™ headset on the market, a new streamlined design and improved performance. Go mobile and multi-task up to 350 feet (100 metres) from your desk with answer, end and mute controls at your fingertips. And no matter what your preference, with three wearing choices there’s one to match your personal style. Enjoy the sleek contemporary design of the new system, premium wideband audio quality and wireless mobility, all with the same reliability for hands-free productivity that has made the CS family a bestseller for nearly a decade. With the new CS500 Series we are raising the bar once again for desk phone headset systems.
Superior call management
snom 720 IP phone: Pure functionality
The snom 720 phone addresses office users that require excellent audio and a large number of programmable PBX-style keys. It combines a state-of-the-art hardware with the proven snom SIP software.
All in all, the snom 720 raises the bar for VoIP phones in its class both in terms of voice quality, available features and day-to-day usability.
snom 760 IP phone: High-level functionality coupled with a multitude of professional features
The snom 760 phone addresses office users that require excellent audio, PBX-style keys, and rich visual information. It combines a state-of-the-art hardware with the proven snom SIP software.
The snom 760 not only provides you with comprehensive IP telephone functionality, but also a whole range of extra features which really put it in a class of its own compared to similar products.
La GXV3611IR_HD es un cámara IP infrarroja (IR) fija tipo domo para interiores, con una lente de 2.8 mm de alta definición – haciéndola ideal para la monitorización con ángulo amplio de personas cercanas en entornos como bancos, hoteles, comercio minorista, oficinas o entradas de edificios.
Su avanzado Procesador y Sensor de Imágenes (ISP) es accionado por un algoritmo para control de auto-exposición/balance de blancos que hace posible un desempeño excepcional en todas las condiciones de iluminación.
Conecte la GXV3611IR_HD con la Videograbadora de Red (NVR) GVR3550 de Grandstream, la cual soporta la tecnología Plug-and-Play con todas las cámaras IP Grandstream, para crear una poderosa solución de grabación y monitorización. La GXV3611IR_HD también puede ser manejada con GSURF Pro (software gratuito de gestión de video de Grandstream que controla hasta 72 cámaras simultáneamente) junto con otros sistemas de gestión de video compatibles con ONVIF.
La cámara IP GXV3611IR_HD ofrece una solución SIP/VoIP líder en la industria para transmisión de dos vías de audio y video tanto a videoteléfonos, como a teléfonos inteligentes. Contiene PoE integrado, IR-CUT para modo de día y noche, micrófono, altavoz y un HTTP API flexible para una fácil integración con otros sistemas de vigilancia.
M325 DECT Unicelullar Cell
The M325 is composed of M300 and M25 phone solution; a single-cell DECT powerful package that supports up to 20 phones and has a range of up to 300 meters outdoors and 50 meters indoors. In addition, the base has a wide coverage can be expanded by adding up to 3 repeaters Snom M5.
M300 features
M25 features
With the additional expansion microphones, the CP860 is the ideal conference phone for medium to large size conference room that need extended coverage at the far ends of a conference table. The expansion microphones extend the range of the CP860 by an additional 10 feet respectively.
Specification
Microphonefeatures
Physical features
Package features
A série T2 de telefones IP Yealink representa a próxima geração de telefones VoIP projetados especificamente para usuários corporativos que precisam de recursos de telefonia importantes, uma interface de usuário amigável e excelente qualidade de voz.
A série T2 oferece alta qualidade e definição de voz através de fones de ouvido HD, alto-falantes de alta definição e o codec HD (G.722). Um grande display gráfico de alta resoluçao, combinado com um máximo de 48 teclas, garante uma excelente experiência do usuário em termos de opções de configuração.
Além disso, para garantir a confidencialidade dos dados de áudio, também suporta os padroes de segurança de criptografia TLS, SRTP, HTTPS, 802.1X, Open VPN e AES. Esses protocolos protegem contra espionagem eletrônica e roubo de dados.
A série T4 da Yealink é destinada a usuários com altas expectativas em telefones IP. Ele foi projetado especificamente para pessoas que têm grande satisfação em experimentar um excelente serviço.
Revolucionária em sua aparência e design técnico avançado, a série T4 não é apenas atraente e confortável de usar, mas também oferece telas extra grandes de até 4.3 polegadas e 480x272 pixels de resolução, o que torna o lápis e papel desnecessário em o ambiente de escritório ocupado. De fato, a série T4 representa a vanguarda da tecnologia VoIP contemporânea em ação.
Recursos avançados que incluem compatibilidade com redes Gigabit e Bluetooth USB para facilitar o uso do fone de ouvido Bluetooth. Melhor qualidade de som que é fornecida pelo sistema de voz de alta definição (HD) e que está em conformidade com os padrões de certificação TIA 920. Na série T4, a Yealink alcançou a combinação ideal de perfeição e desempenho.
Além disso, recentemente, a série T4S foi adicionada a uma linha aprimorada projetada para executivos e gerentes que oferece qualidade de áudio de alta definição, integrando recursos de ponta, como Wi-Fi, conectividade Bluetooth e o codec OPUS.
A Yealink oferece uma variedade de telefones DECT projetados especificamente para pequenas e médias empresas; terminais sem fio que permitem a construção de sistemas de comunicações móveis baseados em custos de economia de SIP escaláveis.
O Yealink W52P é o resultado da busca eterna da Yealink pela tecnologia VoIP de última geração, que oferece qualidade suprema sob os mais rigorosos testes de qualidade e o mais alto nível de operação e simplicidade.
Este terminal não só satisfará todas as suas necessidades de telefone sem fio; Vai tornar as suas chamadas diárias uma experiência de participação muito agradável. Recentemente Yealink, também incorporou no mercado o W56P, uma versão melhorada do seu terminal DECT que incorpora algumas novas características, como maior tamanho de tela colorida, seu conector de 3,5 mm ou o design atraente de sua caixa.
Enterprise HD IP Phone
SIP-T29G
SIP-T29G IP Phone is the most advanced model in the Yealink T2x IP terminal series. It has a high-resolution TFT color display,delivers a rich visual experience.
Yealink Optima HD technology enables rich, clear, life-like voice communications.Supports Gigabit Ethernet, a variety of device connections, including EHS headset and USB. With programmable keys, the IP Phone supports vast productivity enhancing features.
Audio Features
Phone Features
Directory
IP-PBX Features
Display and Indicator
Feature keys
Interface
Other Physical Features
Management
Network and Security
Package Features
Entry-level IP Phone
SIP-T21P
Yealink’s new SIP-T21P takes entry-level IP phones to a level never achieved before. Making full-use of high-quality materials, plus an extra-large 132 x 64-pixel graphical LCD showing a clear 5-line data display, it offers a smoother user experience, much more visual information at a glance, plus HD Voice characteristics.
Dual 10/100 Mbps network ports. The T21P supports two VoIP account, simple, flexible and secure installation options, plus support for IPv6, Open VPN and a redundancy server. It also operates with SRTP/ HTTPS/ TLS, 802.1x. As a very cost-effective and powerful IP solution, the T21P maximizing productivity in both small and large office environments.
Power Supply Video Phone Yealink (VP530)
Power Supply Yealink Phones (T29,T46,T48)
Power Supply Yealink Phones (T20,T22,T26,T27,T28,T41,T42)
PSU Yealink 5V,600mA(T21,T19,W52P,T23)
SIP-T23G
Yealink SIP-T23G features intuitive user interface and enhanced functionality which make it easy for people to interact and maximize productivity.
Yealink HD technology enables rich, clear, life-like voice communications, outsourced management options, flexible deployment and third-party communications applications. As a cost effective IP solution, it helps users to streamline business processes, delivery a powerful, security and consistent communication experience for small and large offices environment
Sangoma SBC for small business - 30 calls
The SBC helps to reduce infrastructure costs versus analog networks; It also allows protect you from malicious attacks as an intermediary in their IP infrastructure. It's algo interoperable with leading IP PBX allowing more flexibility in day to day in the company.
Electronic Hook Switch- communicates electronically with the phone, eliminating the need for a HL10 lifter
Snom Expansion module D3
D3 expansion module provides extra functionalities to your Snom; in addition to expanding the number of keys it allows you to add a lot of extensions and calls thanks to its 18 programmable keys.
You can connect up to 3 modules and add up to 54 function keys; besides the connection is very simple thanks to its USB cable (connected directly to your main phone). If you want to extend up to 3 modules you can use the last USB to connect other peripheral devices such as headphones or wireless.
Compatible with...
_____
* When two of three modules are used with one phone, a separate power supply needs to be connected to the second module.
Alcatel D135
Telefone Alcatel D135 com mini base, ideal para instalar em qualquer lugar.
Entre suas vantagens, destacamos sua tela alfanumérica retroiluminada que garante um maior conforto visual.
O calendário com capacidade para 20 nomes e números permite que você chame seus contatos preferidos da maneira mais simples e o registro de chamadas recebidas permite que você controle as chamadas receidas.
Recursos em destaque
Gateway ATA HT802 (2FXS + 1THS)
The Grandstream HT802 is an analog phone adapter (ATA) that has two FXS ports and an Ethernet port. This device is ideal for residential and office environments that are looking for an easy-to-manage and quality IP telephony solution.
Grandstream GXP1615 Easy to use, ideal for small and medium business
Grandstream GXP1615 is an IP phone ideal for small and medium business and it's similar to GXP1610. GXP1615 provides a single SIP account, LCD display, call waiting option and 3-way conference; furthermore, GXP1615 also offers integrated PoE to power the device and provide a network connection.
An IP phone very easy to use with high-quality. These are its main features:
Cyberdata Talk Back - Modelo 011398 (Signal White)
Cyberdata's new SIP Talk-Back Speaker enables two-way conversations in settings such as classrooms, offices, medical facilities and clinics. By use of remote call button (sold separately, Part 011185).
During the active calls, the LED light on the switch can be programmed to blink to show call activity. Alternatively, a call can be placed to the speaker to initiate either a page or two-way conversation.
Cyberdata Talk Back - Modelo 011397 (Gray White)
Cyberdata VoIP ceiling speaker (signal white)
The VoIP Ceiling speaker from Cyberdata is an IP device powered by Power over Ethernet (PoE 802.3af) technology that supports most SIP switches on the market. In addition, thanks to its small size and weight make it a very easy device to mount anywhere.
In particular, it has a volume control through the network, supports multicast and also has two kits, the "Cyberdata Wall Mount" and the "Cyberdata Wall Mount Clock Kit, including the latter a clock in the housing of integration in the wall.
In addition, the Cyberdata VoIP loudspeaker can be configured as a "Night Ringer" with two SIP extensions (an extension can be assigned to a paging group with autoresponder). The second extension is a night call group with the rule "First (or if the caller hangs up) the IP loudspeaker stops ringing. If the caller's IP phone number is dialed, the caller's IP address will be dialed.
Cyberdata SIP Speaker - 011393
The Cyberdata SIP Speaker is a power-over-ethernet (PoE 802.3af/802.3at) and Voice-over-IP (VoIP) public address loudspeaker that easily connects into existing local area networks with a single CAT5 cable connection. The speaker is compatible with most SIP-based IP PBX. In a non-SIP environment, the speaker is capable of playing audio from a multicast source. Its small footsprint and low height allows the speaker to be discreetly mounted almost anywhere.
New features
Konftel EGO Your personal conferencyng system
Your Konftel Ego is a conference system ideal for meetings wherever you are thanks to its compact size and portability. Furthermore, despite its small size, it provides crystal clear sound thanks to its audio technology OmniSound. Ideal for use with other tools and conferencing applications such as Skype Business, Cisco Jabber or Avaya Communicator, among others, and to play music.
Multi purpose switch ADD-818
Yealink W56H | Handset
Yealink DECT W56H is presented as the new generation of hands-free phones of Yealink. It's ideal for business thanks to its compatibility and the features offered. It also presents an excellent lithium battery, audio quality, a color display with 2.4'' and backlit LCD offering an excellent design and style.
Grandstream Terminal DECT - DP720
Grandstream DP720 is a DECT phone that allows freedom of movement of its VoIP network. Ideal for any business, large areas or residential environments that require advanced features, typical of a business phone and wireless autonomy.
Each handset DP720 allows up to 10 SIP accounts and provides a range of up to 300m outdoors and 50m indoors thanks to its base station DP750, which can be added up to 5 handset DP720.
Professional IP Phone - Snom D375
The Snom VoIP phone D375 is the new generation of professionals desk phone. It has a wide TFT color screen with 4.3 inch, 12 programmable keys bicolored function (BLF), 2 Gigabit Ethernet ports for connecting various network devices and Bluetooth compatibility built to connect wireless headsets and Wi-Fi antennas.
It also presents a design for easier dialing with 12 SIP identities and 10 function keys.
Alcatel Conference 1500
Alcatel equipment Conference 1500 is the best choice for meetings in small groups (6-8 participants) thanks to full duplex technology and its sound quality. And thanks to its removable microphones DECT communication is much more fluid thanks to its sound quality.
Video Conferencing Solution GVC3202
Grandstream GVC3202 is a ground-breaking solution with high felixibility and ability for support multiple protocols and platforms. Furthermore, GVC3202 is easy to install and can also support IPVideoTalk platform, has plug and play and can be interoperable with a third videoconferencing platform.
Compared with GVC3200, the GVC3202 is a small-scale solution that fits the needs of small and medium business.
Characteristics of GVC3202
IP Phone Grandstream GXP2135
With LCD TFT and color display, 4-way conferencing, PoE, Gigabit and Bluetooth integrated, Grandstream GXP2135 is a powerful option in the range of executive phones of Grandstream.
In addiction, this model also incorporates 8 line keys BLF and LED, XML programmable keys and 4 menu keys, ideal for users with a large demand.
It's ideal for...small and medium business that are looking for a telephony solution with more advanced features and have a more intense working pace.
GXP2135: Some features
Cyberdata Intercomunicador para interior
Cyberdata has an extensive line of Indoor and Outdoor SIP-enables Intercoms. This model (011211) delivers two-way communication and secure access control for your VoIP phone system. These devices are perfect for settings such as commercial / residential facilities, schools, and universities, retail establishments, warehouse and manufacturing plants, SMB, parking garages and shipyards and so much more.
Snom M300 DECT for M25 or M65
Snom Base Station M300 can be extended up to 20 additional Snom DECT handsets with a range or up to 300 meters outdoors and aproximately 50 meters indoors. In addition, this coverage can be extended by adding up to three M5 repeaters. . Una cobertura que, además, se puede ampliar aun más con hasta 3 repetidores Snom M5.
The solution is interoperable with the full spectrum of major IP PBX systems and, furthermore, it doesn't need extra licenses for additional codecs, features, audio channels, etc. resulting in no unexpected additional costs or surprises.
Alcatel Conference Solution IP1850 | SIP version of Alcatel Conference 1800
Alcatel audioconference solution, IP1850, is the ideal system por medium-sized meeting rooms to benefit high quality sound and carry out natural and fluid conversation thanks to its Full Dúplex technology that make it much more fluid communication with its simultaneous bidirectional transmission.
Furthermore, thanks to its removable DECT phones can be distributed making the room more comfortable experience and expanding the conference audio coverage depending on the needs of each occasion.
This is the SIP version of the model IP1800 (analog). We show you its main characteristics:
Konftel 300Wx
Com o sistema de conferência sem fio Konftel 300Wx, você pode realizar reuniões onde quiser sem ter que pensar em tomadas de rede ou tomadas graças à sua tecnologia DECT.
Para isso, você pode selecionar uma estação base compatível com o ambiente de telefonia de sua empresa (SIP ou analógico) ou conectar-se a um sistema DECT já instalado.
Além disso, sua bateria recarregável oferece até 60 horas de conversação. Esta estação base pode ser combinada com o Konftel IP DECT 10 ou com a Base Analógica DECT da Konftel para conexão analógica.
Outra vantagem de trabalhar com esses dispositivos é que eles podem se conectar ao seu aplicativo móvel Konftel Unite. Por meio do acessório Adaptador do Konftel Unite, você pode conectar o aplicativo para gerenciar sem esforço o sistema de conferência.
Konftel DECT IP 10 - Base Station for Konftel 300Wxs
This Konftel DECT base station can join the conference phone Konftel 300Wxs and allows wireless connection with HD sound (more natural than others conventional phones). It also allows up to 20 Konftel 300Wxs with HD capacity and up to 5 simultaneous calls.
Alcatel Temporis IP300
This Temporis IP300 is a versatile Ip desktop that offers multiple benefits. Its integrated DECT base allows to associate a DECT phone and a wireless headset, providing a scalable solution to the needs of everyday life in the company.
It also offers other qualities such as:
Yealink EXP20 Keyboard
The Yealink EXP20 keyboard is a powerful add-on that enhances the functionality of the Yealink IP telephone: T27G, T27P and T29G. Thanks to its graphical display with 160x320px and its 20 dual-color LED keys, it allows you to track all incoming calls.
A tool that makes easier the user experience, simplifying and optimizing your work time. Up to six EXP20 phone systems can be strung together on a single daisy chain. Ideal for receptionists, assistants, clerks or workers who need to monitor and manage a large volume of calls on a regular basis.
The SBA App beroNet (Survival Branch Appliance) is the solution if you lose connection with an off-site telephone system when local telephones are unable to communicate with each other or the outside world. This SBA App, transforms a beroNet VoIP Gateway or card into a survival branch appliance, enabling the beroNet device to function as a local backup telephone system. It offers functions such as:
Furthermore, It's easy to install and configure and can accomodate up to 20 users in one office.
AC100-240V input, 5Vdc/1000mA output
Grandstream DP750 - Base Station
This base station for cordless phones Grandstream is the ideal solution for businesses that need a wireless converage on several floors or inside large buildings solution. They can be added up to 5 DP720 providing the ideal wireless solution for business.
It offers up to 300m outdoors and 50m indoors, 3-say conferencing, full HD and PoE. It also supports a variety of methods and auto-provisioning TLS/SRTP/HTTPS.
Yealink W56P | Handset + Base Station
Yealink W56P is presented as the new SIP wireless system that combines quality, flexibility and confidence. It offers all the features of a handset but without losing its qualities SIP providing greater comfort, autonomy and freedom of movement. In addicion, it is ideal for business thanks to its compatibility and has an excellent performance. It's a VoIP phone ideal for small and medium enterprises solution.
Snom D745
El teléfono IP de sobremesa D745 tiene un diseño muy atractivo a la vez que funcional gracias a sus prestaciones; es por ello, que como herramienta de trabajo es clave para el día a día en la empresa gracias a sus completas prestaciones como: segunda pantalla con teclas LED programables multicolores y posibilidad de ampliación de hasta 3 módulos de expansión Snom D7.
Además, el Snom D745 presenta un puerto USB, Gigabit y VPN.
Características principales
Snom D345 with a second screen
The D345 is an IP Phone with 12 SIP identities, high quality and very elegant terminal. It also provides a second screen with 4 virtual pages, each of which has 12 programmable function keys with LED multicolored, allowing a variety of features and contacts.
Teléfono Sobremesa IP Snom D315
The D315 desktop IP phone is a professional terminal with high resolution screen, 4 identities, Gigabit Switch and USB support. In addition, you can incorporate a variety of accessories such as D3 module or USB WiFI stick support. Furthermore, thanks to its Gigabit switch you can connect to a Gigabit Ethernet LAN. A safe bet if you are looking for a rugged and versatile terminal.
Snom D305 - Desk IP Phone
Snom D305 is a professional phone with a high-resolution display and excellent cost-performance ratio. Furthermore, it has an audio architecture that has been designed and perfected, 4 SIP identities and high-resolution backlit display for excellent viewing screen.
In addition, it is a very ergonomic model with a slight tilt more comfortable for users.
Spectralink DECT 7622
O telefone Spectralink 7622 é altamente durável, projetado especificamente para ambientes de fabricação e incorpora uma gama de recursos, como botão de alarme, tela colorida e ícones de alarme específicos de fabricação.
Suporte de parede Yealink T27P e T29G
New Video Phone - Yealink SIP T49G
The new videophone Yealink SIP VP-T49G HD incorporates new features that provide a perfect balance between simplicity and sophistication to executives and employees. A powerful tool to make more productive your jorney (wherever you are).
Sangoma Netborder Carrier SBC 500 calls
Sangoma Netborder Carrier SBC offers network security and greater flexibility adding devices, protocols or VoIP Networks through VoIP islands and acting as gateway. Interoperability and protection against internal and external threats are the main highlight of this product.
Some of its features:
The Spectralink DECT Repeater is a building block used to extend the wireless coverage area within a Spectralink DECT network. The DECT Repeater provides a larger geographical spread of the traffic channels and increases the coverage area. But because it does not increase the number of traffic channels available, the DECT Repeater is mainly used in areas with limited traffic.
Every base station or IP-DECT 400 can be expanded up to 3 repeaters, in star or chain
To work fine, repeaters must be numerated accordind the rules defined by Spectralink. Numeration and upgrade of repeaters is made with an adittional kit programming sold separately.
The Vega Enterprise VM SBC provides Security and Interoperability for Enterprise Networks. The Vega VM eSBC is possible to provision and manage thanks to the browser-based GUI. The straight-forward, session-based licensing model also makes the Vega VM eSBC one of the most cost-effective SBCs to deploy and maintain in the field.
The fact that the Vega VM eSBC can be implemented within an existing VM infrastructure further contributes to it value. It can be deployed into an existing infrastructure without any additional Power, Space or Cabling concerns or adding Points of Failure. The VM solution offers unmatched flexibility, redundancy and durability.
Protection from Enterprise Security Threats:
The Konftel 300Wx allows you to hold conference calls where and when it is convenient – without worrying about phone jacks and power outlets. The rechargeable lithiumion battery ensures superior performance with up to 60 hours of talk time, this allows a full working week without worrying about recharging!
The Konftel 300Wx supports DECT with GAP/CAT-Iq standard for less interference and clearer conversations and moreover is easily integrated with existing DECT systems for complete coverage.
The Konftel 300Wx can also be easily connected to your cell phone or computer via USB for VoIP calls over the internet. Its embedded line mode enables you to connect DECT, cell phones and USB simultaneously for multi-party calls.
The Konftel 300Wx has several smart features to make your meetings as easy and efficient as possible. Record calls on a SD memory card, listen to it later or share it with others. The built-in conference guide helps you dial multi-party calls, store call groups and initiate reoccurring meetings. The Konftel 300Wx is expandable with microphones, equipped with Konftel’s patented OmniSound audio technology for crystal clear sound and its elegant Scandinavian design makes it a welcome addition to any conference room.
The Vega 400 VoIP gateway connects digital telephony equipment to IP networks. All Vega 400 gateways are supplied with four E1/T1 interfaces, regardless of the license purchased.
The unit is purchased pre-licensed to suit the initial requirements of the customer for the quantity of concurrent VoIP calls desired through to 120 VoIP channels. Future expansion is easily achieved in the field & can be provisioned by means of further licenses and expansion modules.
Each E1/T1 interface can be independently configured as network side or terminal side. The Vega 400 gateway can therefore be connected to a PBX & the PSTN simultaneously. This configuration provides:
Integrated Bypass Relays For Resiliency The Vega 400 gateway incorporates an additional four RJ45 sockets and fails over to these during outages. This resource can be utilised to achieve hardwired connection from the PBX to the PSTN for instances when the Vega is installed between the two. Or alternatively to failover to a back-up Vega 400 & thereby providing dual redundancy.
The ultra professional corded phone. You've liked Temporis 700, you'll love Temporis 780. Its pure and contemporary line magnifies the phone design. You'll appreciate its large, legible and ultra comfortable display. You'll also be seduced by the 10 lockable direct memory keys. Moreover, if you're looking for more confientiality, Temporis 780 offers the possibility to erase on a daily regular basis the redial log memory. Boost your company productivity thanks to the large capacity and maximised directory. It can also be equipped with miscellaneous headset as it is compliant with different headset brands.
Alcatel Temporis 580 (white) GREAT features, GREAT comfort
Temporis 580, the new born of Alcatel Temporis range, is fitted with a 2 line display and handsfree feature.
You'll love its multiple assets:
- The handsfree feature to organize your conference calls.
- The display to know who is calling you and live an even more user-friendliness experience.
- The freedom and the audio quality of your communications in headset mode.
Its incredible features and performance will continuously amaze you.
Alcatel Temporis 580 GREAT features, GREAT comfort
Temporis 380 the new generation.
You'll be seduced by its state-to-the- art and compact design for a minimal footprint.
This easy-to-use phone is fitted with high professional features such as :
Alcatel Temporis 180
The professional telephone that keeps things simple
A professional telephone that is ultra-simple to use.
The dedicated secrecy key is very handy for quickly cutting off the microphone when you need to have a private word with someone else.
Adjust the volume easily with the + and - keys.
The practical new message indicator light tells you at a glance if you have received messages* in your absence.
*Subject to availability of PABX functionality or service provider subscription
Upgrade para los gateway de Sangoma Vega 400. Con este upgrade su gateway Sangoma Vega 400 podrá licenciar 30 canales adicionales.
Business HD IP DECT Phone
W52P
Yealink W52P is a SIP Cordless Phone System designed for small business and SoHo who are looking for immediate cost saving but scalable SIP-based mobile communications system.
Combining the benefits of wireless communication with rich business features of Voice over IP telephony, User can benefit from freedom of movement, lifelike voice communications, multi-tasking convenience, professional features like intercom, transfer, call forward, 3-way conferencing, PoE etc.
This system works with widely-known Broadsoft, Asterisk, 3CX and supports quick and easy configuration.
Personalization
Voice and Codecs Features
Networks Features
Security
Connectors
Physicalk Features
Business HD IP DECT Phone (Additional Handset)
W52H
The Yealink SIP-W52H cordless units allow you to extend the numbers of users whilst providing increased flexibility and scalability when used in conjunction with the SIP-W52P handset and base station.
Sangoma Netborder Carrier SBC 250 calls
The Sangoma NetBorder SBC Carrier offers a wide range of features offered to its IP telephony network security and greater flexibility in implementating any device or network protocol. In this range of SBC stands out above all his remarkable e interoperability, with special care in converting SIP protocols, security against internal and external threats, connections and connection terminals SIP VoIP islands to form IP infrastructure.
Dual 10/100 Mbps network ports with integrated PoE are ideal for extended network use. The T21P supports two VoIP account, simple, flexible and secure installation options, plus support for IPv6, Open VPN and a redundancy server. It also operates with SRTP/ HTTPS/ TLS, 802.1x. As a very cost-effective and powerful IP solution, the T21P maximizing productivity in both small and large office environments.
Cyberdata Switch Ethernet 3 puertos Gigabit - Modelo 011236
The Cyberdata 3-Port Gigabit Ethernet Switch enables users of a PC to split a single Gigabit Ethernet port into two Gigabit Ethernet ports. The power for the switch is supplied by the PC.
Wireless Headset Adapter
EHS36
The new advanced Yealink Headset Adapter EHS36 provides the technical interface between Yealink SIP-T48G/T46G/T42G/T41P/T38G/T28P/T26P telephones and a compatible wireless headset.
It is approved for use with wireless models made by major manufactures, including Jabra, Plantronics and Sennheiser. The unit is easy to install via a simple link from the EHS36 to the EXT phone port. Its `plug-and-play’ mode gives you direct control of your Yealink phone, with the ability to answer and hang-up calls remotely.
It has been designed specifically to ensure maximum effectiveness in reception areas, call-centers and general telephone use. The EHS36 is ideal for corporate, financial, health, government, educational, industrial and SME/SoHo market sectors.
* Full compatibility with Jabra, Plantronics and Sennheiser * Phone control through a wireless headset * Plug-and-play, easy to use
Physical Features
GXV3674_FHD Outdoor Day/night HP IP camera
The GXV3674 series are powerful weatherproof Infrared (IR) IP cameras offering an adjustable vari-focal High-Definition lens making them ideal for any outdoor/indoor setting where weather and light conditions constantly change. Each camera’s variable-focal lens allows the user to adjust the lens to best fit their monitoring needs, which allows the GXV3674 to monitor nearby areas, such as a building entrance, and distant focuses, such as a parking lot.
Its’ advanced Image Sensor Processor (ISP) is powered with a state of the art auto-exposure/auto-white balance algorithm which allows for exceptional performance in all lighting conditions, including low light settings. The GXV3674 series can be managed with GSURF Pro (Grandstream’s FREE Video Management Software that allows simultaneous control of up to 36 cameras) along with other ONVIF complaint video management systems.
It features industry-leading SIP/VoIP for 2-way audio (using the camera’s Audio_In port) and video streaming to both video and mobile (smart) phones. The GXV3674 contains integrated PoE, IR-CUT for day and night mode, advanced security protection and flexible HTTP API for easy integration with other monitoring systems.
The GXV3662 series are outdoor weatherproof IP dome cameras that are ideal for outdoor use in areas where weather conditions are a concern. The GXV3662 series models are built with a vandal-proof and tamper-proof casing which allows them to stand up to possible tampering in settings such as banks and retail stores. Featuring a variable focal lens that can be manually adjusted from 3.3mm to 12mm, the GXV3662 series can be used for any indoor or outdoor monitoring situation – whether it be monitoring an area in the distance or areas in close-proximity to the camera. Like all Grandstream IP Video Surveillance cameras, the GXV3672 series is OnVIF compliant and features built-in PoE.
Digium A4B: Analog Telephony Cards 4-Port PCIexpress
The A4 Series of analog cards supports up to four (4) connections per card in your Asterisk system. Using Digium's single-port interface modules, A4 series cards can scale from one (1) to four (4) ports.
The modular nature of the cards allows you to mix and match between line (FXO) and station (FXS) interfaces, giving you the exact port configuration you need. Digium A4 Series analog cards are available in low profile, half-length PCI and PCI Express form factors.
Digium cards are compatible with all versions of Asterisk using the DAHDI driver framework. Asterisk and DAHDI are available for free from the Asterisk.org website. All cards include a five (5) year warranty.
(Optional accesory low profile bracket)
Digium A4A: Analog Telephony Cards 4-Port PCI
IP Phone Grandstream GXP2160
The GXP2160 is a state-of-the-art enterprise grade IP phone that features up to 6 lines, 4.3 inch TFT Color LCD, 5 XML programmable context-sensitive soft keys, dual Gigabit network ports, integrated PoE and Bluetooth, 5-way voice conferencing, and Electronic Hook Switch (EHS).
The GXP2160 delivers superior HD audio quality on the handset and speakerphone, rich and cutting edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.
Ideal for enterprises and SMBs, the GXP2160 is the perfect choice for users looking for a high quality, feature rich multi-line executive IP phone with advanced functionality and performance.
The best option!
IP Phone Grandstream GXP2140
The GXP2140 is a state-of-the-art enterprise grade IP phone that features up to 4 lines, 4.3 inch TFT Color LCD, 5 XML programmable context-sensitive soft keys, dual Gigabit network ports, integrated PoE, 5-way voice conferencing, and Electronic Hook Switch (EHS).
The GXP2140 delivers superior HD audio quality on the handset and speakerphone, rich and cutting edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.
The GXP2140 is compatible with the GXP2200 Extension Module, allowing for quick and easy access to up to 160 contacts at the touch of a finger. Ideal for enterprises and SMBs, the GXP2140 is the perfect choice for users looking for a high quality, feature rich multi-line executive IP phone with advanced functionality and performance.
There are two easy-access cosmetic battery capacity options:
Polycom® SpectraLink® 8400 Series Wireless Telephones Transforming Workflows with Mobile Unified Communications
Polycom SpectraLink 8400 Wireless Telephones improve productivity and responsiveness for on-site mobile professionals across a wide range of industries, including healthcare, retail, manufacturing and hospitality.
Built on open standards, the SpectraLink 8400 series transforms the delivery of mobile enterprise applications by bringing the power of thin client and browser technology to front-line professionals in an easy-to-use and easy-to-manage interface. Additionally, the SpectraLink 8400 series supports the industry’s broadest range of interfaces to enterprise-grade PBX, wireless LAN, and infrastructures to deliver maximum interoperability with the lowest overall cost of ownership in the industry.
Features and Benefits
Transform workflows with application, voice, and data integration
Supports the exacting needs of on-site mobile professionals
Broadest interoperability based on open standards
Lowest total cost of ownership in the industry
Solutions
Polycom wireless telephones are used daily by thousands of businesses throughout the globe.
Healthcare – Hospitals and clinics use wireless telephones to streamline communication to quickly respond to patient needs, improving patient satisfaction and overall quality of care. Increases in staff efficiency enable hospitals to address a shortage of trained health workers while improving nurse retention levels. Specialized applications such as Bar Code Medicine Administration also improve patient safety by reducing medical errors.
Retail – Managers and shop floor staff can be mobile, visible, and spend more time in front of customers, offering better customer service with stock, product, or promotional inquiries and responding more quickly to in-store incidents, such as spillages and theft.
Manufacturing – Wireless handsets eliminate paging delays, improve operations and enable faster responses to problems and emergencies, reducing downtime and production slippage. Push-to-Talk (PTT) technology replaces “walkie-talkie” systems with improved functionality.
Hospitality – Workplace wireless systems are an alternative to using expensive cellular phones, enabling organizations to offer a higher level of customer service much more efficiently.
General Office – Employees such as project managers, administrative assistants, and IT professionals use wireless telephones to minimize “telephone tag,” increasing productivity for workers who are frequently in meetings or otherwise away from their desks.
The Spectralink IP-DECT Base Station controls the traffic in the air and provides the link between Spectralink wireless handsets and Spectralink DECT Servers. The Spectralink IP-DECT Base Station interoperates with the entire Spectralink DECT Server portfolio.
Each Spectralink IP-DECT Base Station has 12 speech channels and covers a circular area between 30-300 meters in diameter. Because coverage depends on location specifics such as building materials and interference, a site assessment before installation helps to determine the proper number and placement of base stations.
The Spectralink IP-DECT Base Station integrates seamlessly with its predecessor (Kirk SIP-DECT bse), but there are several advantages when upgrading to the new IP-DECT Base Station:
Switch Cyberdata 011187
The Cyberdata 011187 switch is designed to expand a single PoE network connection on one or two PoE network ports. It has two ports.
It has power from a single 802.3at or 802.3af network switch and passes it through one or two of the ports.
In the basic configuration, one of the Gigabit down-stream ports can drive a VoIP phone directly with PoE and the other Gigabit port can be connected to a user's PC. No external power suplies would be required.
The Vega Enterprise SBC provides security and interoperability for enterprise networks. It’s your firewall for voice. The browser-based GUI makes the Vega eSBC one of the easiest to provision and manage. The straight-forward, session-based licensing model also makes the Vega eSBC one of the most cost-effective SBCs to deploy and maintain in the field.
Supports 25-250 simultaneous sessions/calls
Field upgradeable
Browser-based GUI for easier provisioning and management
Security and QOS for enterprise networks
DoS/DDoS attack protection
Network interconnect point for SIP trunking
Topology hiding for fraud protection
Hardware based transcoding
The A8 Series of analog cards supports up to eight (8) connections per card in your Asterisk system. Using Digium's advanced quad-port interface modules, A8 Series cards can scale from one (1) to eight (8) ports.
The modular nature of the cards allows you to mix and match between line (FXO) and station (FXS) interfaces, giving you exactly the ports you need. The A8 Series cards are available in half-length PCI and PCI Express form factors.
Digium cards are compatible with all versions of Asterisk using the DAHDI driver framework. Asterisk and DAHDI are available for free from the Asterisk.org website. All cards include a five (5) year warranty
Modules
GXV3610_HD Day/Night Fixed Dome HD IP Camera
The GXV3610 series is a powerful weather-proof Infrared fixed dome HD IP camera of outstanding performance and quality. Its advanced ISP (Image Sensor Processor) powered with state-of-the-art auto-exposure/auto-white-balance algorithm and a high quality lens, ensures high fidelity video quality that matches digital still camera color grade in a wide range of light environments.
It features cutting edge H.264 real-time video compression with excellent image clarity, industry leading SIP/VoIP for 2-way audio and video streaming to mobile phones and video phones, integrated PoE, IR-CUT for day/night mode, and advanced security protection. The GXV3610 series can be managed with GSurf_Pro (Grandstream’s intuitive FREE video management software that controls up to 36 cameras simultaneously) as well as other ONVIF compliant video management systems. It also offers an advanced and flexible HTTP API for easy integration with other monitoring systems.
The GXV3610 series is a powerful network camera for professional surveillance application in both indoor and outdoor environments.
Spectralink 8453 | Skype for Business | Scanners de código de barras integrados
Os terminais da série Spectralink 8800 são equipamentos ideais para profissionais que precisam de autonomia em seu ambiente de trabalho. Além disso, eles são fáceis de usar equipamentos graças à sua interface e têm um design ergonômico e leve ideal para transportar o terminal em qualquier lugar.
Com base em padrões abertos, a série Spectralink 8400 transofrma aplicativos de negócios móveis, colocando o poder do thin client e das tecnologias de navegação ao alcance dos professionais que trabalham na linha de frente por meio de uma interface fácil de usar e gerenciável.
Em particular, o modelo Spectralink 8453 incorpora todos os recursos da série, mas também incorpora scanners de código de barras integrados 1D/2D. Ele também inclui o Spectralink SAFE para ajudar a proteger o telefone e outros recursos relacionados à segurança, como "homem para baixo", que permite que um alarme seja configurado quando o trabalhador que transporta o dispositivo sofrer um acidente.
--- Não inclui fonte de alimentação
Spectralink 8453
Spectralink 8440 | Skype for Business | Spectralink SAFE
Especificamente, o Spectralink 8841 incorpora todos os recursos da série 8440, mas também inclui o Spectralink SAFE para ajudar a proteger o telefone e outros recursos relacionados à segurança, como "man down", que permite definir um alarme quando o trabalhador transporta o dispositivo sofre um acidente.
Mitel RFP 34 IP (outdoor)
The outdoor base station RFP 34 IP performs the outdoor operating requirements (class of protection IP 65). If required, radio relay antennas can also be used instead of the dipole antennas. The RFP 34 IP is powered using Power-over-Ethernet.
The Radio Fixed Parts RFP (L)32 IP and RFP (L)34 IP are connected directly to the LAN like a VoIP device and use the benefits of established DECT technology for radio transmission. This ensures full compatibility with cordless DECT terminals, which are available as system telephones and standard GAP terminals.
DECT * 120 DECT channels supported for maximum use of DECT capacity * 8 simultaneous voice channels per RFP, 4 additional channels for handover * GAP standard supported * Connection handover in line with the GAP standard * DSAA authentication between base and handset * Support of DECT encryption * Adicional antenna for RFP 32/34 IP. * Type of ingress protection: IP 65 * Flame resistance UL94 V0
Digium Modulo VPM435
Para los usuarios de Asterisk que se conectan a la red RTC (PSTN), el tipo más común de eco es el eco híbrido: el eco introducido por la falta de concordancia entre los circuitos telefónicos de 2 hilos y 4 hilos. El eco se manifiesta como un reflejo distorsionado y retrasado de la voz del usuario, en una conversación con una parte externa a través de la RTC (PSTN). Asterisk ofrece una serie de rutinas basadas en software de código abierto de cancelación de eco que son moderadamente efectivos para eliminar el desajuste del eco híbrido que experimenta la mayoría de usuarios RTC (PSTN) (a cargo de los recursos de la CPU). Sin embargo, hay casos en los que estos algoritmos no son eficaces. Para combatir esto, Digium presentó módulos de cancelación de eco hardware basados en DSP, para los distintas tarjetas telefónicas de múltiples puertos Digitales y Analógicos
For Asterisk users connecting to the PSTN, echo with the voice communication is possible and can be managed effectively with echo cancellation modules from Digium. Asterisk has a range of software-based Open Source echo cancellation routines that are moderately effective in eliminating the echo that most Asterisk users experience. However, there are cases in which these algorithms are not effective. To combat this, Digium offers a line of DSP-based echo cancellation modules for all the Digium analog and digital (T1/E1/J1) cards with the VPMOCT series modules.
TE435 (4E1 PCI Express-low profile) Digital Telephony Cards Quad Span (1TE435F - Four span digital T1/E1/J1/PRI PCI-Express x1 card)
Quad span digital cards are available in PCI and PCI-Express form factors. All quad span models can be combined with an Octasic DSP-based echo cancellation module from Digium to provide effective hardware echo cancellation across all channels.
Note: The TE410P model supports a 3.3v PCI slot only - typically available on newer server motherboards and in 64-bit PCI bus architectures. The TE405P is for use only with a 5.0 volt PCI slot.
Terminal DECT Siemens A540H
O telefone Gigaset DECT A540H oferece som de alta qualidade HSP e chamadas hands-free. Incorpora um repertório de 150 nombres e números com o nome do chamador e número. Além da função VIP, uma melodia específica pode ser atribuída ao contato.
Fone de ouvido sem fio Plantronics CS510
Esse fone de ouvido é conectado a uma base simplesmente pelo suporte do fone de ouvido, permitindo configuração rápida em ambientes compartilhados. A banda larga CAT-iq, o processamento digital de sinais (DSP) e outros recursos avançados de som oferecem um som de chamada natural com excelente nitidez.
Elastix® Call Center PRO es una solución potente, robusta, flexible y fácil de usar, diseñada para la automatización y gestión eficiente de los Contact Centers, permitiendo la colaboración en tiempo real y mejorando la productividad entre agentes y supervisores a través de una aplicación unificada y reconocida como lo es Elastix® y con la experiencia de nuestro addon Call Center.
Está diseñado para manejar campañas de llamadas entrantes y salientes mediante una consola de agente fácil de usar, una interfaz de administración de llamadas y un protocolo de comunicación propietario para el módulo (ECCP), lo convierten en una poderosa solución para Call Centers.
Elastix® Call Center PRO cuenta con un sinnúmero de funcionalidades que permiten la gestión y puesta en marcha de su Call Center y que se suman también a las características ofrecidas en nuestra versión Community de Call Center disponible a través del Marketplace.
Elastix® Call Center PRO puede ser integrado fácilmente a sus sistemas existentes y ofrece escalabilidad acorde al crecimiento de su negocio.
Posee una interfaz de agente fácil de usar, pero con herramientas poderosas como formularios dinámicos, que le permitirá generar fluidez en cada llamada.
Acceda a reportes detallados y precisos en tiempo real, que le permitirán mejorar la eficiencia y productividad de sus agentes.
Características
Gestiona las llamadas entrantes, transfiriéndolas a un operador. Cada uno de ellos puede estar habilitado con diferentes estrategias de llamado, como por ejemplo: ring all, round robin, fewest calls; permitiendo una personalización completa de la gestión, tiempos, audios, desbordes y escalado de llamadas.
Los operadores disponen de la información del usuario a través de una interfaz gráfica al momento de la comunicación, así como los formularios diseñados específicamente para la toma de datos de la campaña en curso.
El supervisor del call center no solo tiene acceso al estado de cada campaña, de los agentes y del sistema, en tiempo real, sino que adicionalmente administra todo tipo de información como el de las llamadas en cola, en espera, abandonadas, atendidas y tiempos promedios; e información, cantidad, promedios de atención y status del agente.
La interfaz web del Call Center Pro, faculta al supervisor el análisis del estado de las campañas desde cualquier computadora con un navegador web.
Es la aplicación en la que agente interacciona con el sistema. El operador inicia su sesión, gestiona su estado e inmediatamente dispone de toda la información de la campaña, como los datos del contacto y la información a completar durante la llamada.
El operador puede acceder a sistemas externos de terceros, configurados previamente por el administrador, obteniendo así una comunicación directa con otras aplicaciones web, como un CRM o aplicaciones propias del cliente.
Elastix Call Center PRO permite integrarse con su base de datos dispuesta en otro servidor, lo cual permitirá mejorar el rendimiento del sistema.
Se podrá formar grupos de trabajo conformado por muchos agentes y un supervisor, lo cual permitiría reportes basados en grupos, o la aplicación de filtros para revisar el rendimiento del agente por supervisor, de esta forma la gestión de la campaña es independiente a cada supervisor, brindándole mayor facilidad e información detallada acorde a sus objetivos.
La base de dato s de clientes para llamadas salientes puede contener varios números telefónicos por cliente, brindando mayores posibilidades de contactar al cliente.
Los formularios pueden ser públicos, disponibles para todos los teams y pueden ser usados por otros supervisores. Pero también puede crear formularios privados, de tal forma que los mismos puedan ser usados únicamente por las campañas del team al que pertenece. Esta prestación está habilitada para todos los formularios, sean estáticos o dinámicos
Al finalizar una campaña saliente se puede reciclar los contactos para que se vuelvan a gestionar. Permitiéndole utilizar nuevamente aquellos contactos que captaron interés para una segunda campaña. El reciclado también permite adicionar nuevos números a la campaña.
Las campañas salientes operan con múltiples listas de contactos, administradas por ventanas de tiempo y canales concurrentes. El sistema contempla lista de “no llamar” dentro de la campaña; y funciona en modo predictivo, calculando el volumen de llamadas requeridas dependiendo de las líneas, agentes disponibles y el tiempo de conversación de las llamadas; o en modo progresivo, realizando las llamadas a medida que existan agentes disponibles para atenderlas.
Al igual que en las campañas entrantes, los operadores disponen de la información del usuario así como de los formulario necesarios presentados a través de una interfaz gráfica al momento de la comunicación.
El supervisor gestiona mediante formularios, la información que los operadores ingresan durante la interacción con el cliente, siendo completamente configurables según sea el requerimiento de la campaña, como por ejemplo: las etiquetas informativas, comentarios, fechas, campos de selección de opciones. Cada registro ingresado, se almacena en una base de datos junto con los datos de cada llamada.
Al iniciar la sesión, el operador puede trabajar en modo alta disponibilidad o CallBack. En modo alta disponibilidad, el operador permanece siempre conectado al sistema y toda llamada transferida es automáticamente atendida por la consola del agente; aumentando la productividad del call center.
En modo CallBack, el operador permanece conectado al sistema, pero al momento de transferirle una llamada, debe responderla manualmente. Esta función es útil cuando el agente comparte otras tareas dentro del callcenter, permitiéndole realizar y recibir llamadas externas a las campañas.
Gestione a cada uno de los usuarios a través de una interfaz propia para el usuario supervisor. Los supervisores tienen propiedades exclusivas en el call center, y son los únicos que pueden gestionarlo. Están los supervisores generales y los supervisores de teams; los supervisores generales realizan los trabajos de administración del Call Center, mientras que los de teams son los encargados de sus respectivas campañas, clientes y reportes.
Podrá crear formularios que varían acorde a los datos que se vayan recopilando en las diferentes llamadas con clientes. El script desplegado en la interfaz de agente interactuará con él acorde a las respuestas que sean ingresadas, brindándole posibilidades de respuesta ante varios escenarios.
Podrá habilitar la detección por software, de máquinas contestadores, faxes, o IVRs para campañas salientes.
Cada administrador, tiene el control de su base de clientes independiente en cada campaña.
Vega Enterprise SBC 25 Call Upgrade
Digium’s new TE133 and TE134 single span digital cards are the latest additions to the Telephony Card family. The TE133 and TE134 design utilizes state of the art technologies to support T1/E1/PRI environments that require a high-performance, cost effective digital telephony interface card. The TE133 and TE134 have the ability to create a seamless network, interconnecting traditional telephony systems with Voice over IP technologies.
The TE133 and TE134 cards support industry standard telephony protocols, including Primary Rate ISDN (both North American and Euro Standard) protocol families for voice. Both line-side and trunk-side interfaces are supported, as well as advanced call features. Octasic based hardware echo cancellation is built in to the cards, which removes the task of echo cancellation from the systems CPU and increases overall system performance and call quality.
Digium cards are compatible with all versions of Asterisk using the DAHDi driver framework. Asterisk and DAHDi are available for free from the Asterisk.org website. All cards include a five (5) year warranty and are eligibile for Digium's no-risk ESP guarantee.
With the FlexBRI, now you can connect your analog lines into your BRI system using only one interface card.
Ideal for small or home offices, Sangoma’s industry-leading hybrid FlexBRI solution will seamlessly integrate youranalog fax machine with your BRI phone system, using only one PCI or PCI express slot and will fit in even the smallest 1U servers.
A single PCI or PCI Express slot hosts the connection for up to 4 ports of BRI and 2 ports of analog FXO or FXS and ensures common synchronous clocking for all channels with absolutely no signaling issues. The card is 100% software configurable.
Telco-grade, hardware echo cancellation is included.
Spectralink 8440 | Skype for Business
Especificamente, o modelo Spectralink 8840 incorpora:
Gateway Beronet BF4002GSMbox (2 GSM) BF4002GSMBOX
The berofix Gateways are a powerful and flexible Hardware Solution to connect 2 GSM Lines to any SIP based VoIP system.
The gateway embedded a berofix baseboard cards and each gateway can carry up to 2 modules BF2GSM
Specifications:
M5 DECT Repeater
The Snom M5 DECT repeater is ideal for extending the range of your singl-cell or multicell DECT solution in locations requiring mobile coverage across several floors or throughout large buildings. Furthermore, the repeater increases the reception range of individual bases and handsets and bridges gaps between base stations, increasing the free movement between base stations in a single seamless network and allowing calls to continue uninterrupted.
Suitable for
Professional DECT telephony Combining versatile business communication functionality with the intuitive features of the mobile carrier world, with the intuitive features of the mobile carrier world,
Wide coverage With its sleek appearance and comprehensive range of features, the M65 is ideal for customers who expect more of their phone system and require mobile coverage across several floors or throughout large buildings.
Up-to-date User Interface The M65 has a large backlit 2” color display and a backlit keypad for easy dialing in poor light conditions. Wideband audio ensures crystal-clear voice quality for your calls, and, in addition to 6 polyphonic ringtones, the handset also features vibration alert for incoming calls. An integrated tricolor LED keeps you instantly informed of missed calls, voicemail messages, and low battery status. The lithium-ion battery has a standby time of up to 250 hours and up to 17 hours of talk time for heavy usage between charging.
Comfort features When the M65 is used with a snom M700(1) or M300(2) base you enjoy many phone system features such as voicemail, automatic call forwarding, call lists, caller identification, or direct search in the corporate directory
The base station for cordless telephones snom M700 is ideal for companies who need a wireless coverage on several floors or inside large buildings solution.
Multicellular snom DECT solution can connect up to 40 bases including M700, creating a network with perfect functionality that ensures the seamless transfer between a base and another. Calls initiated on a base station continue without interruption as they move from one base to another.
Multicellular solution can be easily scalable and extended up to 40 bases M700, 200 mobile and 100 repeaters on the same network. Moreover, this multistage device supports the snom DECT repeaters M5 encryption to increase the radius mostly reception and fill the possible absence of signal from one base to another.
M700 DECT
The synchronization and installation is automatic to avoid having to use a costly additional management devices DECT. Multicellular configuration mode is simple and intuitive thanks to the integrated installation of M65 phones.
With the multicellular system M700 in your offices, all snom M65 phones connected offer multiple telephone features such as direct search in the corporate directory and HD sound. In addition, the system excludes any wireless security problem: the DECT encryption and TLS and SRTP protocols ensure maximum protection of all SIP communications. Moreover, no additional license codec functions, audio channels, etc. are needed.
Finally, thanks to the functionality offered by snom provisioning, the system is fully interoperable with leading VoIP systems.
Patton Licence upgrade 1 session adicional SN5300
The SN5300 connects to the IP-PBX or UC system in the Enterprise’s LAN and to an Internet telephony service provider (ITSP), creating a single conduit for multimedia components including voice, video, and data.
Patton Licence upgrade 1 session adicional SN5300 (up to 250 sessions in Patton SN5300)
Patton SmartNode 5300 ESBR
SN5300 is a CPE based Session Border Controller, delivering the features you need for advanced multiservice voice and data network applications. It combines highly flexible SIP routing and manipulation features with powerful quality of service IP routing functions to build professional and reliable VoIP and data networks.
Patton ESBR SN5300 4 sessions SBC, upgradeable to 250
The SN5300 enables Universal SIP Trunking and provides a single Integrated Access Device with features like IP Routing, VoIP and IP Security and a SIP registrar for survivability. The SN5300 connects to the IP-PBX or UC system in the Enterprise’s LAN and to an Internet telephony service provider (ITSP), creating a single conduit for multimedia components including voice, video, and data. Whether it is a new installation or an existing deployment, this device will aid you in, deploying, troubleshooting, logging, and security while increasing the flexibility of your network.
Applications The SN5300 enables protocol conversion between two networks to solve interop problems for devices using SIP TCP signaling only. The SmartNode is able to convert SIP TCP or SIP TLS signaling into SIP UDP signaling. Using the built-in QoS engine, the SmartNode ensures that voice traffic gets top priority resulting in good voice quality across the SIP Trunk over a public network.
The industry’s first fully expandable, SIP-based tabletop conference phone, MAX IP delivers unrivaled audio clarity and room coverage for your VoIP phone system. Sporting ClearOne’s legacy HDConference® audio-processing technology, the MAX IP system provides unparalleled sound quality for natural-sounding phone conferences.
Link up to four phone units, providing full coverage for larger conference rooms with multiple speakers, multiple microphones, and multiple dial pads distributed throughout the room for unrivaled coverage.
Unsurpassed audio clarity
ClearOne’s, market-leading, HDConference™ audio technology
Proprietary echo and noise cancellation algorithms prevent distracting audio
Full-duplex audio means no cutting in and out
Automatic gain and level controls adjust mic and speaker levels automatically
First-mic priority intelligently focuses mic levels based on who is speaking
Adaptive modeling-continuously samples room acoustics for any changes
Three built-in microphones array with 360° audio pickup up to 12 feet
Large loudspeaker for rich, clear conferencing or audio playback
SIP suite of features
3-way calling - allows for ad-hoc conferences without need for a conference bridge
VLAN tagging - allows users to manage bandwidth usage on the network
TLS & SRTP encryption-ready (with future release of firmware upgrade) - secures voice communications over the network
Field upgradeability - allows users to easily download firmware upgrades from ClearOne website and load directly into the conference phone
Ease of use
True plug-and-play capability
WARRANTY: 2 years, From date of purchase
The industry’s first wireless, tabletop conference phone, the MAX® Wireless system can turn any room into a conference room. The wireless capabilities entail far more than a clean look; providing the freedom to take the conference with you from room to room, with secure, encrypted transmissions.
Powerful HDConference® processing provides the crisp, clean audio you expect from ClearOne, in a professional conferencing system that can move from the executive office to small conference room with ease.
Front Panel of NLX4000 for TE133
CyberData’s new SIP-enabled Talk Back Speaker enables two-way conversations in settings such as classrooms, offices, medical facilities and clinics. A remote call button (which is included for the SIP-enabled version), enables calls to a predetermined extension that can be initiated from the room with the speaker. The new SIP-enabled Push-to-Talk Talkback Speaker also includes a monitor mode function where the speaker may be called from a remote phone that discretely monitors the activity in the room.
The Push-To-Talk function for the speaker does the best job of addressing high background noise levels, effectively providing clear two-way communication.
The SIP-enabled Talk Back Speaker easily connects into local area networks with a single CAT5/6 cable from your PoE switch. Its small footprint allows the speaker to be mounted almost anywhere with multiple mounting options available.
Gateway Beronet BF4004GSMbox (4 GSM) BF4004GSMBOX
The berofix Gateways are a powerful and flexible Hardware Solution to connect 4 GSM Lines to any SIP based VoIP system.
The Vega Enterprise SBC provides Security and Interoperability for Enterprise Networks. The browser-based GUI makes the Vega eSBC one of the easiest to provision and manage. The straight-forward, session-based licensing model also makes the Vega eSBC one of the most cost-effective SBCs to deploy and maintain in the field.
Denial of Services
Configuration errors
Theft of service / Fraud
BYOD
El GXP2130 es un avanzado teléfono IP de categoría empresarial con hasta 3 líneas, TFT LCD color de 2,8”, 4 teclas XML programables sensibles al contexto, 8 teclas de extensión BLF programables, doble puerto de red Gigabit, PoE integrado, conferencia de voz de 4 vías y Electronic Hook Switch (EHS). El GXP2130 ofrece calidad de audio superior en el auricular y el altavoz, recursos telefónicos valiosos y de última generación, información personalizada y servicio de aplicación personalizable, aprovisionamiento automático para fácil instalación, protección de seguridad avanzada para mayor privacidad e interoperabilidad amplia con la mayoría de los dispositivos SIP de terceros y las principales plataformas SIP/NGN/IMS. Ideal para pequeñas y medianas empresas, grandes negocios, oficinas pequeñas y home offices, el GXP2130 es la elección perfecta para usuarios que buscan un teléfono IP de alta calidad, con variedad de recursos, funcionalidades avanzadas y sencillo de usar.
SIP-T19
The SIP-T19 is one of Yealink’s latest answers for the entry-level IP phone that offers features and performance normally associated with much more advanced phones.
The quite intentional choice of high-quality materials, combined with a generously large 132 x 64-pixel graphical LCD that gives a clear 5-line display, guarantees both a smoother user experience and easy access to much more visual information at a glance. Dual 10/100 Mbps network ports.
The SIP-T19P supports a single VoIP account, simple, flexible and secure installation options,plus IPv6 and SRTP/ HTTPS/ TLS, VLAN and QoS. It includes headset use, is wall-mountable and has been designed very specifically for better business.
beroNet Small Business Line Gateway: 1S0. BFSB1S0 (1 BRI)
BFSB1S0 (1 BRI) Specifications:
Sangoma Netborder Carrier SBC 2000 calls
The Sangoma NetBorder SBC Carrier provides flexibility and security to its IP telephone network acting as an intermediary between devices, networks and protocols and as a barrier against internal and external threats that may arise. Some of its salient features:
Serviço de consultoria - 10 horas extras
Nosso departamento de pesquisa e desenvolvimento fornece um catálogo completo de serviços de engenharia que inclui consultoria professional e suporte técnico para projetos de telefonia e comunicações.
Você pode contratar bônus de horas ou únicas (10 e 50 horas) que incluem suporte por telefone e por e-mail.
Serviço de consultoria - 50 horas extras
Cyberdata SIP Paging Adapter - 011233
The Cyberdata SIP Paging Adapter is a VoIP endpoint that interfaces analog paging systems with SIP and Multicast-based audio sources.
The SPA (SIP Paging Adapter) can be configured to support two separate SIP extensions. SIP extension one passes audio through to the analog output spoken from the caller's handset. When called this SIP extension two plays a bell audio that can be used as a night ringer when configured in a night ring group.
The SPA supports up to 9 user-uploadable messages that can be played by a DTMF command. The SPA supports a line-IN input for playing background music. Messages can be played either:
LCD Expansion Module
EXP40
The EXP40 Expansion Module for the SIP-T46G and SIP-T48G, expanding the functional capability of your sip phone to a whole new level. It features a large graphic LCD.
Two pages of 20 flexible buttons are shown on the display can be programmed up to 40 various features, the productivity-enhancing features include BLF/BLA, speed dialing, call forward, transfer, park, pickup, etc.
Display
Features Keys and Indicator
Physical Feature
The GXV3240 Video IP Phone for Android™ combines a 6-line IP video phone with a multi-platform video conferencing solution and an Android smartphone to offer an all-in-one communications solution. This Video IP Phone runs the Android Operating System and therefore offers full access to the many Android apps in the Google Play Store™, including popular productivity and business apps. The phone features integrated Bluetooth, Gigabit ports, a built-in web browser, integrated WiFi, a 4.3 inch touch screen, HD audio and PoE. By combining the power of multiple devices into one, the GXV3240 is the ideal solution for all communication, productivity and video conferencing needs.
With its 8 programmable keys, XML capabilities, native DHSG/EHS headset support, a true HD handset and a speakerphone that delivers remarkable wideband HD audio quality, the Mitel 6865i SIP phone is ideally suited for the small to large business market that needs Gigabit throughput for PC connectivity.
All 6800i Series SIP phones feature Mitel’s high definition Hi-Q™ audio technology to deliver enhanced performance and voice clarity. Integrating HD wideband audio codecs, advanced audio processing and hardware components that support a true wideband frequency range, the 6865i offers a superior voice experience on each audio path – handset, speakerphone or headset port – making conversations crystal clear and more life-like.
With an extensive storage capacity for personal directories, callers logs, redial lists, and 8 programmable keys, the Mitel 6865i is designed to easily manage access to all of the most frequently used call management features of the phone. Even more productivity enhancing features are available at the push of a button including: Shared Call Appearance (SCA), Busy Lamp Fields (BLF), 3-way conferencing, transfer, call waiting, call park, call pickup, intercom and paging. The 6865i is also an expandable desktop phone supporting up to three expansion modules further enhancing its productivity capabilities with additional programmable keys that can be used to access the broad list of advanced features offered by this powerful phone.
The Mitel 6865i features an innovative headset port that uniquely provides dual support for DHSG/EHS and modular 4-pin headset connections. Users with wireless headsets that support DHSG/EHS can now connect directly to the 6865i without the added cost of additional adaptors or cables.
Environmentally Friendly All Mitel 6800i Series SIP phones have been designed to be environmentally friendly. Using dynamic PoE class reporting, the 6865i has a PoE Class 2 rating that automatically switches classes when an expansion module is connected enabling the phone and network switches to efficiently manage its power consumption. The Mitel 6865i also supports an optional Efficiency Level “V” compliant power adaptor if required. With smaller packaging that includes 100% recycled and biodegradable material, the Mitel 6800i Series is one of the most environmentally friendly family of SIP phones on the market.
Cyberdata Switch Ethernet de 3 puertos Gigabit - Modelo 011259
The Cyberdata 3-Port USB Gigabit Port Mirroring Switch enables users of a network-attached device to split a single Gigabit Ethernet port into two Gigabit Ethernet ports for diagnostics purposes.
Engineers, programmers and field service personnel can utiliza this port mirroring function to monitor the network traffic to and from that device with network-monitoring software such as Wireshark. Power for the switch can be supplied by the host PC's USB port or from a standard +5V USB phone/camera charger.
Alphatech Portero IPDP01 Antivandálico (SIP 1 botón)
Este Portero Automático o Interfono VoIP (solo audio) logrará satisfacer sus necesidades de comunicación con las personas en la puerta principal del edificio, la entrada de la empresa, o la puerta de casa familiar. La flexibilidad de la solución reside en la posibilidad de conectar este dispositivo a una red Ethernet, una centralita VoIP o registrada con su servidor SIP a través de Internet.
Terminación metálica para uso en situaciones donde requiere protección adicional frente vandalismo.
El portero automático puede ser alimentado con un transformador AC/DC a 12V (no incluido) o por PoE (Power over Ethernet) haciendo innecesario el cableado adicional.
Una de sus características destacadas es que incluye dos relés de apertura (p.ej para 2 puertas por medio de cerraduras eléctricas conectadas). Lleva un servidor web integrado, que puede ser controlado desde cualquier navegador web, por ejemplo, Firefox, IE, Mozilla.
Características generales:
- Intercomunicador SIP para llamadas P2P o a través PBX.
- Solución compacta con botón de llamada, altavoz/micro.
- Iluminación constante del botón de llamada.
- Hasta 64 x 2 memorias de extensiones programadas (hasta 16 dígitos), usado de dos formas:
a) en modo noche/día, según esté el intercomunicador en día o noche, la pulsación de sus botones elije el número del grupo de noche o de día
b) en modo grupos, si la llamada a la extensión del primer grupo falla se intenta con el segundo grupo.
- Dos relés independientes accionables en 5 modos diferentes:
1. Apertura de puertas mediante marcación de secuencias de DTMF desde el teléfono interior durante la llamada o mediante passwords marcadas en el teclado del intercomunicador sin necesidad de llamada.
2. y 3. Modos cámara-luz, el relé se activa durante la llamada.
4. Modo timbre, el relé se activa al iniciar la llamada
5. Modo gradual, la actuación del primer relé dispara tras un tiempo la actuación del segundo relé.
- Dos códigos DTMF para finalizar las llamadas desde el teléfono interior.
- Duración máxima programable de la llamada, con posible extensión marcando * ó # desde el teléfono interior.
- Acceso WEB para modificación de parámetros, actualizaciones de firmware y video JPEG.
- Utilidad de PC UDVGuard (Windows) con la que es posible acceder al portero y activar sus relés directamente; programable para que emerja ante una llamada entrante.
- Audio G711, GSM, G726-23, DTMF RFC2833 o SIP INFO.
- Ethernet 10/100 con Auto-MDIX.
- Alimentación por POE IEEE802.3af clase 3 o mediante fuente 12V @ 1A AC/DC.
- Basado en Linux 2.6
- Grado de protección IP44
ESTRUCTURA METALICA
Sangoma Netborder Carrier SBC 1000 calls
The Sangoma NetBorder SBC Carrier provides security and flexibility to your VoIP network acting as an intermediary between devices, protocols and networks. It offers interoperability, easy configuration via GUI and security against internal and external threats that may arise in the VoIP infrastructure.
Atualização do Skype for Business
Com esse serviço, oferecemos ao cliente a possibilidade de atualizar seu sistema do Lync 2010 ou Lync 2013 para o Skype for Business 2015.
Dependendo das necessidades do cliente a da possibilidade ou não de interrupção dos serviços, a maneira apropiada será escolhida para realizar a atualização corretamente.
Se você quiser atualizar do Lync 2010 para o Skype for Business 2015, escolha a opção "micração" lado a lado", na qual uma nova infraestrutura do Skype for Business é adicionada à topologia do Lync 2010. Posteriormente, usuários, serviços e pontos de extremidade para a infraestrtura do Skype antes de desatribuir o Lync 2010 antes de desatribuir o Lync. Não haveria interrupção do serviço.
Se e atualização for feita do Lync 2013 para o Skype for Business, usaremos a opção "atualização no local" na qual temos dois modos possíveis:
No final de 2018, está previsto o lançamento do Skype for Business 2019. O Avanzada 7 também oferecerá o serviço de atualização par aesta nova versão.
Khomp UMG Server Modular PRO i3
O UMG Server Modular PRO é um appliance integrado por um gateway de voz (aceita diferentes interfaces de telefonia) e um servidor com placa-mãe e processador de alto desempenho, dedicado à instalação de qualquer plataforma baseada em Windows, Linux ou FreeBSD.
Com este appliance, você pode desenvolver um producto completo (como um centro de comunicação unificado ou uma central telefônica com roteamento de chamadas) e criar soluções de firewall, com a possibilidade de configurar acionadores de alarme por IP ou chamada de celular.
Este UMG Server Modular PRO pode ser integrado com várias opções de armazenamento e processamento, bem como interfaces de telefonia que melhor se adquam ao negócio no qual ele será aplicado. Existem 3 módulos disponíveis que suportam interfaces de telefonia modulares: E1, FXS, FXO e GSM 2G ou 3G, RAM que pode alcançar até 16GB e 4 portas SATA para conexão com SSD ou HD 2.5''.
Aplicações
Gateway Khomp UMG Server 104
UMG Server 104 is a device designed for integrators who want to develop a centralized solution based on E1/T1 and SIP calls for their final client. This device is "all in one": server + gateway, to load your software application for PABX, call center and dialer, among others, with a motherboard for the installation of any platform based on Windows, Linux or FreeBSD.
The UMG Server 104 can be composed with various storage options, allied to the telephony E1/T1 interface, a RAM module of up to 8GB and two p orts of SATA type for connection with SSD or 2.5'' HD with storage for up to 1TB each one.
Gateway features
Optional elements
Khomp UMG Server Modular 00/00/00
O UMG Server Modular é um appliance integrado por um UMG Modular 300 configurável com diferentes interfaces de telefonia e um servidor com placa-mãe para a instalação de qualquer solução de telefonia baseada em Windows ou Linux. Isso torna o UMG Server Modular um plataforma ideal para desenvolvedores de soluções SIP, pois oferece hardware de alto desempenho e confiabilidade para integrar seus aplicativos de PBX, IP PBX, marcadores de call center, URA, etc.
Tem capacidade para processar até 46 chamadas simultâneas através dos canais de telefonia física, que podem ser utilizados entre E1/T1, FXS, FXO e GSM. Além disso, é possível rotear canais VoIP, permitindo que chamadas sejam feitas de um PBX IP para um operador de telefonia VoIP, por exemplo. Toda a convergência de sinais e roteamento de chamadas é processada através do gateway, liberando assim o processamento da placa base para uso exclusivo do sistema operacional e do aplicativo que será instalado no dispositivo.
Hardware
Gateway Khomp UMG 104 (RJ)
The Khomp UMG 104 gateway is an ideal device for small and medium businesses looking for a competitive investment team and high reliability.
The UMG 104 can be connected to 3 VoIP operators assigning each link to a particular region in this way reduce the costs with national and internacional calls at much lower values than the conventional rates. In addition, it can be applied to VoIP operators that work with the sale of minutes to professionalize services among many other options.
The UMG 104 has 30 E1 channels and 30 VoIP channels, with the capacity to operform up to 30 simultaneous calls. This UMG model has 4 Ethernet ports, an easy-to-use web interface, failover of routes and greater control of expenses thanks to the possibility of configuring the routing by prefixes and / or by the loyalty of operators.
UMG 104 is a compact system that can be divided into 3 basic parts:
Gateway UMG 100 (Conector BNC)
The UMG 100 is another of the gateways belonging to the "User Media Gateway" series of Khomp for small scenarios with high performance guarantee. In particular, this model supports 1 E1 link, up to 30 VoIP channels, registration in up to 10 different SIP accounts and is configured to connect to the Public Telephony Network (STFC), VoIP links, soft-switches and PABX equipment.
This model incorporates failover of routes, this avoids the inoperability of the calls in case of failure in a SIP server by means of resource Keep Alive, in this way when Keep Alive is active, the UMG sends messages of type OPTIONS for the SIP server for monitor your status.
Specifically, this UMG model is a compact system in which three basic parts can be defined:
Integration with IP PBX
Integración con PBX Tradicional
This UMG model is a compact system in which three basic parts can be defined:
Khomp Gateway KMG SBC 90
KMG SBC 90 is a low density device of the B2BUA type for small scenarios. It has the capacity to perform up to 300 SIP sections or up to 90 calls with transcoding, it has powerful signal processors for converting media and protocols between networks. The SBC 90 guarantees a secure connection between the local network and the VoIP oprator and also includes the VoIP operator and also includes the external telephony modules, with the GSM, E1/T1, FXO and/or FXS technologies, as long as the maximum limit of 60 is respected. STFC telephony channels.
It's an ideal device for call centers and professionals that provide small services that need complete administration of the IP telephony operation with advanced resources.
Gateway Khomp KMG 400 MS
The ÇUMG 400 MS gateway is a Khomp media gateway with up to 120 channels, modular interfaces and integrated SBC.
It also presents the possibility of adding 2 external telephony modules. These modules can include E1/T1, GSM, FXO and / or FXS, as long as the maximum of 120 telephony channels is respected.
In addition, this KMG 400 MS series offers the ideal solution for companies and institutions with communiation needs through IP telephony exchanges since SIP connection sections can be made through this series.
Gateway Khomp 200 MS - 2 E1 (Conector BNC)
The UMG 200 MS is a gateway Khomp with SBC integrated and configurable thanks to its modular interfaces. This particular model contains 2 internal E1 links and BNC connector.
It also presents the possibility of adding 2 external telephony modules. These modules can include E1/T1, GSM, FXO, and / or FXS, as long as the maximum of 60 STFC telephony channels are respected and two of their network ports are used. (See more details on product technical sheet).
In addition, this series KMG 200MS offers the ideal solution for companies and institutions with communication needs through IP telephone exchanges since SIP connection sections can be made through this series.
Application model
Khomp UMG Modular 300 - 00/00/00
The UMG Modular 300 é um dispositivo ideal para empresas de pequeno e médio porte que buscam implementar os benefícios da telefonia VoIP em seu ritmo de trabalho e integrá-lo aos links: E1/T1, GSM 2G e 3G, FXS e / ou FXO. Desta forma, a interconexão pode ser feita entre redes digitais, analógicas, móveis e IP na mesma plataforma de telefonia.
O UMG Modular 300 tem capacidade para processar até 46 chamadas simultâneas, divididas em 3 slots para qualquer combinação entre as interfaces de telefonia existentes. Além disso, um canal VoIP pode ser adicionado para cada canal selecionado.
Ele se conecta a um máximo de 2 diferentes operadoras de VoIP, o que permite redirecionar as chamadas para rotas de menor custo, sejam nacionais ou internacionais. Isso reduz os gastos com telefonia em valores muito mais baixos do que as taxas convencionais.
Como o UMG Modular 300, você pode conectar telefones analógicos e/ou telefones IP a uma solução IP PBX. Desta forma, uma série de funcionalidades adicionais são adicionadas em relação a um PBX convencional. Entre as possibilidades de uso do UMG Modular 300 estão a interconexão das extensões da matriz e das subsidiárias da empresa, em qualquer parte do mundo. Assim, as unidades podem fazer ligações entre si através da Internet sem custo, entre muitas outras vantagens.
Aplicaões
Khomp UMG Server Modular PRO i5
Khomp UMG Server Modular PRO i7
Gateway Khomp 200 MS - 1 E1 (Conector RJ)
The UMG 200 MS is a gateway Khomp with SBC integrated and configurable thanks to its modular interfaces. This particular model contains 1 internal E1 link and RJ connector.
Auditoria Skype for Business
Com este serviço de auditoria, verifica-se o estado atual da plataforma de comunicação, atentando-se para a forma de obter os diferentes dados e sua segurança dentro do sistema.
Dependendo do escopo acordado com o cliente, as diferentes camadas ou níveis que compõem o sistema e seus relacionamentos podem ser analisados:
Inclui também tarefas como:
O resultado da auditoria será um relatório com uma redação simpleas e concisa que destaca os problemas encontrados, os efeitos e as recomendações
Nota: Se o serviço atingir a infraestrutura física e de rede, será necessária uma visita às instalações do cliente, na qual informações dessas áreas serão coletadas.
Skype for Business consulting
Dentro deste serviço de consultoria, suporte e consultas remotas estão incluídos na plataforma Skype for Business. A resolução de incidentes seria tratada como propostas independentes. O escopo deste serviço seria:
O nível de serviço seria um "próximo dia útil" 5x8
Instalação do Skype for Business
O Skype for Business é a solução Microsoft Unified Communications oferece serviços como:
Oferecemos o serviço de planejamento, implementação e configuração da solução Skype for Business mais adequada ao cliente. Podemos escolher a solução local, pelo Skype for Business Online (Office 365) ou por uma solução híbrida.
Este serviço inclui:
1.- Planejamento
2.- Desenvolvimento
3.- Administração
Curso Online SIPWISE NGCP Formación dirigida a operadores de telefonía que quieren dar servicio residencial, empresarial o wholesale. Con este curso, los técnicos de cualquier operador, ya sea pequeño o enorme, aprenderán a implementar y mantener una solución de telefonía de primer nivel. El curso es una iniciación y puesta en marcha de la plataforma NGCP SIP: Provider CE, que está considerada la mejor solución para operadores por su calidad, estabilidad y escalabilidad. Durante el curso, los asistentes aprenderán los conceptos, arquitecturas y tecnologías más eficientes para la puesta en marcha y el mantenimiento de sus propias soluciones. Este curso está dirigido a: Ingenieros de VoIP que quieran adentrarse en el mundo de las soluciones de operador. Departamento técnico de operadores. Ya sea desarrollo, sistemas o soporte. Empresas de servicio a Operadores que quieran ofrecer una solución de telefonía a sus clientes. Características del curso SIPWISE NGCP Duración 4 días. Tipo de curso: Online Tipo de docencia: Teórico Idioma del curso: Castellano Horario del curso: 4 sesiones online de 3 horas. 15:30 - 18:30 Perfil Docente: El curso será impartido por Jon Bonilla. Jon estuvo trabajando durante 5 años en Sipwise y fue parte fundamental del desarrollo de la solución presentada por lo que conoce mejor que nadie los entresijos de la misma. Jon ha realizado docenas de despliegues de la solución en operadores de 4 continentes y ha impartido docenas de cursos como este. Se trata de la persona con más experiencia teórica y práctica que se puede encontrar.
Videophone IP Grandstream GXV3370
O GXV3370 é um videofone para empresas da Grandstream com tela sensível ao toque de 7 polegadas, câmera com qualidade HD para videoconferência, Wi-Fi e Bluetooth embutidos, Gigabit e com recursos inovadores de telefonia. Entre eles, descobrimos que ele funciona no Android 7.0 e incorpora suporte flexível ao SDK para aplicativos personalizados.
Além disso, o Grandstream GXV3370 é interoperável com quase todas as principais plataformas SIP e pode ser integrado com produtos Grandstream, incluindo câmeras de segurança baseadas em SIP, IP PBX, videoconferència, etc.
Yealink Dongle DD10K
O Yealink DD10K DECT Dongle permite que os modelos T41S e T42S trabalhem simultaneamente com o sistema Yealink DECT com um único telefone após o emparelhamento com a estação base Yealink W60B DECT IP.
Ele possui uma poderosa velocidade de transmissão de 552kbit/s, proporcionando uma velocidade de conexão DECT rápida e confiável, sem problemas de fiação, oferecendo o privilégio de desfrutar de comunicações sem fio em minutos.
Como complemento à série Yealink DECT, o dongle DD10K oferece ao seu telefone de mesa a opção de aumentar as funções do telefone de mesa com os recursos DECT. Além disso, o Yealink DD10K está em conformidade com os padrões CAT-iq 2.0 e sua implementação é fácil e segura, garantindo uma experiência de usuário otimizada e flexível.
Gateway Mediatrix G7 - 24 FXO
O Mediatrix G7 é um adaptador analógico VoIP confiável e seguro e um gateway para SMBs.
Especificamente, esta versão do Mediatrix G7 incorpora 24 FXO.
É amplamente interoperável com SIP softswitch e fornecedor.
Gateway Khomp KMG 750 SBC
The SBC 750 KMG gateway is a Khomp media gateway capable of supporting up to 32 E1/T1, or 960 telephony channels, which can use E1/T1, GSM, FXO and/or FXS technologies.
It supports 450 VoIP SBC and is ideal for use in network structures that need the highest voice quality. In addition, it has 13 network ports, one of them specific for high availability.
Khomp Gateway KMG 3200 MS
Khomp KMG 3200 gateway is part of Khomp's medium gateway, a high density device capable of supporting up to 32 E1 / T1 links or 960 telephony channels, which ca use E1 / T1, GSM, GXO and / or FXS. In addition to the STFC channels, it also supports 450 sIP VoIP and is ideal for reliable network structures that need the highest voice quality.
It has 13 network ports, one of them specific for high availability, with the ability to transfer the processing of calls from the gateway to a chassis in standby mode, in the event of a hardware failure. In this way, network configurations, including IP addresses, are maintained.
Gateway Khomp 200 MS - 1 E1 (Conector BNC)
The UMG 200 MS is a gateway Khomp with SBC integrated and configurable thanks to its modular interfaces. This particular model contains 1 internal E1 link and BNC connector.
The UMG 200 MS is a gateway Khomp with SBC integrated and configurable thanks to its modular interfaces. This particular model contains 2 internal E1 link and RJ connector.
Gateway Khomp UMG 240 FXS
The UMG FXS 240 is a mid-line gateway of the Khomp brand. The device is ready to connect to VoIP links, software switches and PBX equipment and can have up to 24 FXS channels and up to 10 registered SIP accounts.
Furthermore, it is compatible with the main signals and codeecs of the market, in addition to the control and routing of calls, according to programmed rules.
It has two internal plates:
Gateway Khomp UMG 104 (BNC)
Telefone IP Sangoma S705
Os telefones Sangoma IP são smartphones fáceis de usar, rápidos e intuitivos, projetados para funcionar com o FreePBX e o PBXact. Ao instalar o telefone, o servidor de redirecionamento aponta automaticamente para o FreePBX / PBXact para sua configuração, para que sua configuração seja rápida e fácil. Também é compatível com outros provedores adicionando os detalhes do IP PBX.
Por outro lado, o software EndPoint Manager é ativado automaticamente ao usar um telefone Sangoma. Isso permite que os usuários controlem configurações flobais, programem suas chaves de telefone, extensões, façam upload de imagens, baixem novos firmwares e outras funcionalidades.
O S705 é um telefone com funções muito completas. Ele tem 6 contas SIP a um preço muito competitivo e é provisionado automaticamente em apenas alguns segundos. Além disso, incorpora conexão WiFi e Bluetooth para uso de fones de ouvido sem fio.
Snom D120 IP Phone
The new IP phone of Snom D120 is an ideal phone for SMEs and self-employed thanks to its comfort and efficiency. An ideal terminal for any scenario where you need to cover the basic communication needs of the user.
This IP desk phone comes equipped with a backlit graphic display and offers extraordinary visibility of calls and other functions that place it above standard desk phones. In addition, it also has other features such as LED indicator for notification of incoming calls, calls in progress and pending messages and has four configurable dynamic keys to suit the user to automate the telephone functions you want. A phone with low energy consumption ideal for its excellent quality / price ratio.
Grandstream Repetidor DECT DP760
El repetidor de Grandstream DP760 es un poderoso repetidor DECT de banda ancha (estación de retransmisión inalámbrica) que se asocia a la estación base DECT DP750 de Grandstream. De esta manera el sistema ofrece un alcance adicional de 300m al aire libre y 50m en interior para dar a los usuarios la libertad de mantenerse activo en su casa o lugar de trabajo.
Retransmite hasta 2 llamadas HD simultáneas. Su conexión de Ethernet proporciona PoE para su instalación conveniente y una variedad de funciones remotas incluyendo aprovisionamiento, monitoreo de estado y actualizaciones de firmware del repetidor.
Cuando se conecta con la estación base DECT DP750 y los auriculares DP720 de Grandstream, el DP760 ofrece una poderosa solución DECT extendida para aquellos usuarios que buscan añadir cobertura a su sistema VoIP DECT.
Cabo para CUBO 78xx 88xx suprimentos
PSU Cisco CUBE-4
Para terminais da série 8800 é Cisco | 110/220 Volts AC
PSU Cisco CUBE-3
Para terminais da série 7800 é Cisco | 110/220 Volts AC
Terminal de conferência de áudio Cisco 7832
O terminal de conferência de áudio Cisco 7832 oferece qualidade de áudio de alta definição para escritórios executivos e pequenas salas de reuniões com até 6 participantes. Ele incorpora uma cobertura de microfone de 360 graus graças a 4 microfones que permitem que os usuários falem com uma voz normal e sejam ouvidos claramente a uma distância de até 2.1m.
The EGW208 Elastix GSM Gateway allows you to extend your Elastix server with GSM trunks, which will allow you at a low cost to reach those places where cellular signal is the only means available. Thanks to its friendly user interface and modular design, you shall be able to easily configure your EGW208 GSM Gateway, as well as to complement a development solution via AMI.
Its modular design will allow the expansion according to need; the EGW208 GSM Gateway provides two 4-GSM channel modules. It will enable to interconnect with a broad range of codecs, including G.711A , G.711u, G.729, G.722, G.723, G.726 and GSM, to the GSM cellular networks, which will help you significantly reduce costs and optimize communications in your business.
You will never have to stop communications at your business thanks to the hot-swap capability of the EGW208; remove or change SIM cards, or if you prefer, the whole module, without altering the course of your communications.
Insert a SIM card in the gateway, create a trunk, and enable telephony within minutes!
Potential Applications
Technical Specifications
GSM
SIP
Network
Elastix GSM EGW202
The EGW202 Elastix GSM Gateway allows you to extend your Elastix server with GSM trunks, which will allow you at a low cost to reach those places where cellular signal is the only means available. Thanks to its friendly user interface and modular design, you shall be able to easily configure your EGW202 GSM Gateway, as well as to complement a development solution via AMI.
It's modular design allow the expansion according to need; the EGW202 GSM Gateway provides one 2-GSM channel module. It will enable to interconnect with a broad range of codecs, including G.711A, G.711u, G.729, G.722, G.723, ÑG.726 and GSM, to the GSM cellular networks, which will help you significantly reduce costs and optimize communications in your business.
The EGW204 Elastix GSM Gateway allows you to extend your Elastix server with GSM trunks, which will allow you at a low cost to reach those places where cellular signal is the only means available. Thanks to its friendly user interface and modular design, you shall be able to easily configure your EGW204 GSM Gateway, as well as to complement a development solution via AMI.
Its modular design will allow the expansion according to need; the EGW204 GSM Gateway provides one 4-GSM channel module, but you shall be able to add one additional module, increasing the capacity to 8 GSM channels. It will enable to interconnect with a broad range of codecs, including G.711A , G.711u, G.729, G.722, G.723, G.726 and GSM, to the GSM cellular networks, which will help you significantly reduce costs and optimize communications in your business.
You will never have to stop communications at your business thanks to the hot-swap capability of the EGW204; remove or change SIM cards, or if you prefer, the whole module, without altering the course of your communications.
Issabel® is an Open Source Software to establish Unified Communications. About this concept, Issabel® goal is to incorporate all the communication alternatives, available at an enterprise level, into a unique solution..
Telephony was the traditional way that lead communications the last century, that’s why many Companies and users focus their requirements on their necessities to establish telephony communications on their organizations, and confuse unified communications “distros” with a telephone exchange system. Issabel®, not only provides telephony, it integrates other communication alternatives to make your organization environment more productive and efficient. .
Licensing in Issabel®
Issabel® is an open source entrepreneur tool relased under the license GPLv2. You are free to use it for commercial of personal purposes subject to the conditions as described under its License.
Issabel® doesn’t have a cost related with licensing or functionality.
All the Issabel® versions are full versions without limitation in its use or its features. Nor the addition of modules or users, in an Issabel® implementation have a cost involved to the integrator, organizations or enterprises in desire to use Issabel®.
We work every day of the year thinking how to improve your communications and designing new versions of Issabel®.
If you are interested in a personalized solution, please contact our comercial department: sales@avanzada7.com
O Elastix Não apenas fornece telefonia, mas também integra outros meios de comunicação para tornar seu ambiente de trabalho mais eficiente e produtivo. Todos os dias há novas maneiras de se comunicar, e a adição de recursos e funcionalidades deve ser constante.
Alguns dos recursos básicos do Elastix incluem:
Desktop IP phone Snom D785
The D785 phone belongs to the latest generation of Snom Ip advanced phones. It presents a large high-resolution color screen and a second screen for managing dynamic keys, as well as integrated bluetooth. A premium phone that includes all the necessary functionalities to satisfy the requirements of the most demanding users.
It incorporates a digital signal processor (DSP) and HD audio quality. In addition, its modern and elegant design is ideal for any environment and thanks to its second screen and its user interface is a very intuitive and easy to use terminal. Another advantage of its second screen is that it allows you to manage applications quickly and easily.
Patton SmartNode SBC SN5501
With your eSBC SmartNode 5500 you can connect through a router with a SIP trunk link or a hosted PBX service. The SN5500 supports up to 200 SIP to SIP calls (under ideal conditions), 16 which can be transcoding, for remote connectivity, branches and All-IP operator services.
The SN5500 series acts as a Session Border Controller, access router and QoS CPE, all in one device. You can also carry out the evaluation and monitoring of the network in the client's facilities using the PacketSmart agent to prevent, reduce and solve network and voice quality problems.
Why SmartNode?
Valid for this two applications:
Patton SmartNode SBC SN5501 4B / EUI
Centralita Panasonic KX-HTS32
La centralita Panasonic KX-HTS32 está diseñada para responder a las necesidades de las pequeñas oficinas y los despachos. Se puede utilizar no sólo para facilitar comunicaciones remotas, de video y de voz, sino también para crear sistemas asequibles de supervisión y seguridad.
Todas las funciones y características necesarias se integran en el hardware del dispositivo. Además, incluye un punto de acceso Wi-Fi para ordenadores y smartphones, gestión de llamadas, capacidad troncal SIP múltiple, funcionalidad para conferencias y opciones de buscapersonas.
Fone de ouvido Addcom ADD-55 binaural
Auscultadores Binaural com qualidade HD de banda larga e um design elegante e robusto para o seu escritório ou centro de atendimento.
Gateway Vega 60G - 4 BRI
The Vega 60G Gateway is a device designed to simplify the integration of an analog telephone or basic rate ISDN (BRI) devices to a VoIP network.
Each BRI interface can be configured independently as a network side or terminal side. The Vega 60G Media Gateway can, therefore, be used for IP-PBX connectivity to PSTN and SIP trunking offers. This configuration provides:
Gateway Vega 60G BRI
Telefone DECT Mitel 632d V2
O 632d V2 da Mitel é um terminal DECT ideal para aplicações externas ou industriais. Graças à sua tecnologia sem fio, é ideal para mercados verticais.
Além disso, é fácil de limpar e complementa os requisitos de higiene para o setor de saúde, pois tem uma chave para chamadas de emergência ideais para hospitais ou setores relacionados à segurança. Além disso, incorpora um sensor de alarme que pode ser usado como medida de segurança.
O Mitel 632d V2 está equipado com uma interface Bluetooth para auscultadores sem fios. É compatível com o padrão GAP e suporta o download de firmware para manter seus telefones atualizados.
Heaset ADDCOM Bluetooth ADD695: conectividade 3-em-1: Smartphone, softphone e deskphone
Os fones de ouvido permitem conectar todos os seus telefones a um fone de ouvido. Você pode alternar entre seu telefone de mesa, smartphone ou softphone para obter a melhor qualidade de som ADDCOM.
Os protetores de ouvido coloridos são intercambiáveis pelo usuário para poder identificar rapidamente os fones de ouvido de cada pessoa ou equipamento.
O ADD695 permite mover até 100m da estação base e fornecer até 8 horas contínuas de conversação. Sua tecnologia de cancelamento de ruído proporciona conversas claras durante as chamadas e sua proteção acústica protege os usuários contra ondas repentinas de rúido.
Grandstream GVC3210 - Video conferencing device
Ideal video conferencing device for companies thanks to its ease of use and power. A Grandstream product that supports high-level video resolutions with ultra-high-definition capability of up to 4K.
The GVC3210 allows the user to customize their communication solution according to their needs. In addition, it incorporates the Android operating system and offers full access to the Google Play Store offering the possibility of using any conference or communication application for Android.
Another feature to highlight is the incorporation of Noise-Shield technology that blocks background noise and maximizes audio quality.
Grandstream GDS3705 - Audio Portero IP
O gateway de áudio IP GDS3705 da Grandstream é ideal para usuários que procuram uma solução de controle de acesso sólida para instalações com monitoramento de áudio e segurança que podem ser implementados em ambientes de qualquer tamanho.
Este equipamento é composto por um sistema de intercomunicação que inclui dos microfones e um alto-falante HD com tecnologia avançzada AEC; isso permite oferecer uma função de intercomunicação. Ele também pode suportar chamadas SIP para telefones IP e incorpora um leitor de cartões RFID e teclado para acesso seguro.
Quanto ao acabamento, o GDS3705 vem equipado com uma liga de zinco, tornando-o impermeável e à prova de vandalismo
Gateway Grandstream ATA HT818
O Grandstraem ATA HT818 é um poderoso gateway para converter tecnologia analógica em VoIP. Especificamente, este modelo incorpora 8 portas FXS e um roteador Gitabit NAT integrado. Além disso, possui excepcional qualidade de voz em diversos ambientes de aplicativos, criptografia robusta com um único certificado de segurança por unidade, provisionamiento automático e excelente desempenho de rede, ideal para uso comercial.
Patton SmartNode SBC SN5501 8P / EUI
With your eSBC SmartNode 5500 you can connect through a router with a SIP trunk link or a hosted PBX service. The SN5500 supports up to 200 SIP to SIP calls (under ideal conditions), 8 which can be transcoding, for remote connectivity, branches and All-IP operator services.
Ponto de acesso WiFi 802.11ac Wave-2 com Tecnologia MU-MIMO 4x4:4 para Empresas
O Grandstream GWN7630 é um ponto de acesso Wi-Fi Wave-2 802.11ac de alto desempenho para pequenas e médias emrpesas, escritórios com vários andares, shopping centers e agências. Ele oferece a tecnologia MU-MIMO 4x4:4 de banda dupla e um design de antena sofisticado para desempenho máximo da rede e cobertura Wi-Fi expandida. PAra garantir a facilidade de instalação e operação, o GWN7630 usa um design de gerenciamento de rede distribuído sem controladores onde o controlador é integrado à interface de usuário da web do produto. O GWN7630 também é apoiado pelo GWN.
O GWN7630 é um hotspot Wi-Fi ideal para implementações de médio porte de redes sem fio com densidade de usuário média a alta.
Enterprise Appliance para voz, vídeo, dados e mobilidade
Dispositivo projetado para fornecer uma solução de comunicação centralizada. Série PBX IP da Issabel - O ISS Micro UCR combina as soluções que sua empresa precisa em um servidor fácil de usar.
Este PBX permite unificar várias tecnologias de comunicação, como voz, chamadas de vídeo, videoconferência, videovigilância, ferramentas de dados, o pções de mobilidade e gerenciamento de acesso a instalações em uma rede comum que pode ser gerenciada e / ou acessada remotamente graças ao Assistente de Configuração da GUI do Assistente Issabel.
A série ISS Micro segura e confiável oferece recursos de qualidade empresarial, sem taxas de licença, custos por função ou taxas recorrentes.
Também possui a função Survival Ready: com este recurso, 2 linhas e 2 extensões analógicas funcionam apesar de perda de energia.
Gateway Mediatrix C720 - 2 BRI
A série Mediatrix C7 são gateways e adaptadores (ATA) que podem incorporar portas FXS, FXO ou interfaces BRI. Ele oferece a solução de preços mais eficaz para filiais e PMEs, garantindo uma conexão simples e transparente a PSTN e terminais analógicos, como telefones, modems e máquinas de fax.
As plataformas são totalmente interoperáveis com os principais fabricantes de telefones IP, softswitches e serviços de nuvem em todo o mundo.
No caso deste modelo especifico, o C720 incorpora 2 BRI portas.
-- Operadores --
-- Integradores de sistemas --
Gateway VoIP Digital E1/T1/J1
A série GXW4500 inclui os gateways VoIP digitais E1 / T1, que possibilitam a integração de troncos RDIS e RPTC digitais com redes VoIP. Por meio da conexão da série GXW4500 com uma rede VoIP e um provedor de PBX ou E1 / T1 tradicional, as empresas podem aumentar drasticamente o número de troncos RPTC / RDIS integrados a sua rede VoIP.
Este modelo em particular oferece 1 seção E1/T1/J1 e suporta 30 chamadas simultâneas para satisfazer as necessidades de VoIP de grandes e médias empresas.
Telefone IP Yealink T57W - Eficiência e produtividade
O Yealink T57W é um telefone IP de mesa fácil de usar que ajudará o usuário a otimizar seu trabalho diário. Tem um ecrã táctil a cores de 7 '' 800x480 que pode ser ajustado para ajustar o ângulo de visão de acordo com as necessidades pessoais e o ambiente de trabalho.
O Yealink T57W IP Phone não apenas fornece o Bluetooth 4.2 integrado para a sincronização de telefones celulares e contatos Bluetooth, mas também o Wi-Fi integrado de banda dupla para conectividade Wi-Fi, permitindo que você acesse o 51G Wi-Fi conectividade com facilidade.
Além disso, se você quiser expandir seus horizontes para ambientes ocupados ou compartilhar um sistema de telefonia com sua pequena equipe adicionando vários aparelhos, simplesmente transforme seu telefone IP no telefone sem fio com fio por meio da tecnologia DECT. Além disso, o Telefone IP pode funcionar em conjunto com o Yealink VC Desktop para compartilhar conteúdo do seu laptop, tornando a colaboração muito mais fácil do que antes. * O firmware relacionado do Yealink VCD é a versão 28, que será lançada em março de 2019.
Ele também inclui uma porta USB 2.0 para gravar chamadas, conectar o fone de ouvido a um conector USB ou incorporar até três teclados de expansão Yealink EXP50 (não incluido).
Telefone IP Yealink T54W - Eficiência e produtividade
O Yealink T54W é um telefone IP de mesa fácil de usar que ajudará o usuário a otimizar seu trabalho diário. Ele possui uma tela gráfica LCD de 4.3'' que pode ser ajustada para ajustar o ângulo de visão de acordo com as necessidades pessoais e o ambiente de trabalho.
O Yealink T54-W IP Phone não apenas fornece o Bluetooth 4.2 integrado para a sincronização de telefones celulares e contatos Bluetooth, mas também o Wi-Fi integrado de banda dupla para conectividade Wi-Fi, permitindo que você acesse o 51G Wi-Fi conectividade com facilidade.
Telefone IP Yealink T53W - Eficiência e produtividade
O Yealink T53W é um telefone IP de mesa fácil de usar que ajudará o usuário a otimizar seu trabalho diário. Ele possui uma tela gráfica LCD de 3.7'' que pode ser ajustada para ajustar o ângulo de visão de acordo com as necessidades pessoais e o ambiente de trabalho.
O Yealink T53-W IP Phone não apenas fornece o Bluetooth 4.2 integrado para a sincronização de telefones celulares e contatos Bluetooth, mas também o Wi-Fi integrado de banda dupla para conectividade Wi-Fi, permitindo que você acesse o 51G Wi-Fi conectividade com facilidade.
Videofone Yealink VP59
O videofone Yealink VP59 foi projetado para executivos e teletrabalhadores que buscam um equilibrio perfeito entre simplicidade e sofisticação, permitindo comunicações de alta qualidade para seus usuários.
Incorpora o sistema operacional Andoid 7.1 e incorpora tela LCD colorida multi-touch; também permite a instalação de aplicativos de terceiros (como Skype, SFB...) para personalização do terminal, bem como dos da Doorphone.
Usando o Dongle DD10K permite converter o terminal em um telefone sem fio e permite sincronizar até 4 aparelhos DECT.
Além disso, incorpora WiFi 802.11 a/b/g/n/ac bem como Bluetooth 4.2. Essa combinação facilita a transmissão de dados mais rápida e convenientemente; por outro lado, incorpora uma saída HDMI para a transmissão de dados de alta definição.
Teléfono HD DECT Inalámbrico para Movilidad
O DP730 é um telefone com tecnologia DECT que permite aos usuários mobilizar sua rede VoIP em qualquer empresa e / ou ambiente residencial. É compatível com as estações base VoIP DECT DP750 e DP752 da Grandstream e ofereceuma combinação de mobilidade e excelentes qualidades técnicas de telefonia.
Cada estação base permite até cinco aparelhos DP730, e os DP730 têm alcance de até 400 metros ao ar livre (com o DP752) e 50 metros em local fechado, com 40 horas de conversa e 500 horas em espera. Ele apresenta um conjunto de rescursos de telefonia avançados, como suporte para até 10 contas SIP por aparelho, áudio Full HD, tela colorida de 2.4 polegadas, conector para fones de 3.5mm, PTT, viva-voz e muito mais. Quando emparelhado com as estações base DECT da Grandstream, o DP730 é um aparelho de ponta, que oferece uma solução DECT sem fio muito eficiente para qualquer usuário empresarial ou residencial.
Estação base VoIP DECT de longo alcance
O DP752 é uma avançada estação base VoIP DECT para emparelhamento com até 5 aparelhos DECT da série DP da Grandstream para proporcionar mobilidade a usuários empresariais e residenciais.
Ele suporta um alcance em abientes externos de até 400 metros com aparelhos DP730 ou de até 350 metros com aparelhos DP722 / DP720, além de alcance em ambientes internos de até 50 metros, proporcionando aos usuários liberdade para se movimentar em casa ou no escritório.
Essa estação base VoIP DECT suporta até 10 contas SIP e 5 chamadas simultâneas permitindo também audioconferência de 3 vias, áudio Full HD e PoE integrado. Uma conta SIP compartilhada em todos os aparelho acrescenta recursos unificados integrados para permitir que os usuários respondam a todas as chamadas em tempo real, independente de sua localização.
O DP752 é com patível com diversos métodos de provisionamento automatizado e a segurança da criptografia TLS/SRTP/HTTPS. Quando emparelhado com os aparelhos DP720, DP722 ou DP730 da Grandstream, o DP752 oferece uma solução DECT sem fio muito eficiente para qualquer usuário empresarial ou residencial.
Yealink Wireless DECT Conference Phone
O telefone de conferência sem fio Yealink CP930W DECT foi projetado para redes sem fio e ambientes de comunicação, como salas de reuniões pequenas e médias. Além disso, graças ao fato de ser livre de cabos, permite maior amplitude e conforto em se espaço de trabalho.
Outra das características marcantes deste terminal é que ele permite que você sincronize com seu celular / smartphone ou PC / tablet via Bluetooth e porta Micro-B USB, permitindo a sincronização com outros dispositivos. Isso permite que sua experiência seja muito mais flexível, evitando limitações de eespaço e abandonando a chamada, podendo enviá-la para outros dispositivos disponíveis dentro da sua cobertura DECT.
OpenVox Gateway Module 4 x E1/T1 PRI 2004
O módulo 2004 é uma solução VoIP baseada no Asterisk para operadores e call center.s Esse tipo de gateway conecta o sistema de telefonia tradicional às redes IP e integra o PBX VoIP com a PSTN à perfeição. Ele incorpora uma GUI amigável para que os usuários possam configurar facilmente sua conexão.
Este modelo incorpora 4 x PRI E1/T1 e suporta até 120 chamadas simultâneas.
OpenVox UC120 PBX Server - 1FXO, 1FXS, 1GSM
É uma nova geração de equipamentos de comunicações unificadas que combina voz e dados. É compacto e leve, e fornece as seguintes interfaces de voz: FXS, FXO, GSM / LTE (o último em desenvolvimento) e Φ 3.5 interface de áudio.
Suporta múltiplas plataformas e terminais de serviço e pode conectarse facilmente a redes VoIP, redes telefônicas tradicionais (PSTN) e redes móveis (PLMN) e fornecer várias soluções de comunicações convergentes.
Por outro lado, o UC120 pode conectar-se a telefones tradicionais, faxes e PBXs analógicos via interface de voz padrão; o IC120 usa um protocolo SIP padrão e é compatível com a maioria dos Ip PBX, soft switches e plataformas de rede baseadas em SIP. Ao mesmo tempo, o UC120 suporta 4 bandas GSM para atender aos requisitos das redes globais de comunicações móveis.
Uma das características marcantes deste produto é que ele combina uma porta de áudio de 3.5 que pode ser conectada diretamente aos fones de ouvido de um computador para fazer chamadas de voz.
Este produto pode ser usado como um produto de comunicação pessoal, mas também pode ser usado como um produto de comunicação centralizado para PMEs, que fornece acesso à Internet de alta velocidade, comunicação de voz empresarial e transmissão e recepção de mensagens curtas.
Gateway Mediatrix C740 - 4 BRI
No caso deste modelo especifico, o C740 incorpora 4BRI portas.
Gateway Mediatrix C720h - 1BRI
No caso deste modelo especifico, o C720h incorpora 1BRI porta.
Grandstream GRP2614
O telefone Grandstream GRP2614 é um telefone IP de 4 linhas operado pela operadora, projetado com provisionamento automático para fácil operação e energia. Inclui 40 chaves multiuso virtuais (VPKs), WiFi integrado, suporte a Bluetooth, duas portas Gigabit ou uma tela LCD colorida com faceplates intercambiáveis para permitir a fácil personalização do logotipo, além de outras funções.
Por outro lado, esta série de GRP inclui recursos de segurança de nível de operadora para fornecer segurança em nível corporativo, como inicialização segura, imagens de firmware duplas e armazenamento de datos criptografados.
Este telefone também tem o suporte do serviço de nuvem chamado Grandstream Device Management System (GDMS).
Grandstream GRP2613
O telefone Grandstream GRP2613 é um telefone IP de 3 linhas para operadoras projetado com provisionamento automático poderoso e fácil de usar. Inclui 24 teclas multiuso virtuais (VPKs), 2 portas Gigabit, uma tela LCD colorida com placas frontais intercambiáveis para permitir fácil personalização do logotipo, além de outras funções.
Por outro lado, esta série de GRP inclui recursos de segurança de nível de operadora para fornecer segurança em nível corporativo, como inicialização segura, imagens de firmware duplas e armazenamento de dados criptografados.
Grandstream GRP2612P
O telefone Grandstream GRP2612P é um telefone IP de 2 linhas de nível de operadora projetado com provisionamento automático poderoso e fácil de usar. Inclui 16 teclas multiuso virtuais (VPKs), uma tela LCD colorida com faceplates intercambiáveis para permitir a fácil personalização do logotipo, além de outras funções.
Grandstream GRP2612
O telefone Grandstream GRP2612 é um telefone IP de 2 linhas de nível de operadora projetado com provisionamento automático poderoso e fácil de usar. Inclui 16 teclas multiuso virtuais (VPKs), uma tela LCD colorida com faceplates intercambiáveis para permitir a fácil personalização do logotipo, além de outras funções.
Virtual SmartNode vSN por Patton
Este SmartNode Virtual vSN da Patton oferece os recursos que você precisa para aplicativos avançados de rede de voz. Além disso, elimina as limitações de hardware necessárias para o desempenho do aplicativo.
Ele fornece o conjunto completo de funções necessárias para você, como empresa ou provedor de serviços, que procura virtualizar sua infraestrutura.
A prevenção de DoS, a confiança entre colegas, a prevenção da fraude de chamadas, a qualidade do serviço do firewall do estado, a avaliação e monitoramento da qualidade de voz e muito mais fazem parte do Patton SmartNode virtualizado, bem como Roteamento IP, incluindo as funções de rede IPv4 e IPv6.
Os SmartNodes são os CPEs principais e testados no campo que executa o sistema operacional Patton. O mesmo software rico em recursos usado em dispositivos de construção específicos de hardware também pode ser usado em ambientes virtuais para um alto nível de escalabilidade e disponibilidade de serviço.
O Virtual SmartNode foi projetado para ser executado em redes de negócios virtualizadas, data centers de provedores de serviços de telefonia, infraestrutura de provedores de nuvem, etc. Ele pode atender às diferentes necessidades de capacidade de chamada dos clientes, dimensionando e reduzindo as licenças necessárias no SmartNode virtual usando uma licença de modelo flutuante por meio da nuvem Patton. A nuvem Patton distribui automaticamente as licenças obtidas para onde elas são ncessárias (vSN e outros dispositivos Patton).
Isso permite escalar facilmente serviços SBC up/downstream em SmartNodes virtuais, mas também em dispositivos SmartNode normais.
SmartNode Virtual como SBC no cenário de entroncamento SIP
- Detalhes da aplicação
- Beneficios:
Mais informações: https://www.patton.com/session-border-controller/vsn/
Alto-falante SIP GSC3505 Megaphone Unidirecional
O GSC3505 é um alto-falante SIP megafone unidirecional que permite que escritórios, escolas, hospitais, edifícios e outros estabelecimientos desenvolvam soluções poderosas de transmissão de endereços públicos que expandem a segurança e a comunicação. O GSC3505 oferece funcionalidade de áudio HD graças ao alto-falante de 8W HD. Também é compatível com dispositivos Bluetooth, lista branca e lista negra integrados para bloquear facilmente chamadas indesejadas, paging SIP e multicast e Wi-Fi integrado de banda dupla.
Alto-falante e microfone de intercomunicação GSC3510 SIP
O GSC3510 é um alto-falante e microfone de intercomunicação SIP muito forte que oferece recursos de voz graças ao seu alto-falante de 8W HD de alta fidelidade e seus 3 microfones direcionais.
Também é compatível com uma ampla variedade de periféricos, incluindo dispositivos Bluetooth, lista branca integrada e lista negra, Wi-Fi dual band integrado e cancelamento de eco acústico avançado.
Gateway VoIP Digital GXW4504
Este modelo em particular oferece 4 seção E1/T1/J1 e suporta 120 chamadas simultâneas para satisfazer as necessidades de VoIP de grandes e médias empresas.
Gateway VoIP Digital GXW4502
Este modelo em particular oferece 2 seção E1/T1/J1 e suporta 60 chamadas simultâneas para satisfazer as necessidades de VoIP de grandes e médias empresas.
Sangoma S206 - Telefone IP Entry Level
Os telefones S206 são telefones Sangoma IP projetados para funcionar com o FreePBX e o PBXact. O Sangoma S206 é um telefone cheio de recursos com 2 contas SIP a um preço muito competitivo.
Eles podem detectar rapidamente o FreePBX / PBXact assim que se conectam a uma conexão com a Internet e são automaticamente provisionados em poucos segundos. Eles também têm jPoE, portando não requerem cabos de alimentação.
Eles incorporam o 'EndPoint Manager' para que, quando o telefone for usado, o transporte dentro do FreePBX / PBXact seja ativado automaticamente. Isso permite aos usuários um controle global do terminal fácil e imediatamente, programando as teclas, extensões, upload de imagens, baixar o novo firmware e muito mais.
Gateway Mediatrix C725 - 2 BRI + 2 FXS
No caso deste modelo especifico, o C725 incorpora 2 BRI + 2 FXS portas.
OpenVox Gateway Module 2 x E1/T1 PRI 2002
O módulo 2002 é uma solução VoIP baseada no Asterisk para operadores e call center.s Esse tipo de gateway conecta o sistema de telefonia tradicional às redes IP e integra o PBX VoIP com a PSTN à perfeição. Ele incorpora uma GUI amigável para que os usuários possam configurar facilmente sua conexão.
Este modelo incorpora 2 x PRI E1/T1 e suporta até 60 chamadas simultâneas.
OpenVox Gateway Module 1 x E1/T1 PRI 2001
O módulo 2001 é uma solução VoIP baseada no Asterisk para operadores e call center.s Esse tipo de gateway conecta o sistema de telefonia tradicional às redes IP e integra o PBX VoIP com a PSTN à perfeição. Ele incorpora uma GUI amigável para que os usuários possam configurar facilmente sua conexão.
Este modelo incorpora 1 x PRI E1/T1 e suporta até 30 chamadas simultâneas.
OpenVox GWM420 4 x Módulos de Gateway LTE
O módulo Gateway da série OpenVox inclui o GWM420G e permite que os gateways das séries VS- GW1200 / 1600 / 2120 suportem a conexão LTE com disposisitvos VoIP. Cada módulo oferece 4 canais LTE.
Eles podem oferecer-Ihe um excelente serviço de voz HD com vários codecs, incluindo G.711u, G.711A, GSM, G.722, G.723, G.726, G.729 e também serviço SMS flexível com várias APIs SMS.
100% compatível com as plataformas Asterisk, FreePBX, FreeSwitch e VOS, VoIP; Também ajudará os usuários a reduzir os custos de telecomunicações e comunicação.
Módulo de expanção - Snom D7 White
Est eteclado é uma expansão do seu telefone IP de desktop que permitirá expandir a funcionalidade. Entre eles, este telefone adiciona 18 teclas programáveis LED multicoloridas para executar qualquier uma das funções disponíveis nos telefones Snom, tais como: linha, indicador de presença, etc.
A implementação deste teclado no seu sistema de comunicação deste teclado no seu sistema de comunicação é muito fácil, pois você só precisa conectar seu cabo USB. Além disso, você pode conectar até três módulos de expansão D7 em série, e você pode adicionar até 54 teclas de função.
Telefone IP Snom D715
O telefone Snom D715 é um telefone de mesa IP que garante conexões de alta velocidade graças ao seu switch Gigabit Ethernet integrado. Por outro lado, oferece qualidade de som HD e uma porta USB dando ao terminal maior flexibilidade e melhores recursos para uso com fones de ouvido USB com o módulo adicional D7 ou modo Wi-Fi.
Graças à sua certificação pré-instalada, este modelo é compatível com a última geração de padrões de segurança VoIP. Além de oferecer funções avaçadas de gerenciamento e provisionamento remoto, permite sua instalação em grande escala, tornando-se a solução perfeita para todos os provedores de sistemas VoIP.
Telefone IP Snom D785 Branco
O telefone D785 pertence à última geração de telefones Snom. Especificamente, este modelo D785 incorpora uma tela colorida de alta resolução e outra segunda tela para gerenciar teclas dinâmicas, bem como bluetooth integrado.
Além disso, entre outros recursos, incorpora um processador de sinal digital (DSP) e qualidade de áudio HD. Além disso, seu design moderno e elegante é ideal para qualquer ambiente e graças à sua segunda tela e sua interface de usuário é um terminal muito intuitivo e fácil de usar. Outra vantagem desta segunda tela é que ela permite gerenciar aplicativos de maneira rápida e fácil.
Desktop IP Phone D375 Branco
Um telefone de mesa ideal para profissionais que buscam otimizar sua experiência e reduzir o tempo na execução de determinadas tarefas. Graças à personalização de sua interface, o usuário pode pesquisar dados importantes graças a acessos rápidos, como a busca de listas de chamadas. Além disso, permite a personalização de sua interface, uma vez que incorpora a opção de personalizar determinados dados, como nome da empresa, logotipo, hora..., esses dados aparecerão em sua interface integrada na comunicação corporativa da empresa.
Desktop IP Phone D375
Auscultadores Snom DECT A170
O fone de ouvido sem fio Snom A170 é compatível com telefones de mesa a Snom apresenta um design muito cuidadoso e funcional, pois permite três estilos de uso: sobre a cabeça, pescoço e sobre a orelha.
Além disso, incorpora LED de estado polivalente para visualizar o status da chamada, carga e status da bateria; e permite controlar chamadas do próprio aparelho.
SERVIÇO DE MANUTENÇÃO DE VoIP
Desde o departamento de manutenção da Avanzada 7, uma manutenção diária e constante das máquinas que os clientes possuem e possuem este serviço contratado. Os serviços que são realizados estão incluídos em 3 grupos de ações:
MONITORAMENTO
Um painel de monitoramento está disponível nad instalações da Avanzada 7, onde todas e cada uma das instalações são constantemente monitoradas para que nossos técnicos possam avançar na solução dos possíveis incidentes que possam localizar em seus sistemas de telefonia Asterisk
Da mesma forma, alertas são recebidos via e-mail dos diferentes sistemas que nossos técnicos recebem inmediatamente, para tomar as ações e verificações pertinentes antes que o dano ocorra em seus sistemas.
Da mesma forma, tanto por telefone como por email, eles podem notificá-lo dentro do cronograma contratado e nossos técnicos Ihe darão uma resposta imediata e a resolução do incidente no menos tempo possível, geramente dentro de alguns minutos após o alerta recebido.
Existe una plataforma de tickets, onde você pode acompanhar tanto o cliente quanto a equipe do Avanzada 7 sl dos incidentes reportados, bem como os tempos de resposta dos mesmos.
CÓPIAS DE SEGURANÇA
Para Avanzada 7 sl, garantir a disponibilidade na frente de catástrofes dos sitemas de manutenção. Para aquelas plataformas que não têm alta disponibilidade, assim como aquelas que o fazem, o Avanzada 7sl faz cópias de backup diárias da configuração do Asterisk, bem como cópias periódicas de bancos de dados, logs, locuções ou configuração básica do sistema para pode ser recuperado em caso de séria perda dele (ferragem ou catástrofe natural).
Este processo é automático e monitorado, você não deve se preocupar com nada.
SEGURANÇA:
Para o Avanzada 7 sl, a segurança de sua plataforma é muito importante, por isso sempre são feitas atualizações nos pacotes que comprometem a segurança ou corrigem falhas graves e que podem ser afetadas, bem como o Asterisk, mantendo as últimas versões estáveis e corrigindo erros e erros. Bugs importantes em termos de funcionalidade e segurança.
Da mesma forma, todos os nossos sistemas mantidos possuem chaves de acesso de firewall e RSA, e somente a equipe do Avanzada 7 tem acesso remoto ao seu sistem, independentemente de você ter ou não um firewall em sua infraestrutura.
RECONFIGURAÇÃO E MUDANÇAS
Dentro da manutenção você pode solicitar pequenas alterações de configuração, extensão de ramais, cancelamento, modificação de saída de chamadas (dialplan), bem como problemas detalhados no contrato que você possui.
ACTUALIZAÇOES
Todos e cada um dos sistemas que possuem Avanzada 7sl receber atualizações periódicas do sistema de telefonia Asterisk e dos pacotes do sistema operacional. É para Avanzada 7 uma prioridade que os sistemas sejam atualizados em face de problemas e violações de segurança, bem como em termos de estabilidade.
Os sistemas são atualizados para as versões mais recentes do Asterisk disponíveis que foram aprovadas para produção pelo Avanzada 7, bem como correções que aparecem para erros na segurança e estabilidade dos sistemas. As instalações são atualizadas para as versões mais recentes do sistema operacional LINUX em geral, para buscar a estabilidade e integridade das mesmas.
RELATÓRIOS DE MANUTENÇÃO MENSAL
Todo mês um relatório é enviado ao cliente para que ele possa avaliar o status de seu sistema para possíveis atualizações de hardware ou manterse informado que foram executadas no sistema. Este relatório inclui as seguintes informações com valores máximo, mínimo e médio:
Projetos de desenvolvimento e turnkey
Nossa experiência em projetos de engenharia abrange setores muito diversos além da telefonia que é o núcleo de nossa atividade.
Participamos de projetos de inteligência artificial aplicados ao processamento digital de imagens, sistemas M2M para telegestão em aplicações logísticas, projeto de software de baixo nível para arquiteturas de hardware de propósito específico ou desenvolvimento web no setor de call center.
Se você deseja entrar em contato com uma engenharia versátil e multidisciplinar para planejar, projetar e executar um projeto, não hesite em nos contatar.
Auditoria VoIP
O Avanzada 7 realiza auditorias profissionais e específicas em áreas-chave, como a segurança de uma infra-estrtura de comunicações, a capacidade de uma rede de transporte de voz e vídeo ou estudios teóricos de cobertura de rádio para implantações DECT (que realizamos gratuitamente).
Integração Skype for Business - Asterisk
Devido à nossa ampla experiência com o Asterisk, podemos ajudá-lo a integrar seu PBX Asterisk para que os clientes do Skype possam chamar as extensões do Asterisk e vice-versa.
Essa integração pode ser feita se você tiver uma solução local do Skype for Business ou se tiver um Skype for Business on-line e quiser que a conexão com a PSTN seja feita por meio do Asterisk.
Migração para as equipes da Microsoft Teams
O Skype for Business na nuvem ingressará no Microsoft Teams. O objetivo é ter um único centro de trabalho é ter um único centro de trabalho em equipe com vídeo e voz totalmente integrados. O Microsoft Teams oferece uma infraestrutura de nuvem moderna que permite que você obtenha o máximo dos ativos de inteligência artificial.
O Microsoft Teams, que trabalha com a nova infraestrutura de back-end do Skype, foi projetado para a nuvem com uma arquitetura de microserviço altamente escalável que também é eficiente em termos de consumo de largura de banda, fornece uma telemetria mais robusta e permite executar tarefas de manutenção e atualizações com um nível mínimo de interrupções. Esta infra-estrutura moderna permite fácil acesso ao Microsoft Cognitive Services, que possui funções de transcrição, tradução, reconhecimento de fala e aprendizado automático.
No momento, não há planos para eliminar o Skype for Business das assinaturas do Office 365 e não há data final para o suporte, mas a Microsoft concentrará o investimento de melhorias no Microsoft Teams para que o grupos dos principais seja incorporado pouco a pouco. Skype Empresarial para funcionalidades do Office 365.
Você também pode implementar um piloto de equipes de pequena escala, no qual é possível validar a interoperabilidade com o Skype for Business. Para obter resultados mais realistas, o programa piloto deve reproduzir exatamente como os usuários estão se comunicando e colaborando no momento e verificar qual seria o cénario de implementação ideal como o Skype for Business e as equipes da Microsoft. Se a organização está pensando em executar o Skype for Business e o Microsoft Teams em paralelo ou alternar para o Microsoft Teams posteriormente, um piloto pode ajudar a identificar o caminho que deve ser seguido para cada organização.
Gateway Vega 60G - 8 FXO
Konftel 3000ipx
O Konftel 300IPx, juntamente com o aplicativo Konftel Unite, traz simplicidade ao seu sistema de conferência. Com o aplicativo móvel, você só precisa de um clique para iniciar a reunião ou participar dela.
Você pode ligar para sua lista telefônica pessoal do seu celular e controlar as funções do telefone durante a reunião.
Sua tecnología patentada OmniSound produz um som HD nítido e natural. Como um sistema de conferência IP, o Konftel 300IPx é adequado tanto para pequenas salas de reunião como para grandes auditórios, graças à sua capacidade de conectar microfones adicionais, alto-falantes e fones de ouvido sem fio.
Além disso, graças à sua conexão USB pode ser usado com todos os aplicativos e colaborar através da Internet: Skype Empresarial, Cisco Webex, Google Hangouts...
Telefone IP Snom D385
O telefone Snom D385 IP incorpora uma tela TFT de 4.3'' de alta resoulção, mas também incorpora uma tela dedicada a monitorar extenções. Possui 48 chaves BLF (12 físicas) e uma interface gráfica aprimorada que oferece ao usuário uma experiência mais otimizada para o dia a dia na empresa.
Por outro lado, incorpora portas switch PoE IEEE 802.3af, Class3 e duas Gigabit Ethernet (RJ45).
OpenVox GWM420 4 x Módulos de Gateway GSM
O módulo Gateway da série OpenVox inclui o GWM420G e permite que os gateways das séries VS- GW1200 / 1600 / 2120 suportem a conexão GSM com disposisitvos VoIP. Cada módulo oferece 4 canais GSM.
Portas FXS do módulo 8 do OpenVox
Módulo de interface FXS para gateways analógicos OpenVox VS-GW 1600 / 2120 V2
Portas FXO do módulo 8 do OpenVox
Módulo de interface FXO para gateways analógicos OpenVox VS-GW 1600 / 2120 V2
OpenVox GW2120 V2
O Gateway VS-GW2120 V2 é a principal solução de gateway baseada no Asterisk para PMEs e SOHOs. Com uma interface gráfica de usuário fácil de usar e um design modular exclusivo, os usuários podem configurar facilmente seu gateway personalizado, graças ao qual os usuários podem adicionar ou remover módulos para a expansão de seu dispositivo. Você também pode concluir o desenvolvimento secundário por meio do AMI (Asterisk Management Interface).
Também suporta o envio e recebimento de SMS e envio de grupos e SMS para e-mail. Este modelo em particular usa o protocolo SIP padrão e é compatível com a plataforma líder IMS / NGN, IP PBX e servidores SIP e é compatível com a maioria das plataformas operacionais VoIP, como Asterisk, FreeSwitch, Broadsoft.
OpenVox GW1600 V2
O Gateway VS-GW1600 V2 é a principal solução de gateway baseada no Asterisk para PMEs e SOHOs. Com uma interface gráfica de usuário fácil de usar e um design modular exclusivo, os usuários podem configurar facilmente seu gateway personalizado, graças ao qual os usuários podem adicionar ou remover módulos para a expansão de seu dispositivo. Você também pode concluir o desenvolvimento secundário por meio do AMI (Asterisk Management Interface).
Grandstream RFID
Leitor de cartões para o video porteiro Grandstream GDS3710
Sangoma combo DECT DC201
O combo Sangoma DC201 é um sistema de telefone DECT projetado especificamente para sistemas de telefonia com FreePBX e PBXact, o pacote DC201 oferece às pequenas e médias empresas um sistema DECT sem fio de alta qualidade que integra seu IP-PBX.
Este combo DC201 suporta até 20 usuários, o que dá ao usuário liberdade e eficiência enquanto se move usando tecnologia wireless. Por outro lado, outra das vantagens adicionais é o provisionamento automático com os telefones FreePBX e PBXact da Sangoma; usando a ferramenta EndPoint Manager, já incorporada, que permite o autoprovisionamento do restante dos hansets.
O Sangoma DC201 oferece aos funcionários a capacidade de transportar sua extensão enquanto se deslocam pelo escritório, tornando-o perfeito para:
Este pacote inclui:
Telefone Grandstream WiFi WP820
O telefone WiFi WP820 é ideal para uma ampla variedade de ambientes de negócios e aplicações, como logística, serviços de saúde, segurança...É um telefone cuja principal demanda é o suporte WiFi 802.11a/b/g/dual band integrado, design avançado de antena e roaming e suporte a Bluetooth para conexão com aparelhos auditivos e dispositivos móveis. No entanto, ele possui muitos outros recursos que o tornam um terminal muito completo e ideal para empresas.
CloudToWay
O CloudToWay é uma plataforma na nuvem que permite o gerenciamento remoto dos dispositivos da rede LAN de uma empresa ou Administração; dessa forma, o CloudToWay permite gerenciar proativamente esses dispositivos de rede e permite que o usuário fique constantemente informado sobre todos os eventos que ocorrem na rede da sua empresa, graças ao seu sistema de alarme e rastreamento.
O CloudToWay também é uma plataforma flexível porque se baseia em uma plataforma de software aberta e é compatível com qualquer dispositivo, o que nos permite adaptar o ambiente a qualquer fabricante.
Yealink W53H
O telefone Yealink W53H é um telefone sem fio projetado para empresas que exigem qualidade e alta compatibilidade. Os usuários podem aproveitar a liberdade de movimento sem dispensar excelentes recursos profissionais.
Apresenta tela TFT colorida 1.8'' na qual destacamos sua aparência moderna e qualidade de som HD.
O W53P pode ser sincronizado com até 8 monofones W53H.
Terminal Yealink DECT W53P
O telefone sem fio SIP de alto desempenho é a solução ideal para pequenas e médias empresas. O W53P pode sincronizar até um total de 8 headsets W53H, permitindo-Ihe desfrutar de excelente mobilidade e flexibilidade.
Por outro lado, para fornecer um melhor desempenho, também acelera o seu login e conexão de sinal e reduz o tempo de inatividade.
Ao incorporar o codec OPUS, ele oferece excelente qualidade de áudio profissional em alta largura de banda e condições de rede ruins, em comparação com outros codecs de áudio.
Migração do Skype for Business
Oferecemos o serviço de migração do Skype for Business local para o Skype for Business on-line.
O Skype for Business in-line está incluído no Office 365.
A mudança para a nuvem requer planejamento, implementação, implantação e migração. As vantagens de ter o Skype for Business Online são principalmente o fato de que o cliente não precisa cuidar da muntenção de máquinas físicas e pode ter as inovações sem ter que planejar e executar os administradores de TI.
Para este proceso seguimos la metodología SOF de Microsoft
Grandstream HT801
The Grandstream Analog Telephone Adapter, HT801, allows users to create a high-quality, easily-managed IP telephony solution, ideal for residential and office environments. This ATA is ideal for individual use because its ultra-compact size, voice quality and advanced size, voice quality and advanced VoIP features (such as security protection and automatic provisioning) allow the user to take full advantage of their analog phones.
Gigaset SL750H PRO
The Gigaset DECT phone, SL750H PRO, is a comfortable and easy-to-use phone packed with the professional features of Gigaset pro that can be adapted to specific needs. It works with N720 IP and single-cell N510 IP Gigaset pro systems and can also be connected to other base stations compatible with the GAP protocol, in single or multiple cell environments.
Its 2.4'' color screen is resistant to scratches and disinfectants thanks to the fact that it is manufactured with a high quality polymer. Thanks to its quality of voice and hands-free, the user perceives the sound with clarity and sound quality HDSP / CAT-IQ 2.0.
Panasonic KX - HDV230 Black
The IP phone Panasonic KX - HDV230 offers the ideal solution for any business environment. Thanks to its simple installation, reduced maintenance and functionality, the KX - HDV230 offers an excellent communication solution for companies of any size.
To deliver crip communications, the Panasonic KX - HDV230 integrates a combination of HD audio features, such as full duplex, full acoustic duplex, acoustic echo cancellation and hardware and software packet loss concealment. In this way, this IP solution guarantees the highest voice quality and performance associated with broadband communications.
Yealink YHS33 Headset Compatible with any Yealink phone
The Yealink YHS33 headset is an upgrade from the previous YHS32 model. This headset is compatible with the full range of Yealink of IP Phones.
It features a slim, lightweight design and pads with a soft, comfortable bio-mimetic material that allows you to hold conversations for long periods of time reducing hearing fatigue. It also features noise cancellation by filtering background noise to ensure perfect voice transmission between the caller and the listener.
Touch IP Phone - Digium D80
Digium presents this revolutionary IP Phone with ' inch HD display with BLF and dual 10/100/1000 Mbps Gigabit Network port. In addition, the D80 is specially designed for users who want a stlylish, elegant and quality terminal.
The D80 is designed exclusively for use with Asterisk and Switchvox and incorporates plug-and-play provisioning. All models include HD Voice and plug-and-play display and can also be optimized with the integration of advanced applications such as voice mail, call log, contacts, phone status etc.
Yealink IP Phone T48S
The Yealink T48S IP Phone offers a faster and more responsive interface than the T48G and offers better overall performance. This line offers the same look and feel as the T4 line but with notable improvements that favor its interoperability and collaboration. As a novelty, the F48S, like all other phones in this range, incorporates Optima HD Voice Technology and Opus broadband codecs.
Furthermore, it works with set with Bluetooth USB Dongle and Wi-Fi Dongle:
Yealink IP Phone T46S
The Yealink T46S IP Phone offers a faster and more responsive interface than the G46G and offers better overall performance. This line offers the same look and feel as the T4 line but with notable improvements that favor its interoperability and collaboration. As a novelty, the T46S, like all other phones in this range, incorporates Optima HD Voice technology and Opus broadband codec.
Unlike the T41S or T42S, this model features a high resolution TFT color display.
Yealink IP Phone T42S - Opus
The Yealink T42S IP Phone offers a faster and more responsive interface than the T42G and offers better overall performance. This line offers the same look and feel as the T4 line but with notable improvements that favor its interoperability and collaboration. As a novelty, the T42S, like all other phones in this range, incorporates Optima HD Voice Technology and Opus broadband codec.
Furthermore, this model incorproates a new USB port for future Bluetooth, Wi-Fi and USB recording functions.
Yealink IP Phone T41S - Opus
The Yealink T41S IP Phone offers quality features that enrich the user experience with its fast interface. This line offers the same look and feel as the T4 line but with notable improvements that favor its interoperability and collaboration. As a novelty, the T41S, like all other phones in this range, incorporates Optima HD Voice technology and Opus broadband codec.
Furthermore, this model incorproates a new USB port for future Bluetooth, Wi-Fi, USB recording functions and future expandability options.
WiFi Access Point - Grandstream GWN7610
The Grandstream GWN7610 is a high performance 802.11ac wireless access point for small and medium businesses, malls, offices, etc. A device that allows you to manage your own wireless network independently without using separate hardware / software controller and without a single point of failure. In addition, you can connect it with external routers and the Grandstream GWN series.
It also offers 3x3 technology MIMO, and an antenna of maximun network performance and expanded coverage of WiFi coverage. In addition, for an use more easily, your controller is built into the product's web-based user interface.
Kit de encastre para GDS3710
Yealink T52S | Color screen and OPUS codec
The Yealink T52S desktop phone is specially designed for professionals and executives; In addition to offering color screen presents a new look with an ergonomic design, comfortable to use, and with a very intuitive user interface.
In addition, it offers HD sound quality thanks to its Optima HD technology and the addition of the Opus Codec offering a crystal clear audio experience. With Gigabit, Bluetooth and USB 2.0 port, the Yealkink T52S becomes a safe bet for professionals that are looking for a quality terminal with higher performance.
Teclado de expansão HDV20
Console de expansão com 20 teclas / 40 contatos para HDV230 / 330 / 430. Incorpora LEDs de status luminoso.
Yealink T40G
The Yealink T40G is an IP Phone that provides secure and flexible provisioning by incorporating standard encryption protocols for users to perform both internal and remote updates.
This T40G also incorporates Gigabit Ethernet in order to facilitate the handling of calls and, besides being easy to use, it BLF, SCA, call transfer, etc. A complete phone that comes equipped with PoE and presents an excellent quality of sound and image.
Android videophone - Yealink T56AV
The Yealink T56A desktop IP Phone is a simple-to-use phone that provides an enriched HD audio experience ideal for business professionals. It is based on the Android 5.1.1 operating system and has a fixed seven-inch multipoint touch screen, integrated Wi-Fi and Bluetooth 4.0 + EDR, and is coupled with a built-in web browser, calendar, etc.
In addition, it incorporates other functionalities like calendar, recorder and the possibility of installing applications of third parties to personalize to 100% this professional telephone.
A multifunctional tool that offers an ideal balance between its simplicity and sophistication ideal for all types of professionals and for those who need to work from a distance.
Grandstream Gigabit Router GWN7000
The GWN7000 is a powerful multi-WAN Gigabit VPN router ideal for all types of businesses. It supports complete Wi-Fi and VPN solutions that can be shared across one or many different physical locations. It offers high-performance routing and switching power and a hardware-accelerated VPN client/server for secure connectivity between offices.
To maximize network reliability, the GWN7000 supports traffic load balancing and failover. the GWN7000 includes an integrated controller and a more automated provision that can configure and manage up to 300 GWN Series WiFi Access Points on the network.
GWN7600 WiFi Access Point
The GWN7600 is a mid-tier 802.11ac Wave-2 WiFi access point for small to medium sized businesses, multiple floor offices, commercial locations and branch offices. It offers dual-band 2x2:2 MU-MIMO with beam-forming thechnology and a sophisticated antenna design for maximum network throughput and expanded Wi-Fi coverage range.
To ensure easy installation and management, the GWN7600 uses a controller-less distributed network management design in which the controller is embedded within the product's web user interface. This allows each access point to manage a network of up to 30 GWN76XX series APs independently without needing separete controller hardware / software and without a single point - of - failure.
Android videophone - Yealink T58V
The Yealink T58V desktop phone is a very intuitive and easy-to-use videophone based on the Android 5.1.1 operating system. It features an adjustable 7'' HD screen, 2Mpx HD CAM50 camera, built-in Wi-Fi and Bluetooth 4.0 + EDR.
Mediatrix G7 - 1PRI
El Mediatrix G7 es un adaptador analógico VoIP fiable y seguro y un Gateway para SMBs.
En concreto, este Mediatrix G7 de 1 PRI ofrece la mejor solución para conectar equipos para servicios de telefonía en nube y sistemas de IP PBX a líneas terrestres PSTN.
Es ampliamente interoperable con softswitch SIP y proveedores de IMS, la serie Mediatrix G7 ofrece integración transparente de los sistemas PBX antiguos para las aplicaciones SIP Trunking y PSTN. Entres sus características más destacadas señalamos:
Patton CL1101E Extenders Kit
The CopperLink 1101E PoE Industrial amplifier kit can work with existing twisted pair and coax cable to provide an Ethernet service with PoE to IoT gateways, IP access points, telephones, cameras, digital signals, PCLs and more.
The CL1101E has the capacity to reach 10/100 Ethernet to more than 1000m and to supply more than 15V through Ethernet in 802.3af (inherited) to the available IP devices; it prevents interruptions, delays and the high cost of installing new cabling, allowing users to instantly benefit from the power and flexibility of IP communications.
Since it works on existing copper, (up to 2 or 4 twisted pairs) or coaxial cable the CopperLink 1101E solution allows the migration to All-IP and Internet of Things (IoT) giving a new life to circuits previously installed for traditional non-IP applications such as RS-232/485 controls, alarms, CCTV, analog phones, intercoms or loudspeakers, among others.
The CL1101E enables instant installation of PoE compatible devices such as wireless access points (WAPs), IP cameras, IP phones, video door phones, intercoms, sensors, intelligent LED lights, card readers, air conditioning and PLCs among others.
The Ethernet extender's extensive reach makes it possible to locate where it is required - which is paramount in applications such as building safety, where increased perimeter dimension and coverage area are critical points.
Characteristics
Patton Extender Kit CL1101
The CopperLink 1101 PoE amplifier kit can work existing twisted pair and coaxial cable to provide an Ethernet service with PoE to IoT gateways, IP access points, telephones, cameras, digital signals, PCLs and more.
The CL1101 has the capacity to reach 10/100 Ethernet up to 1000m and to supply more than 15V through Ethernet in 802.3af (inherited) to the available IP devices; it prevents interruptions, delays and the high cost of installying new cabling, allowing users to instantly benefit from the power and flexibility of IP communications.
Sice it works on existing copper, (up to 2 or 4 twisted pairs) or coaxial cable CopperLink 1101 solution allows the migration to All.IP and Internet of Things (IoT) giving a new life to circuits previously installed for traditional non-IP applications such as RS-232/485, alarms, CCTV, analog phones, intercoms or loudspeakers, among others.
The CL1101 enables instant installation of PoE compatible devices such as wireless access points (WAPs), IP cameras, IP phones, video door phones, intercoms, sensors, smart LED lights, card readers, air conditioning and PLCs among others.
The Ethernet extenders's extensive reach makes it possible to locate where it is required - which is paramount in applications such as building safety, where increased perimeter dimension and coverage area are critical points.
Panasonic KX HDV330 Preto
O telefone Panasonic KX HDV330 é um telefone equipado com uma tela LCD colorida de 4.3'', tecnologia Bluetooth, capacidade para 12 contas SIP, conferência de 3 vias e 24 teclas de função que, juntamente com seu calendário, podem armazenar até 2500 contatos.
Ele oferece grande versatilidade e facilidade de uso e possui capacidade de conexão para várias linhas com som HD de alta clareza.
Possui Gigabit Ethernet PoE e oferece a possibilidade de conectar módulos de expansão de teclado, permitindo aumentar o número de chaves flexíveis.
ATCOM Extension Module
The ATCOM extension module offers an additional feature block with 16 BLF keys and 4.3 inch TFT color display to increase the efficiency in the reception and management of calls.
ATCOM A68W Executive 6SIP + Wifi + PoE + Dual Screen (Color)
The ATCOM A68W IP Phone is an executive phone with dual screen color. Its main screen has 4.3 inch in order to optimize your experience thanks to the quality it offers. In addition, it offers up to 6 SIP accounts, WiFi connection, PoE and HD quality sound. The ATCOM IP phone ideal for companies with the most demanding needs.
ATCOM A48W 4SIP + Wifi + Color Screen
ATCOM A48W IP Phone is a terminal with 4 SIP accounts and a color screen with 3.2 inch (462x278px). It also has optional WiFi and a menu with function keys ideal for companies that need more organization in the day-to-day. In particular, it has: 4 soft-keys, 8 soft keys BLF and an extension module.
Excellent for companies that need enterprise applications with greater organization to join communications in the same device.
Amplificador Altavoz SIP 011404 - AC Powered
El amplificador de altavoz SIP Cyberdata es un dispositivo de localización por voz con alimentación a través de PoE (802.3af o 802.3at) que proporciona un método sencillo para implementar un sistema de búsqueda de sobrecarga basado en IP para instalaciones nuevas y heredadas.
Digium D65 IP Phone
The Digium D65 is an IP Phone with similar features to the D60. D65 also has 4.3 inch, color screen (480x272px) and similar features. Unlike the D60, the D65 has 6 lines and 2 ethernet ports of 10/100/1000Mpbs.
It also highlights, like all other phones in this range, its HD sound quality and its plug-and-pley technology, context-sensitive keys and advanced applications, can access any information instantly.
In addition, the Digium terminals will allow you to take full advantage of the flexibility and customization offered by Asterisk and Switchvox.
Grandstream Switchboard UCM6208
The Grandstream UCM6208 provides a solution to the communication needs of the company. It combines voice, video, data and mobility functions in one easy-to-use solution. In addition, it can be remotely managed and offers technologies such as voice, video call, videoconference, video surveillance, among others, that ensure an optimization of the communications in the line of this series UCM6200.
Grandstream Switchboard UCM6204
The Grandstream UCM6204 provides a solution to the communication needs of the company. It combines voice, video, data and mobility functions in one easy-to-use solution. In addition, it can be remotely managed and offers technologies such as voice, video call, videoconference, video surveillance, among others, that ensure an optimization of the communications in the line of this series UCM6200.
Digium D60 IP Phone
The Digium D60 IP Phone with color display (4.3 inch HD) and 2 line keys; furthermore it has two switches 10/100Mbps network ports. A very complete tool for any user that is looking for a terminal that optimizes the day-a-day at the company.
One of its most outstanding features is its HD sound quality and its plug-and-play technology, context sensitive keys and advanced applications, being able to access any information instantly.
USB WF40 Yealink
Wi-Fi USB Dongle WF40 is a low-power, small form factor device that can be implemented within offices for seamlessly connecting their IP phones to available wireless networks. This dynamic plug-and-play style USB device is an ideal office networking solution for companies - particularly, smaller organizations - seeking affordable, convenient and reliable high-speed wireless connectivity.
A new level of communication for your business
The expansion module Elastix UC46 is supported by the Elastix Color IP Phones UC842 and UC862. The module includes 20 bicolor keys and 2 extension arrow keys to switch page (allowing each module Elastix UC46 support up to 40 contacts / extensions) and offering a large LCD screen with backlight for easy viewing. Each contact can be assigned with an image showed on the display next to the corresponding key, allowing quick and easy identification of the user. Connect up to 6 modules Elastix UC46 to compatible Elastix phones, for a total of 240 phone contacts / extensions. This module supports functions like: Call waiting, parking, pick up
The Elastix UC46 module is the ideal solution for users who attend a high volume of calls, including executives and receptionists
Elastix UC862 is an executive IP phone with high-end quality. It provides multiple features that maximize your business productivity. It includes 4 lines with 4 SIP accounts, a 3.5 "LCD TFT screen with 480x320 pixels and 262K colors. It also has 47 buttons including 14 programmable and supports 5-way conferencing (up to 5 people call), also it has two Gigabit network ports, HD audio and other features. It is an enterprise-level IP solution that allows users optimize their business processes through telephony features and HD audio technology, providing a clear and productive voice communication.
Elastix UC842 is an executive IP phone with high-end quality. It provides multiple features that maximize your business productivity. It includes 4 lines with 3 SIP accounts, a 3.5 "LCD TFT screen with 480x320 pixels and 262K colors. It also has 37 buttons including 4 programmable and supports 5-way conferencing (up to 5 people call), also it has two Gigabit network ports, HD audio and other features. It is an enterprise-level IP solution that allows users optimize their business processes through telephony features and HD audio technology, providing a clear and productive voice comunication.
Clear communications in your business
Elastix UC802 meets the requirements of VoIP telephony and also offers some functions that are indispensable for their deal. Based on its excellent performance price, makes it the perfect choice for small and medium businesses, home offices and private users. A phone with HD sound and multiple functions that maximize business productivity. It includes 2 lines with 2 SIP accounts, a graphical LCD, 7 programmable keys, dual network ports with integrated PoE, and 5-party conference.
Amplificador Altavoz SIP 011405 - PoE
The new Cyberdata SIP-enabled Loudspeaker Amplifier (PoE) provides an easy method for implementing a loud IP-based overhead paging system for loud areas, warehouses, manufacturing areas, and outdoor areas.
Cyberdata Amplificador Paging V2 - Model011324
The new Cyberdata SIP-enabled PAging Amplifier provides an easy method for implementing a loud IP-based overhead paging system for loud areas, warehouses, manufacturing areas, and outdoor areas.
Now with up to 9 user-stored messages and 25 watts of driving power (802.3at) the amplifier provides direct drive of up two 8 ohm horns.
The interface is compatible with most SIP-based IP PBX serves that comply with the SIP RFC 3261. For non-SIP environments, the Paging Amplifier can be configured to listen to multicast address and port number combinations to forma paging zones.
Digium D62 IP Phone
The Digium D62 IP Phone with color display (4.3 inch HD) and 2 line keys; furthermore it has two switches 10/100/1000Mbps network ports. A very complete tool for any user that is looking for a terminal that optimizes the day-a-day at the company.
ATCOM A20W - 2SIP + Wifi
ATCOM's A20W IP Phone is another of the models in its A1X range. Specifically, the A20W stands out for its 2 SIP accounts and for being equipped with Wifi (just like the model A20W of a SIP account). Unlike the A20W, it also has 2 programmable keys.
Elegant design and easy operation are other features highlighted among the features of this phone, ideal for companies that are looking for a easy configuration tool.
ATCOM A10W - 1SIP + Wifi
ATCOM's A10W IP Phone is the basic model within its A1X range. In particular, the A10W is a very complete tool, easy to configure with excellent sound quality. ATCOM emphasizes its elegant design and easy operability as characteristics outstanding among its benefits; ideal for companies that are looking for a simple phone without losing sight of the quality.
treeMT: Multi Tenant Platform para empresas y/o entornos residenciales
treeMT es una plataforma virtual, donde poder crear y gestionar tantas centralitas virtuales como se necesiten, (hasta un máximo de 2.000 extensiones entre todas). Avanzada 7 garantiza además el mantenimiento de la Plataforma Multi Tenant, incorporando nuevas features y correcciones en versiones sucesivas.
Otra de las ventajas de treeMT es que es una máquina virtual que le permite trabajar con su Hypervisor favorito sin tener que adaptar su hardware de trabajo.
treeMT está dirigido a operadores VoIP que necesitan ofrecer Centralitas Virtuales en una plataforma de gestión propia sin cambiar de operador (manteniendo sus carrier habituales).
Caracteristicas Principales:
Alcatel Temporis IP251
El Alcatel Temporis IP251G ofrece una excelente calidad de audio ya sea en modo manos libres o a través de auricular. Su pantalla retro iluminada facilita además las llamadas y configuración del terminal gracias a su diseño intuitivo del menú.
Un terminal IP que se presenta en un diseño compacto y con gran compatibilidad; además ofrece Gigabit de dos puertos Ethernet lo que le permite mejores niveles de rendimiento aprovechando al máximo su red.
Características adicionales
Gateway Patton SN4170 - 1 PRI (T1/E1) - High Precision Clock
The SmartNode 4170 Patton is the next generation model ISDN T1/E1 in this VoIP line of Patton. A network device used for converting voice calls, in real time, between your PBX and VoIP network.
This model, in fact, is ideal for small and medium businesses looking for cost-effective ways to dispose of such lines. Specifically, this model has a T1/E1, Gigabit Ethernet port and provides up to 30 simultaneous calls.
This model, in fact, is ideal for small and medium businesses looking for cost-effective ways to dispose of such lines. Specifically, this model has a T1/E1, Gigabit Ethernet port and provides up to 15 simultaneous calls.
Grandstream GDS3710 IP Video Door Phone
The Grandstream GDS3710 video door phone is an IP video gateway system that can also be used as a high definition IP surveillance camera and to provide access control to security installations and monitoring for building of all sizes.
This video entryphone incorporates an intercom and offers a viewing angle of 180º and includes an RFID reader to enter keyless with total security. It also includes a microphone and speaker to support the functionality of the intercom and also has an incoming and outgoing alarm to integrate with existing security devices.
The GDS3710 also integrates a complete GDS management software that allows the reading of RFID cards and loads the information of said card into the sofware.
Grandstream GXP1782
GXP1782 is a powerful mid-range IP Phone that offers advanced phone features. Like the GXP1780, it incorporates 8 lines, 4 SIP counts, BLF keys and backlit 200x80px LCD screen. In addition, it offers the possibility to customize your ring tone with personalized music. The difference with the GXP1780 is in its dual 10/100/1000Mbps Gigabit port with integrated PoE (10/100Mpbs in the GXP1780 model).
Like other phone in this 17XX series, it's equipped with Kensinigton anti-theft security slot. Ideal for users that are looking for a mid-range phone with quality performance.
Grandstream GXP1780
GXP1780 is a powerful mid-range IP Phone that offers advanced phone features. It incorporates 8 lines, 4 SIP counts, BLF keys and backlit 200x80px LCD screen. In addition, it offers the possibility to customize your ring tone with personalized music and integration with web and business applications. Like other phone in this 17XX series, it's equipped with Kensinigton anti-theft security slot. Ideal for users that are looking for a mid-range phone with quality performance.
Gateway ATA HT814 (4FXS + 2THS)
The Grandstream HT814 is an analog phone adapter (ATA) with 4 FXS ports and 2 integrated Gigabit NAT routers. It offers excellent voice quality, encryption reinforced with single security certificate per unit, automatic provisioning and easy use in the handling of the device. An excellent device to optimize network performance at home in the office.
Gateway ATA HT812 (2FXS + 2THS)
The Grandstream HT812 is an analog phone adapter (ATA) with two FXS ports and an integrated Gigabit NAT router. It offers excellent voice quality, encryption reinforced with single security certificate per unit, automatic provisioning and easy use in the handling of the device. An excellent device to optimize network performance at home in the office.
BF19Bracket DBL 19''
19'' bracket for 2 beronet Gateways on 1U for the installation in server racks. The package contains: 2 parts for the outside as well as one middle piece for the connection of the 2 gateways.
Yealink HD camera CAM50 for SIP T58V / SIP T58A
Yealink CAM50 camera is an HD video camera with high quality, a very easy-to-use complement to the SIP-T58V / SIP-T58A.
It presents a resolution of 2mpx and 720p granting an interactive collaboration with experience in first class video communications. There is no additional software of any type of control is required, you simply have to insert the camera into the USB port located at the top of your terminal and your technology connection will sinchronize the device directly.
Key features and benefits
Yealink T58A
Yealink T58A is an easy-to-use phone that offers HD video and audio recording allowing visual communication that optimizes and increases the productivity of the company. It is based on the Android 5.1.1 operating system, it has 7'' of adjustable multipoint touch screen, integrated WiFi and bluetooth 4.0. As usual in the phone of this T5XX range from Yealink, they are terminals that achieve a great balance between simplicity and sophistication, offering an ideal all-in-one communicacions solution for managers, executives and teleworkers of small and medium-sized companies
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Teléfono IP Cisco 7861
The Cisco 7861 combines a new and attractive ergonomic design lik the rest of the 7800 series phones. Specifically, this Cisco 7861 terminal offers advanced IP telephony and broadband audio features that ensure crystal clear sound with the aim of providing a complete and easy-to-use voice communications experience.
On the other hand, like the rest of the phones in its series, it incorporates fixed function keys providing access to the functions of service, messaging, directory, transfer, etc. An ideal phone for any company that wants upgrade their traditional telephone system to an IP communication system.
Teléfono IP Cisco 7841
The Cisco 7841 combines a new and attractive ergonomic design lik the rest of the 7800 series phones. Specifically, this Cisco 7841 terminal offers advanced IP telephony and broadband audio features that ensure crystal clear sound with the aim of providing a complete and easy-to-use voice communications experience.
Teléfono IP Cisco 7811
The Cisco 7811 combines a new and attractive ergonomic design lik the rest of the 7800 series phones. Specifically, this Cisco 7811 terminal offers advanced IP telephony and broadband audio features that ensure crystal clear sound with the aim of providing a complete and easy-to-use voice communications experience.
Cisco IP Phone 6851
The Cisco 6851 IP Phone offers all features of Cisco 6800 Series. A cost-effective and high-quality team designed to improve your company's communications and reduce costs thanks to the advantages of its VoIP technology.
This 6800 series in particular, combines an ergonomic design with security of its communications. All phones in this series offer advanced IP telephony and broadband audio features and an easy-to-use interface to optimize call control hosted by third parties. Specifically, the model 6851 offers 4 lines and is available in gray.
Teléfono IP Grandstream GXP1760 WiFi
The GXP1760W is an IP phone with integrated WiFi mid-range similar in performance to its compation GXP1760 and excellent performance an ideal design for users in any field.
Specifically, this terminal incorporates 6 lines, 3 SIP accounts, 6 bicolor line keys and 4 programmable context sensitive XML keys on a 200x80px (3.3''), LCD screen with backlit. For additional customization, this model offers ringtone / standby tone with personalized music and integration with advanced web and business applications, as well as a local weather service.
Like other models, this GXP1760W supports the fastest possible connection speeds with two 10/100Mbps network ports with automatic detection, as well as automatic provisioning features with media access control.
Gateway Vega 60G - 4 FXO
Yealink CP920 - Audioconference
The Yealink CP920 phone offers a modern and complete audio conference solution. The CP920 is sensitive to touch and combines simplicity with the sophistication of functions; ideal for small and medium businesses.
Besides having audio quality, it can not only be combined with your movile device: smartphone or PC / tablet through its bluetooth, but it is an ideal option for companies that use a public PSTN telephone network after combining the CPN10 PSTN Box.
It is a valuable addition to your conference room, achieving an excellent balance between user-friendliness and powerful features, giving you a simple and clearly attractive business conference experience.
El teléfono Mitel 612d es el modelo básico para empresas de la familia 600d de teléfonos DECT. Con una pantalla TFT de 2'' a color garantiza una vista clara y un manejo sencillo gracias al menu claramente estructurado. También soporta instalación de firmware actualizado en la web con descarga inalámbrica que ayuda a reducir costes de mantemiento.
Wireless Solution Yealink W60P (Handset + Base Station)
The Yealink W60P wireless solution is the idea solution for small and medium businesses. It can be synchronized with up to 8 handsets from Yealink W52H / W56H, which will allow you to enjoy greater mobility and efficiency in flexibility as well as eliminate load problems and additional wiring significantly. In addition this solution increases its performance in a significant way since it not only supports up to 8 VoIP accounts and concurrent calls, but also accelerates its connection start and signaling while reducing downtime.
Unlike others Yealink's models, this one allows up to 8 handsets instead of 5 and up to 8 simultaneous calls and SIP accounts.
Yealink W60B Base Station DECT
Base station Yealink W60B for small and medium enterprises. Can pair with up to 8 handsets of Yealink W56H. In addition, to support 8 VoIP accounts and up to 8 simultaneous calls ensures high performance thanks to accelerates its start and reduces downtime.
Unlike other wireless packs from Yealink, this base station incorporates the Opus codec, offering excellent audio qualities both in a broadband scenario and in poor network conditions. In addition, this base station supports 'Zero Touch Provisioning', much more efficient and effortless for the user, which makes it easier to maintain and update, saving even more time and costs for the company.
A diferencia de otros packs inalámbricos de Yealink, esta estación base incorpora el códec Opus, ofreciendo excelentes calidades de audio tanto en un escenario de banda ancha como en condiciones de red deficientes. Además, esta estación base admite aprovisionamiento 'Zero Touch', mucho más eficiente y sin esfuerzo para el usuario, lo que hace que sea más fácil de mantener y actualizar, ahorrando aún más tiempo y costes para la empresa.
M215SC: M200SC + M15SC
The Snom M215SC is composed of the M200SC base and the high performance M15SC wireless phone from Snom for sigle cell solutions.
The Snom M1SC is a high-performance cordless phone designed to be combines with single-cell (SC) base stations, making it the ideal solution to meet the day-to-day needs of businesses. Thanks to its 4.3cm widescreen graphic display, its backlit keyboard and crystalline sound is positioned as a multifaceted device with very varied features for multiple contexts.
The M200SC base station allows up to four simultaneous calls and can register up to six M15SC.
Key features M15SC
Key features M200SC
Cordless phone Snom M15 SC
The Snom M15 SC is a high-performance DECT cordless phone from Snom, designed for use in combination with the M200SC single-cell base station. It features a 1.7 widescreen graphic display, backlit keyboard an excellent sound tranmission; qualities very necessary for day to day, either in the office or at home.
Thanks to its combination with the M200 SC base station, up to six M15 SC can be paired and up to four simultaneous calls can be made.
Telefone Gigaset SL450Hx
O Gigaset SL450Hx é um terminal DECT ideal para complementar uma instalação telefônica ou telefônica existente.
É compatível com qualquer telefone sem fio Gigaset (recomendável SL4500 ou SL); Também é compatível com PRO PBXs, que aceitam terminais sem fio DECT / GAP, com pontos de acesso sem fio DECT e repetidor de alcance.
Gateway Mediatrix G7 - 1 PRI + 4 FXS
Gatewa Mediatrix S7
A série Mediatrix de adaptadores VoIP analógicos (ATA) oferece a opção de conectar com perfeição IP e equipamentos legados a um IP-PBX hospedado ou local. É também a solução ideal para conectar um PBX e telefones tradicionais a uma rede central sem afetar a base atual de clientes.
Dessa forma, a série S7 permite que operadoras e provedores usem sistemas híbridos de telefonia IP e analógica com uma ampla variedade de aplicativos lucrativos e fáceis de implementar.
Os gateways Mediatrix são totalmente certificados com softswitches da Broadsoft e são compatíveis com o Skype for Business.
Telefone IP para desktop Snom D712
O telefone IP de mesa Snom D712 foi projetado para garantir som de alta definição; o seu fone de ouvido tem uma ótima configuração de alto-falante e microfone e, graças à grande variedade de codecs, oferece uma qualidade de som cristalina.Um telefone ideal para provedores de soluções de telefonia IP graças ao suporte padrão IPv6, o que torna a solução perfeita para instalações em larga escala.
Um terminal flexível com funções avançadas de gerenciamento que também incorpora os mais recentes protocolos de segurança VoIP, garantindo o máximo de confidencialidade em suas comunicações; um telefone ideal para o dia a dia na empresa.
Características principais
Grandstream GWN7600LR WiFi Access Point
This WiFi long range access point is designed to provide extended coverage support. Ideal for outdoor WiFi solutions thanks to its waterpoof casing and heat resistant technology. The GWN7600LR comes equipped with dual-band 2x2:2 MU-MIMO with beam-forming technology and a sophisticated antenna design for maximum network throughput and extended WiFi coverage range of up to 300 meters.
To ensurey easy installation and management, the GWN7600LR uses a controller-less distributed network management design and an embedded controller within the product's web user interface.
Snom Conference C520
The Snom Conference C520 is an audio-conference solution aimed at professionals and companies of any size, ideal for company meetings. It comes equipped with 3 microphones (one of them integrated in the phone). In addition, it incorporates two DECT microphones that can be easily dismantled and placed remotely to cover the sound of other conference participants that are farthest from the terminal.
Each of thse microphones of the C520 use dynamic reduction and adaptative feedback control to provide crystal clear HD audio transmission, even in crowded or spacious rooms.
On the other hand, in order to cover optimal coverage, microphones are acapable of synchronizing in real time based on their current position. Thanks to this technology, C520 allos the user freedom of movement without even having to raise their voice-regadless of meeting room size.
Nowdays, connectivity is the most important factor in efficient collaboration and therefore, the C520 is equipped with a versatile interface and Bluetooth to allow connection with mobile phones and headphones.
In order to adapt to any environment, the Snom C520 can also expand in coverage by pairing with up to three C52 wireless microphones and can greatly inrease the range of performance.
Yealink Expansion Module EXP50
The Yealink EXP50 expansion module is compatible with Yealink T5 Series telephones including: T58V, T58A, T56A, T54S and T52X. A module designed to expand the functional capacity of the SIP phone to another level.
In addition, it provides you with a simple user interface and advanced call handling capabilities. For example, 3 pages of 20 flexible button shown on the display can be programmed up to 60 various functions.A tool that makes easier the user experience, simplifying and optimizing your work time.
CP960 Audio conference kit ( 1xCP960 + 2xCPW90)
Yealink IP audio conferencing kit that includes this new CP960 equipment and adds two CPW90 microphones.
An equipment with HD sound quality ideal for medium and large rooms offering an excellent combination of quality and desing. It offers among its most outstanding features Android 5.1 support, wireless microphones and synchronization option with your company mobile, smartphone of PC/tablet via Bluetooth or via Micro-B USB.
Wireless Microphone CPW90
The Yealink CPW90 is a wireless expansion microphone compatible with the CP960 audio conference terminal. It incorporates superior audio technology and supports a range of 360º voice reception offering coverage up to 3m.
Ideal for organizations that need to cover connectivity needs between their different headquarters and employees who work remotely.
PROMOTION: 30% off aditional discount for first purchases - Only up to October 31th
Audio conference IP equipment Optima HD CP960
Yealink incorporates to its range of products specialized in IP audio conferencing this new terminal in form of Y like its first letter of their commercial name.
Snom A100M monoaural headset
The Snom A100M is a monoaural wired headset designed for maximum comfort and performance. Its light and ergonomic design makes it possible for daily use to be very comfortable. Its broadband technology also guarantees high definition sound and clear communication. In addition, the flexible rod and the passive noise canceller of your microphone guarantee a clean voice transmission. With its various QD (Quick connect / disconnect) adapters, also available in Avanzada 7, this headset is perfect for use in multiple scenarios.
Snom A100D biaural headset
The Snom A100D is a biaural wired headset designed for maximum comfort and performance. Its light and ergonomic design makes it possible for daily use to be very comfortable. Its broadband technology also guarantees high definition sound and clear communication. In addition, the flexible rod and the passive noise canceller of your microphone guarantee a clean voice transmission. With its various QD (Quick connect / disconnect) adapters, also available in Avanzada 7, this headset is perfect for use in multiple scenarios.
Snom adapter for A100M and A100D headphones
The Snom USB adapter cable allows you to connect the A100M and A100D headphones to all the fixed terminals of Snom that have a USB port. It has integrated quick access keys for the "accept" and "end" functions, call, silence and volume control making it even easier to use the phone.
The Plug & Play technology makes it possible to use it inmmediately by simply connecting it.
Mediatrix Sentinel 100
O Mediatrix Sentinel 100 combina um SBC (Session Border Controller) com um Media Gateway em uma plataforma de serviços sólida, ideal para SMBs. Ele oferece até 120 canais VoIP simultâneos e atende a aplicativos de até 500 usuários.
Ele é projetado para uma ampla variedade de casos de uso, incluindo troncos SIP, serviços hospedados e comunicações unificadas. Oferece suporte para ISDN PRI, E&M e R2 E1/T1 CAS; também oferece conectividade para recuperação de PBX e PSTN herdados.
Empresa de telefonia deve integrar estável software multi-função e hardware confiável para garantir o tráfego de chamadas necessárias com a máxima qualidade de serviço (QoS). Elastix ® entende que o negócio VoIP requer hardware de qualidade com grande capacidade e, mais importante, o apoio. Nós entendemos o seu negócio e quer ajudá-lo a desenvolver a melhor solução possível para os seus clientes e / ou sua empresa.
Aparelhos Elastix ® da Série ELX oferece a mesma Elastix ® software com todas as funcionalidades e confiabilidade que você tenha se acostumado a, com grande hardware de telefonia de grandes produtores de telefonia IP. É certamente a melhor opção disponível no mercado para pequenas e médias empresas.
Características Gerais Design compacto
Nossos aparelhos têm um design simples e compacto, perfeito para a portabilidade e facilidade de manutenção. Com casos metálicas em 1U, 2U 1.5U e formatos, os nossos aparelhos têm capacidade de expansão suficiente para uma ampla gama de aplicações que utilizam portas PCI ou USB para apoiar ElastibankTM bancos de canais. Todos os nossos aparelhos são rack permitindo uma integração rápida à sua infra-estrutura de rede.
Baixo consumo de energia
O design dos nossos aparelhos lhes permite consumir o mínimo de energia possível em condições normais. Isso nos ajuda a economizar energia e reduz o custo de operação. Nós temos feito isso, entre outras coisas para ajudar a contribuir com a conservação do meio ambiente.
Integração Digital e Analógica
O Elastix ® aparelhos Series ELX pode integrar cartões digitais ou analógicos (FXO / FXS, E1/T1) de acordo com suas necessidades. Estamos prontos e prazer em dar-lhe conselhos antes de pedir uma implementação, a fim de certificar-se de que você explorar a funcionalidade do aparelho até o limite. Todo o hardware vem instalado, configurado e testado a partir de nosso armazém.
Extensões opcionais
* Representa uma estimativa segura em um cenário básico
Patton SmartNode RDSI - 2BRI
Patton SN4131 is part of SmartNode SN4130 Series, the new generation of ISDN BRI models from the renowned SmartNode VoIP product family.
The SmartNode 4131 Gateway allows connection to the IP PBX network and other VoIP devices, providing this 2 BRI port. It supports all major VoIP standards 8SIP, H.323, T.38 Fax, Codec G.729), offering high standards of quality and performance.
Its configurations are tailored to the needs of small and medium businesses looking for a cost-effective way to connect PBX systems to multiple sites or connect them to a public Internet telephony service.
Like all Trinity SmartNodes, the SN4130 comes with an integrated Web wizard for quick and easy setup. You can create your own web interface using the web assistant features.
Patton SmartNode RDSI - 4BRI
The SmartNode 4131 Gateway allows connection to the IP PBX network and other VoIP devices, providing this 4 BRI port. It supports all major VoIP standards 8SIP, H.323, T.38 Fax, Codec G.729), offering high standards of quality and performance.
Teclado de expansão Sangoma EXP 100
O módulo de expansão do Sangoma, EXP100, é um teclado que complementa o telefone Sangoma projetado especificamente para usuários que normalmente lidam com um volume maior de chamadas.
Este módulo de expansão é compatível com os telefones Sangoma S500, S700 ou S705.
Snom C52-SP - Audioconference
The Snom C520-WiMi combined with the Snom C52-SP is the perfect solution for people who regularly organize teleconferencing in large rooms or with large numbers of people. Up to 3 C52-SP can be synchronized wirelessly with the Snom C520-WiMi and the function as separate modules thanks to its two built-in microphones and its powerful speaker.
The C52-SP can move to any part of the room allowing a distance of up to 50m thanks to its DECT wireless connection and its battery. In addition, thanks to the automatic voice synchronization of all microphones, attendees will have no problem synchronizing their conversation and HD-Voice quality. On the other hand, the C52-SP also has an autonomy of up to 12 hours thanks to its battery.
2.5mm adapter cable for A100M and A100D
The Snom ACPJ adapter cable allows you to connect the Snom A100M and A100D headphones to any device with a 2.5mm audio port.
It's ideal for Snom DECT terminals (M25, M65, M85), and is compatible with most mobile devices. Compatible with the A100M and A100D headphones as well.
3.5mm adapter cable for A100M and A100D
The Snom ACPJ adapter cable allows you to connect the Snom A100M and A100D headphones to any device with a 3.5mm audio port.
Cyberdata VoIP Servidor Paging V3 011146
The Cyberdata SIP Paging Server enables users, through a single SIP phone extension, to access multiple zones for paging with bell scheduling in a VoIP network and to connect to legacy analog overhead paging systems.
A second SIP extension can be configured as a Night Ringer, playing a user-uploadable audio file.
The SIP Paging Server now has a built in bell scheduler that enables, through a secure web interface, scheduled notifications to be sent to different multicast zones and legacy analog paging systems.
Perfect for schools, universitites, manufacturing or any environment where scheduled audio notification is paramount.
There are up to 25 stored message and 25 individual bells, which are user uploadable.
Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress voice signals, to save on bandwidth requirements. While voice signals from the Public Switched Telephone Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of different voice codecs including such as G.729, G.726, AMR, G.722, iLBC, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.
The D150 card allows to convert numerous simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows running up to 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.
Panasonic´s KX-UT670 Smanrt Desk Phone is the company´s first SIP open source-based operating system desk phone and is equipped with a large 7-inch LCD touchscreen wich allows for easy access to control phone features and installed applications.
Designed as a communication solution and business tool to enhance operation, health care and hospitality, the KX-UT670 gives users the opportunity to develop applications to enhance business operations and could include applications such as nurse call systems and easy ordering from a touchscreen for room service.
Panasonic KX-A422CE specifications
RFP 43 WLAN connects two mobility standards: DECT, which enables the use of mobile system phones, and an integrated WLAN Access Point, which provides flexible access of PCs or other workstations to the companies network.
The WLAN Access Point allows the installation of WLAN hot-spots, e.g. in conference rooms. They can be part of the powerful, company-wide DECT network, established with RFP 32 and 34.
RFP 43 WLAN offers due to integrated IEEE 802.11n compatible WLAN Access Point significant advantages compared to a separat infrastructure for DECT and WLAN:
* Only one access point for both technologies (DECT and WLAN) * Only one switch-port is required (PoE recommended) * Integrated, centralized management and set up for DECT and WLAN * Net View Management: • RFP 43 WLAN Access Points can be grouped in clusters • The WLAN cluster can be managed for all access points simultaneously • It is not required to administer the access points individually
Additionally RFP 43 permits * easy installation due to preinstalled software * simultaneous use as OpenMobilityManager (OMM) in the network
Cyberdata SIP Strobe
When installed, the wall-mounted SIP Strobe operates as a visual alerting device. The device can be set up as part of a ring group (call group) with the Strobe engaged as notification of incoming calls.
Features: ? Meets ADA requirements for telephony signalling and notification ? Event-controlled relay ? Tamper sensor ? Web-based setup ? PoE-powered
Use areas include: ? Classrooms ? Banks or financial institutions ? Court rooms ? Manufacturing warehouses ? Office ? Indoor only
The GXV3500 is an innovative next generation IP video encoder + decoder + public announcement system 3-in-1 combo device.
It features cutting edge H.264 real-time video compression for analog video as well as IP video decoding with excellent image clarity. It offers industry leading SIP/VoIP for 2-way audio, video streaming to mobile phones and video phones, integrated PoE, a large pre-/post-event recording buffer, and advanced security protection. Its integration of comprehensive peripherals including microphone input, alarm control and TV/audio output allows the device to also function as a powerful and flexible voice/video public announcement system using microphones, IP phones, or IP video phones. The GXV3500 can be managed with GSurf, Grandstream’s advanced and intuitive video management software that controls up to 36 cameras simultaneously. It offers an HTTP API and is fully compliant with ONVIF standard. The GXV3500 is an advanced first-of-its-kind IP video encoder + decoder + public announcement system 3-in-1 combo product for professional surveillance and security monitoring applications.
Power supply for Grandstream GXV3175
Product Code: 1VPMOCT256LF
For Asterisk users who connect to the PSTN, the most common type of echo is hybrid echo - the echo introduced by the impedance mismatch between 2-wire and 4-wire telephone circuits. The echo manifests as a distorted and delayed reflection of the users voice while in conversation with an external party through the PSTN. Asterisk itself offers a range of software-based open source echo cancellation routines that are moderately effective in eliminating the hybrid mismatch echo that most Asterisk users experience.
However, there are cases in which these algorithms are not effective. To combat this, Digium introduced DSP-based echo cancellation modules for our multi-port T1/E1/J1 cards and our 24-port analog card.
The Panasonic KX-A239 is an AC adapter for NT 300 / UT 100 Series. It is required for non PoE (power over ethernet) networks when using the KX-NT300 series IP Telephones. The adapter is OPTIONAL for the UT series. KX-A239 Features:
The Panasonic KX-A239 has lots of exciting features for all customers. PremiumStore sells the Panasonic KX-A239 as a Brand New item. These are just a few of the reasons to buy a Panasonic KX-A239 from PremiumStore today!
Cyberdata PoE Power Injector, 802.3at
The new PoE power injector 802.3at is used with CyberData’s new V2 Paging Amplifier (Part # 011061) and V2 Loudspeaker Amplifiers (AC - Part #011095 and Wireless - Part #011096) to provide up to 25 watts of driving power.
The Power Injector 802.3at eliminates the need for a PoE enabled hub. In a typical installation, the Power Injector is connected between a non-PoE hub and a CyberData paging device, providing inline power capability to an Ethernet Cat 5 cable.
Alphatech Door Phone SLIM SDP 04 can be connected to PBX FXS or analog PSTN line.
Door Phone SLIM is easy to install outdoor and indoor. It offers 3 models with 1, 2 or 4 buttons.
Door phone SLIM is provided with a relay for lock control, a blue backlit case and a heating system.
The USB connection and the supplied programming interface for Windows make this door phone easy to configure. Benefits
Configuration
Supervisor Cord Product Code: ADD-1014
The Addcom Supervisor Cord allows two headsets to operate on one telephone for training/monitoring purposes. Both headsets can listen, but only one headset will have a live microphone. The other microphone is muted. The mute switch allows the supervisor to control which microphone is live and which is muted.
To enable users enjoy the advantages of using IP networks together with the DECT technology, Mitel has developed radio fixed parts (RFP) with IP interfaces for integrating DECT into IP networks. These radio fixed parts (RFP L32 IP and RFP L34 IP) are connected to the network like IP terminals. Voice is conveyed via VoIP to the radio fixed part and from the RFP to the air via DECT.
Therefore, employees can always be reached via their call numbers, regardless of whether they are in a branch of the company or at the head office. Using the same IP connections for data and telephony saves additional infrastructure and, thus, costs.
No matter the size and range of the IP network, a single Open Mobility Manager (OMM) is enough to manage all RFPs of the multi-cellular DECT network. This is installed on any of the RFPs by software. The OMM manages up to 256 RFPs and 512 handsets.
The base station RFP L42 WLAN allows the integration of mobile data transmission via WLAN in parallel using the same network. Central administration of the DECT and WLAN network available via a browser interface.
License Mitel RFP V2 (up 10 Mitel DECT bases)
(PL9677-B1)
Asoka Pluglink® to 500MB with pass-through plug
Asoka’s PL9677-B1 PlugLink adapter comes with a single 100M Ethernet port and offers PLC network transmission speeds of up to 500 Mbps. A pass-through plug provides a power socket for additional electrical appliances. It offers the easiest solution for premium quality in-home network performance for high speed transmission of HD video, data, voice and audio between devices. It enables fast, secure wireless connectivity throughout the home or office. No need to run extra cable lines—PlugLink works with existing electrical wiring.
PlugLink is designed to provide secure, reliable communications services ranging from HDTV, IPTV, high data rate broadband sharing, audio and video streaming, VoIP, file and application sharing to network and online gaming.
Suitable Applications
With this module you can upgrade your Sangoma card D500 to handle more sessions.
Cisco SPA112 (2 Port Phone Adapter) High-quality Voice and Fax over the Internet
Now you can use your phone over the Internet, without compromising on voice quality or phone and fax features. The Cisco SPA112 2-Port Adapter offers the benefits of high-quality voice over IP (VoIP) without the need to upgrade your existing analog phones.
Easy to install and use, the SPA112 works over an IP network to connect analog phones and fax machines to a VoIP service provider. It also provides support for additional LAN connections.
The Cisco SPA112 is compact in design and compatible with international voice and data standards. It can be used with residential, home-office, and small-business-VoIP service offerings, including full-featured hosted or open source IP PBX environments.
The Cisco SPA112 2 Port Adapter:
El 6737i combina la excepcional calidad de audio HD con una gran cantidad de características potentes y flexibles basadas en estándar.Con un diseño fino y elegante, pantalla LCD de 144x75 pixeles retroiluminada, 12 teclas contextuales programables. capacidad XML, capacidad de acceder a aplicaciones personalizadas, y soporte hasta 9 llamadas simultáneas.Ese modelo cuenta con capacidad Gigabit de red.
IP Bell - VoIP SIP video door entry phone station
Alphatech BELL expansion module 8 button.
8 single call button modules (combinable with audio-video or only audio door station).
Door station Alphatech BELL series in equipped with surface mounting box and front plate with frame for reduced wiring technology.?
Buttons are to be purchased separately.
Panasonic Wall Mount Kit for KX-UT670
Patton SmartNode 4112 (2FXO) VoIP Media Gateway
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. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. 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 be routed to the PSTN or the IP network 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 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration 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 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.
Patton SmartNode 4112 (2FXS) VoIP Media Gateway
Patton SN4114 - 4FXO SN4114/JO/EUI
VoIP Media Gateway
Patton SN4114 - 4FXS SN4114/JS/EUI
Patton SmartNode 4118 (4FXS + 4FXO) Patton SmartNode 4118 VoIP Media Gateway soporta hasta 8 llamadas VoIP para operadores y el acceso a empresas. Conexión a cualquier extensión analógica o PBX. La serie SN4110 es la elección perfecta para la conectividad de puertos analógicos a VoIP, proporciona tono/timbre, identificador de llamadas y otros servicios. Las llamadas se pueden enrutar automáticamente a la PSTN o la red IP al tiempo que proporciona flexibles planes de numeración. Aplicaciones Extensión de las lineas de su oficina remota. Ahora, en lugar de instalar un PBX separado en la oficina remota, el SmartNode 4110 es capaz de proporcionar líneas PSTN mientras se conectan simultáneamente varias ubicaciones. Se pueden gestionar de forma centralizada y beneficiarse de los servicios PBX como llamada a grupos, enrutamiento de menor coste, y desvío de llamadas. El acceso PSTN permite realizar llamadas locales remotamente. Además, la central corporativa puede cursar llamadas salientes y evitar los cargos de larga distancia mediante el uso de la oficina remota.
Este modelo soporta 4 puertos FXS + 4 puertos FXO.
Características:
(PL9671-A2)
Asoka Pluglink® to 500MB
Asoka PlugLine® adapters provide excellent value, offering the highest performance (500 Mbps) and only lifetime warranty in the industry.
Asoka’s PL9671-A2 PlugLink adapter comes with a single 100M Ethernet port and offers PLC network transmission speeds of up to 500 Mbps. It is the easiest solution for premium quality in-home network performance for high speed transmission of HD video, data, voice and audio between devices. It enables fast, secure wireless connectivity throughout the home or office. No need to run extra cable lines—PlugLink works with existing electrical wiring.
Antena dipolo Mitel (RFP 34 IP y RFP L34 IP)
Logitech HD Pro Webcam C920 Referencia 960-000767
Package Contents
And...
Logitech webcam software:***
Recommended requirements for full HD 1080p and 720p video calling*:
Visit your preferred video calling provider’s website for exact information on system and performance requirements.
Skype® in Full HD 1080p* Get breathtaking Full HD 1080p video calls on Skype for the sharpest video-calling experience. Smoother. Sharper. Richer. Clearer. Logitech Fluid Crystal™ Technology. It’s what makes a Logitech webcam better. It’s smoother video, sharper pictures, richer colors and clearer sound in real-world conditions.
Enjoy widescreen HD 720p video on most major IMs and Logitech Vid™ HD.
One-click HD upload (1080p) to Facebook, Twitter™ and YouTube™ makes it easy to share your life with friends and family. * Please download the latest version of Skype, Skype 5.8 for Windows, which offers 1080p HD video calling.
The Konftel 55 – for efficient web and teleconferencing
The Konftel 55 is an easy-to-use, compact conference unit with impressive, crystal-clear sound thanks to the patented audio technology OmniSound® HD.
The Konftel 55 has been specially designed to be the hub of your communications and connects computers, mobile phones/tablets and desktop phones. Whatever the communication tools, your meetings will be conducted with superior sound quality.
Moreover, VoIP calls can be bridged with calls via a desktop or mobile phone. Combine and switch connections easily on the LCD colour screen's smart user interface that only displays the current connections. Make a call and the meeting is under way!
The Konftel 55 lets you record calls or dictations and you can move the memory card to your computer to save or share the audio files. The Konftel 55 is highly portable and looks just as good on the desktop and in the home office as in the conference room.
Battery Konftel 55 & Konftel 55W Item no: 900102124
Smart Battery 2260mAh Li-Polymer. One-year guarantee.
This rechargeable battery lets you use the Konftel 55W in a completely wireless phone connected via a dedicated cable (accessory request) or Bluetooth ™ (55W). The battery can be recharged via the mains (adapter included) and USB. The Konftel 55W, the battery is charged via the included AC adapter or via USB. It takes about 4 hours to fully charge a discharged battery using the power adapter and 5 hours via USB. Konftel 55W must be connected to a powered USB port (for example, a computer) to charge the battery. Note that it takes much longer to charge via USB if the Konftel 55W is off
Patton SN4118 - 8FXS SN4118/JS/EUI
Paging Zone Controller with 4-Port Audio Out
The CyberData VoIP Zone Controller with Audio-Out enables access to existing paging speakers through a VoIP phone system. The interface is designed to use a standard paging amplifier with audio inputs and supports paging up to 15 zone groups from a VoIP phone.
Operating the VoIP Zone Controller:
Note: Group 00 is configured to Page All Zones.
Ampliación de garantía a 3 años para la Switchvox AA305
Description:
The Quantum Pro Noise Cancelling headset is sleek, stylish and right at home in the most sophisticated of offices. Better still, it’s as hardworking as it is good looking with a range of state-of-the-art features and a reputation for durability. The simple push-button ADDLOCK makes changing from headband to earhook a convenient one-handed operation. A 270° adjustable microphone boom allows for easy flipping from ear to ear while its super flexible rubber design ensures a perfect fit. All-day comfort is further implemented by the headset’s exceptional lightness and a self-aligning, floating earpiece that sits easily on the ear without pressing against the user’s head. Sound quality is crisp, clear and natural thanks to the use of a broader bandwidth for the speaker combined with an innovative speaker cabinet. While a bidirectional microphone provides an exceptionally high signal-to-noise ratio to acoustically transduce the user’s voice.
2-year replacement warranty
The new ADDCOM Performance Plus II has retained all the great features of the original Perfor mance Plus and is enhanced with exceptional listening clarity and voice articulation. With a super flexible microphone boom, reinforced speaker cabinet, steadfast clickstop boom arm and the use of stress and scratch resistant materials, the Per formance Plus II has been designed to withstand the toughest of call centre environments. The lightweight design ensures low operator fatigue all day. The Per formance Plus II also allows the user to find the perfect microphone and headband positions. With our unique steadfast clickstop design, you can be confident that both headband and boom arm positioning will remain all day long without the need for constant re alignment.
ADD-T10 ADDCOM Dialler Product Code: ADDT10 Description: The ADDCOM Dialler has a refined and elegant look to suit a call centre, general work place or home office and eliminates the need to purchase additional telephones or amplifiers.
ADDCOM headsets may be connected directly to the Dialler and there is an additional extension jack allowing for an extra telephone or modem.
FEATURES
Switch box ADDCOM The ADDCOM Switch Box allows you to easily switch between your headset and handset. With the convenient mute switch allowing full control over your headset in one tiny box. Designed small to reduce the clutter on your desk. FEATURES Headset/handset switch for telephones without a dedicated headset socket Receive volume control Mute switch Small size ensures that there is less clutter on your desk Effortless way to connect both headset and handset to your phone Polarity switching to ensure compatibility with most phone systems COMPATIBILITY ADDCOM Range of headsets Most telephone systems PART NUMBER ADDCOM Switch Box ADD-313T
Curly Cord - QD to RJ11 (Grey)
Part Number: ADDQD-01
This cord works with any Addcom headset. It works as an adaptor to transfer the QD connection to an RJ11 connection.
* QD to RJ11 adaptor
NOTE: Compatible with all Addcom headsets.
Please check compatibility for telephones as this cable is not supported for every telephone system.
Curly Cord - QD to RJ11 (Red)
Part Number: ADDQD-02
NOTE:
Compatible with all Addcom headsets.
Addcom QD to 2.5mm Plug (ADDQD-06)
The Addcom QD to 2.5mm Plug for Panasonic and LG is a curly corded adaptor cable for headsets and telephones.
* QD to 2.5 millimetre adaptor
Straight Cable QD to 2 x 3.5mm Audio Sockets Product Code: ADDQD-08
The Addcom QD to 2 conectors of 3.5mm Plug for PC (Mic+Audio). It is a curly corded adaptor cable for headsets and telephones.
*QD to 2 x 3.5mm Audio Sockets
Addcom Geni Cord (ADDQD-14)
The Addcom Geni Cord allows the user to relieve the use of multiple cords to their telephone systems that are in use. Addcom created this one cable that will work with 98% of telephone systems that are on the market today. It is feature packed with a simple mute button, 2 position compatibility switch, 3 position mic gain switch and an easy to use receive volume scroll wheel.
Ampliación de garantía a 3 años para la Switchvox AA65
Garantia Extendida 3 años para AA355
With dual 40 mm speakers that deliver rich, full-range digital sound, the .Audio 655 offers exceptional performance for your PC audio needs. A lightweight design and pillow-soft ear cushions give you the comfort you need for your music and gaming, while the noise-canceling microphone with adjustable boom provides clear conversations without distractions, letting you make crisp-sounding Internet calls.
Cable Telco RJ-21 + PathPanel 24 Puertos
Cables Telco son ideales para teléfono, PBX y aplicaciones del sistema de telefónico IP.
Construido con conductor sólido, cable clasificado, carcasa metálicas de servicio con macho RJ21, 50 pines, al otro extremo conectorizado con PathPanel de 24 puertos RJ11 con color codificado normalizado.
Cross device. Cross platforms. Cross applications. That's Savi Office, the next-generation headset system that lets users connect to multiple communication applications and devices--desk phones, PC softphones, and PC audio--with a single headset. With a touch of a button, professionals can connect a softphone call on a PC with a desk phone call and then attend a Webinar. And thanks to its noise-canceling microphone, wideband PC audio, and integrated DECT technologies, Savi Office offers lifelike fidelity with every call and application and lets users roam up to 350 feet from their desk without compromising on clarity. With Savi Office, real-time collaboration has been redefined.
Gigaset C300H
O Gigaset C300H é um telefone de extensão do Gigaset C300. Com isso, você pode se comunicar economicamente. Ele permite uma grande variedade de funções através de seu menu e suas teclas de função e possui qualidade de som HSP (High Sound Performance).
Auricular Jabra Pro 9465
O hearset Jabra Pro 9465, pertence a uma série de headsets sem fio premium com opção de conexão com até três telefones simultaneamente, o que optimiza a productividade do usuário.
Cable Plantronic gama H QD a 2.5mm
Snom PA1: Public Address System
Snom PA1 is an IP paging device with useful functions. It can be used at office floors, reception areas, waiting rooms in announcements in airports, train and bus stations and waiting lounges. You can use too for monitoring of security-sensitive environments. You can use Snom PA1 with a loudspeaker as a simple PA system or with a microphone as a two-way intercom. The Snom PA1 can be used in both small and large applications due to the inclusion of a 4-watt amplifier for single speaker projects. Larger projects can utilize multiple speakers driven by external high wattage amplifier(s).
Applications for the Snom PA1 include:
Quick Summary
Curly Cord - QD to RJ11 (Black)
Part Number: ADDQD-04
Inline Mute Switch Product Code: ADD-1015
Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress voice signals, to save on bandwidth requirements. While voice signals from the Public Switched Telephone Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of different voice codecs including such as G.729, G.726, AMR, G.722, iLBC, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, but this functionality often requires digital signal processing tasks that are costly, resource intensive and can affect the quality of the voice signals if it introduces too much latency and delay.
The D500 card converts numerous simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows running up to 2000 sessions of any-to-any voice codec conversion, with unmatched quality1. All codecs are fully indemnified; no additional licensing is required for their use2.
The D500 works with both Asterisk and FreeSWITCH. With compatible drivers offered by Sangoma, these applications can use the D500 cards as seamless voice transcoding resources. Alternatively, developers and integrators can use the Transcoding API in C for their own application development.
Alphatech KeyPad for Door Phone SLIM
Code keyboard "SlimKeyBoard".
This keyboard manage widely satisfy your need management codes for competence entry into object. Code keyboard can work separately or in connection with SlimDoorPhone. Separate code keyboard is connect on source 12V (DC/AC) and contain one double throw contact. To control relay is possible use as far as 10 codes.
Further is equipped input for Exit button and switch-on/switch-off heating on board. keyboard is programmes by force of keyboard, or by force of USB cable from PC. At connection keyboard with doorphone SlimDoorPhone by special cable connect to programming connector. Code keyboard then makes it possible to dial phone numbers from SlimDoorPhone, door phone hang up, by Exit button control relay in SlimDoorPhone and use by other 10 codes to control relay in SlimDoorPhone.
NOTE: Recommended USB Cable programming
Technicolor TG672
The TG672 is a wireless VoIP router, with one RJ-11 WAN port, one 10/100/1000Base-T WAN port, one Gigabit Ethernet LAN port, 3 Fast Ethernet LAN ports, 1 FXO analogue port, 2 FXS ports for phone or fax and two USB ports. The router provides 3G back-up WAN connection via USB modem adapter. The integrated access point complies with IEEE 802.11n standard, providing wireless transmission speeds of up to 300Mbps. It supports Multiple SSIDs and uses WMM (QoS) to prioritize the traffic over the network. A WPS button allows for easy wireless security configuration. The router provides a variety of VoIP features, such as Caller ID, Echo Cancellation and 3-way conferencing. The TG672 has a built-in SPI firewall that protects the network against hackers and Denial of Service (DoS) attacks.
The Technicolor TG672 is an Ultra BroadBand business gateway, designed for Corporate/SMEs and SOHOs and offers plenty of possibilities through a complete set of business features. The Technicolor TG672 is a one-box office solution for communicating to the outside world, offering security to the office, allowing to share resources (like printer, hard disk), functioning as a PABX, while being able to be managed by the operator. It also has integrated DECT: DECT handsets supported by the gateway are the Technicolor TH58 and TH52.
Flexible Solution:
The Technicolor TG672 is an Ultra BroadBand business gateway, designed for corporate/SMEs and SOHOs and offers plenty of possibilities through a complete set of business features. The Technicolor TG672 is a one-box office solution for communicating to the outside world, offering security to the office, allowing to share resources (like printer, hard disk), functioning as a PABX, while being able to be managed by the operator. It also has integrated DECT: DECT handsets supported by the gateway are the Technicolor TH58 and TH52.
Features at a Glance ? Universal Ultra Broadband business gateway (supporting as well ADSL2+, VDSL2+, Fiber networks via Gigabit Ethernet Uplink, 3G via USB dongle) ? Embedded DECT: CAT-iq ready ? Secure communication device, with integrated IPSec client (compatible with major VPN servers), corporate firewall and content filtering. ? Dynamic routing support, including RIP, BGP and OSPF ? 3G back-up ? Extensive operator management capabilities (SNMPv3, TACACS+, Syslog) ? Standard built-in access point based on 802.11n ? Easy install / use, embedded firewall, QoS, TR-69 ? Direct connection of printer, hard disk, ... via USB port for office sharing ? SOHO/SME PABX functionality (with integrated SIP server/ Back to Back User Agent) ? Non-Service-Affecting SW upgrades, through dual bank memory configuration
ALPHATECH IPDP Slim RFID completes its range by developing a new generation of door phone based on SIP protocol.
The IPDP Slim RFID is an IP video door station with one pushbutton, a camera and 125kHz miniature RFID reader module integrated inside. Besides the standard functionality of the IPDP Slim, the IPDP Slim RFID door station enables to use EM Marin or HID proximity cards or tokens for door entry access control.
No more wiring. Now, just a network cable is needed for audio, video and data. The IP Door Phone IPDP SLIM naturally connects to your IP network.
In addition to voice, the IP Door Phone IPDP SLIM is able to broadcast the video stream to IP phones or softphones supporting video codecs H.263 / H.264. You can control the door from your telephone or computer directly.
The IP Door Phone IPDP SLIM offers a great flexibility and integration. To avoid wiring, the IP Door Phone IPDP SLIM can interface with a WiFi.
Power over Ethernet (PoE) For network designers and administrators, PoE simplifies the task of powering devices in remote locations - no dependency on AC outlets.
The RFID reader module is avaialble in two versions:
1) AREM 57U-EM reader module (more info in the datasheet, see links below). With this type of module the IPDP Slim RFID is used as a standalone version, programming via master cards, up to 748 users, EM Marin or HID cards.
2) MREM 57U-EM reader module. With this type of reader module the IPDP Slim RFID 125kHz can be used as a network version, you can store archive of events, possible connection of external IP camera, recording images, possible connection on a bus with up to 32 other APS Mini reader modules. Programming via master cards or PC (optional APSLAN 485/Ethernet converter required), up to 748 users, EM Marin or HID cards.
Detailed product description of RFID module integrated inside the IPDP Slim doorphone is here. We use a miniature RFID reader produced by Techfass company, a Czech based access control developer.
* Validated on several IPBXs (Cisco Call Manager, Alcatel Omni PCX, Asterisk, Nexspan, Panasonic, etc.), the IP Door Phone IPDP SLIM integrates easily with SIP v2.
Tarjeta RFID EM blanca 125 KHz para control de accesos
The RS458 adapter to LAN specifacally developed by TechFass for the integration of RFID APS Mini lectors in control's application with APS Home and APS Reader configuration.
The RD485 bus allows to connect up to 32 devices in the same bus with a maxim distance of 1200m, everything controled with an unic PC in local or connected to the Internet.
Montage in DIN
Power supply for Alcatel Temporis IP range. All Alcatel Temporis range of IP-PoE and come standard with no external power supply. Sometimes when switching does not support PoE or PoE injector to use, requires the optional power supply. Compatible terminals: * Alcatel Temporis 200 IP * Alcatel Temporis 600 IP * Alcatel Temporis 800 IP AC Adapter AC100-240V input, output 5Vdc/1.2A
Snom Meeting Point - Optimize your conferences with Snom
The Snom Meeting Point is a device specifically designed to optimize your company meetings. It is designed for use in medium-sized conference rooms and large rooms thanks to its noise cancelling and HD sound quality.
For users who remain in the room, the device comes equipped with microphones with full duplex broadband audio offering a quality sound making the conversation more dynamic and close.
As a product of Snom, this conference equipment presents the advantages and flesibility of this manufacturer such as the security of their CIT and SIP identities and the easy integration of their VoIP infrastructure.
PLX QD to ADDQD Converter (ADDQD-09)
GN QD to ADDQD Converter (ADDQD-15)
Adaptor GN QD to Addcom QD converter.
Ear Foam Performance Plus II
Product Code: ADD-1012
Product Code: ADD-1013
The KX-TGP500 series brings together all the advantages of modern internet HD VoIP voice calls, classic business phone functionality and easy Web based administration, ideally suited to small office and branch office solutions. With a high quality music-on-hold and answering Services, you won’t miss important calls, thanks to this phone’s integration with carrier class services. New messages are displayed on the handset, so you’ll know someone tried to reach you even if you’re away from your desk. For added convenience you can check on messages from wherever you are, or away from the office, with the answering service alerting you when you receive a new message via SMS. The TGP500 series devices support High Definition Sound performance. This superior wideband audio quality lets you hear every detail in the other person’s voice when they speak.
* Telephone base unit + KX-TPA50 cordless handset * SIP Cordless phone system * HD Wideband Audio * Easy Web configuration * Large, backlit LCD display * Supports up to 6 DECT cordless handsets
*Some of the services above may vary depending on your carrier or the services that you have subscribed to
Alphatech Door Phone SLIM SDP 01 can be connected to PBX FXS or analog PSTN line.
Alphatech NUDV Door Entry Phone Station 1 button. NUDV-01
Alphatech NUDV Door Entry Phone Station to use with analog line. Modular system allows to connect 1 to 64 buttons. Up to two 16-digit numbers, including “ r ”, “ # ”, Pause and Flash in tone dialing, can be programmed to each button. The universality lies in possibility to connect this guard to an internal line of your branch exchange regardless to type and producer of this exchange (analog line). 3 Models:
Further the whole system allows to be enlarged by NC-mod4 and NM-mod4 modules up to 64 buttons using the basic mechanical. The whole assembly can be completed with cover frame or rainprotective canopy.
The guard is supplied from branch exchange line – remind a loud telephone. The basic features include the possibility to open up to two doors by means of connected electrical locks Features:
Alphatech NUDV Door Entry Phone Station 2 buttons. NUDV-02
MADRID: 17-21 de Noviembre de 2014
Digium Asterisk® Advanced (MADRID) SIN MATERIAL*
Dirigido a técnicos que ya tengan un mínimo de experiencia propia teórica y práctica con Asterisk y que quieran acelarar de forma efectiva su proceso de aprendizaje para llevar el conocimiento sobre Asterisk hasta un punto que permita llevar a cabo integraciones de sistemas en producción
Conocimientos Previos:
Programa (temario oficial Digium Training Asterisk® Advanced):
Programa:
Asterisk versión: Asterisk 1.8
Horario:
Lunes a Jueves de 09:30 a 18:30 Viernes de 09:30 a 11:30
* ESTE CURSO ESTA DISPONIBLE CON MATERIAL
Posteriormente se realizará el examen oficial que le certifica como profesional Asterisk: dCAP Digium Certified Asterisk Profesional (no incluido en el curso)
Para realizar el examen dCAP, sin asistencia al curso pulse aquí.
Cyberdata analog speaker 011120
This Auxiliary Speaker is an option for the Cyberdata SIP speaker to extend the coverage area.
When connecting this Auxiliary Analog Speaker to the SIP speaker, the total speaker wattage is the same. However, by adding this Auxiliary Analog Speaker, additional coverage can be realized.
This Auxiliary Analog Speaker can be mounted with the same mounting hardware options available for the SIP Speakers.
Cyberdata analog speaker 011121
Expansion microphones for Snom Meeting Point
Additional microphones for the Snom Meeting Point that allow you to extend the audio radio of your conference terminal to 70m2. The OmniSound software also detects additional microphones and the working range or the Snom Meeting Point.
The KIRK Wireless Server 6500 solution is a rack version that consists of a number of different infrastructure elements which can be customized in accordance with your exact telephony needs today and later adjusted to suit any future changes in your organization.
Seamless handover between base stations, extensive radio coverage, messaging to handsets and value-added applications are just some of the benefits of the KIRK Wireless Server 6500.
Customize Your Solution
Up to 256 KIRK IP Base Stations and up to 4,096 wireless users can be subscribed to the KIRK Wireless Server 6500, making it extremely scalable and ideal for growing with your organization. A flexible license option allows you to only pay for the users you need. Should your business expand, you simply add more mobile users.
The KIRK Wireless Server 6500 shares a lot of similarities with the existing KIRK Wireless Server 6000, but it is built on a new platform with capacity to ensure a continued rapid product development. The KIRK Wireless Server 6500 solution is not only able to handle future features, but in time it will also be able to provide radio coverage of a much larger geographical area than the KIRK Wireless Server 6000. That is why the KIRK Wireless Server 6500 is a good match for larger businesses and enterprises.
You can deploy the KIRK Wireless Server 6500 as a redundant solution with automatic failover, which gives you the opportunity to secure your business communication.
A KIRK Wireless Server 6500 solution consists of the KIRK Wireless Server 6500 itself, KIRK Media Resources, KIRK IP Base Stations, KIRK Repeaters and the KIRK Handsets.
More details and documentation in https://support.spectralink.com/products/dect/spectralink-ip-dect-server-6500
NanoBracket Universal
NanoBracket™ Universal – holder for NanoStation and other popular CPEs
NanoBracket™ Universal is a holder, designed for the installation of the most popular CPE devices on the market - NanoStation /both older and newer version/, NanoStation Loco, NanoStation M, NanoStation Loco M from Ubiquiti Networks, StationBox and StationBox Mikro from RF elements, Airmax from Airlive, and TP-Link TL-WA7510N and TL-WA5210G. NanoBracket™ is original design and development of RF elements.
NanoBracket™ Universal is made of durable UV-stabilized ABS plastic and allows easy and quick installation of the device on a wall, pole or a console. Installation of the NanoBracket™ Universal is really simple.
The holder is mounted in the closest possible position to the center of gravity of the mounted device, what makes the construction more stable. The color of the holder match perfectly with the color of the NanoStation and StationBox, therefore the construction looks really compact.
Compatibility
Using Body A (1) suitable for:
UBNT NanoStation M5™, UBNT NanoStation M2™ UBNT NanoStation™ 5, UBNT NanoStation™ 2 AirLive® Airmax 5, AirLive® Airmax 2 TP-LINK® TL-WA7510N, TP-LINK® TL-WA5210G RF elements™ StationBox™
Using Body B (2) suitable for:
UBNT NanoStation 5 LOCO™ UBNT NanoStation 2 LOCO™ UBNT NanoStation Loco M5™ UBNT NanoStation Loco M2™ RF elements™ StationBox™ Mikro
Con un brazo de micrófono flexible y calidad de sonido superior junto con el uso de materiales resistentes a la tensión y arañazos, la serie ADD300 ha sido diseñado para soportar el más duro de los entornos de centros de llamadas.
El diseño ligero asegura una baja fatiga del operador durante el día. La serie ADD300 también permite al usuario encontrar el micrófono y diadema en las posiciones adecuadas. Con nuestro diseño robusto ClickStop reforzado, puede estar seguro de que tanto diadema y la posición del brazo del micrófono se mantendrán durante todo el día sin la necesidad de una alineación constante por parte del usuario.
With the evolution of cable standards, DOCSIS/EuroDOCSIS 3.0 will prove decisive for cable operators wishing to provide high bandwidth data delivery and to stay ahead of the content with game and video over IP delivery
With this new generation of High Speed Data Services Solution, Technicolor introduces up to eight (8) bonded downstream channels and four (4) bonded upstream channels, with a design backwardcompatible for use as a single-channel with older networks without any service interruption. Thanks to the integrated dual split tuner solution, the operator will benefit from flexibility in channel allocation: to ease the management when the channel frequency plan is loaded or to separate data from other services such as video over IP.
The new TCM471 comes in the sleek attractive design Technicolor has adopted as the new "family look".
Cargador simple para los terminales Spectralink 7620 y 7640
The Mediatrix C7 Series offers a wide variety of configurations combining FXS and FXO interfaces for multiple applications in a single platform. It is available with 4 and 8 telephony ports and connects up to 8 analog phones, modems, or fax machines; or connects up to 8 PSTN or PBX trunk lines to an IP networ
The Mediatrix C7 Series VoIP adaptors are high-quality, cost efficient VoIP gateways connecting small to medium branch offices to an IP network, while preserving existing investments in analog devices and linking IP extensions with the PSTN for local toll services or survival scenarios.
KIRK Wireless Server 400
The KIRK Wireless Server 400 is an excellent choice for small to medium-sized businesses (SMBs) that want a simple and flexible wireless solution. It is a scalable SIP solution that supports up to 30 wireless users and 12 channels - depending on how you assemble it and which license keys you add to it.
The KIRK Wireless Server 400 can be deployed as either a single-cell or a multi-cell solution. This scalability ensures that the KIRK Wireless Server 400 can grow with your business when needed. You can upgrade the KIRK Wireless Server 400 by adding license keys. With the license keys you can design the solution to support your exact business requirements.
Today mobility among employees is important, and therefore KIRK Wireless Server 400 is designed so it can be installed in multiple locations. Furthermore it is compatible with all KIRK Handsets, and this gives all employees an opportunity to choose a handset that meets their individual needs.
The SIP-enabled IP Outdoor Intercom with Keypad is a two-way communication and secure access device and one of the newest in our line of Power over Ethernet (PoE 802.3af) and VoIP intercoms.
Combining the versatility of a SIP based keypad intercom with the increased weather protection rating of IP 65, this device is perfect for settings such as commercial/ residential facilities, schools and universities, retail establishments, warehouse and manufacturing plants, parking garages and shipyards, and so much more.
Note: The optional Weather Shroud is sold separately.
Sangoma Netborder Carrier SBC 4000 calls
Security and flexibility are the main added values of this Netborder Carrier SBC of Sangoma. Up to 4000 calls, this SBC is ideal as an intermediary between devices, networks or protocols and as a barrier with potential threats (internal or external) ensuring their IP infraestructure.
Grandstream GXW4216
GXW4216 is a next generation high performance high-density analog VoIP gateway that is fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the the market. It features multiple FXS analog telephone ports, superb voice quality, rich telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection, and strong performance in handling high volume voice calls. The GXW42XX series gateway offers small and medium businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.
Grandstream GXW4224
GXW4224 is a next generation high performance high-density analog VoIP gateway that is fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the the market. It features multiple FXS analog telephone ports, superb voice quality, rich telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection, and strong performance in handling high volume voice calls.
The GXW42XX series gateway offers small and medium businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.
Grandstream GXW4232
GXW4232 is a next generation high performance high-density analog VoIP gateway that is fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the the market. It features multiple FXS analog telephone ports, superb voice quality, rich telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection, and strong performance in handling high volume voice calls.
Grandstream GXW4248
GXW4248 is a next generation high performance high-density analog VoIP gateway that is fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the the market. It features multiple FXS analog telephone ports, superb voice quality, rich telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection, and strong performance in handling high volume voice calls.
Technicolor TG789vn v3
The Technicolor TG789vn v3 is a unique future-proof triple-play service gateway allowing VDSL2 connectivity while providing Voice over IP functions for residential users.
A dedicated Gigabit Ethernet WAN port and WAN Port Auto-Sensing make the Technicolor TG789vn v3 also the ideal service gateway for deployment in mixed DSL and fiber-based access networks.
The Technicolor TG789vn v3 offers next to four Fast Ethernet LAN ports, a IEEE 802.11b/g/n wireless LAN Access Point. Moreover, it acts like a central hub for distribution of all content from any device to any device in the home. You can stream music, data, pictures and video from your gateway to devices connected to your wired or wireless home network.
The Technicolor TG789vn v3 is easily manageable thanks to its best-in-class TR-069 interoperability.
IP Phone Expansion Module
EXP38
The Yealink EXP38 Expansion Module has been designed to improve the power and flexibility of advanced Yealink IP phones T28P y T26P.
It features a 38 fully-programmable DSS key search function, each with a dual-color LED. The module is connected to and controlled by the IP Phone with an RJ-12 cable line. Up to 228 additional programmable extensions are created when six EXP38 units are daisy-chained together with the IP Phone.
Yealink’s advanced IP phones also have IP-PBX support functions, such as speed dialling, and BLF/BLA, intercom, call forward/transfer/hold/park/pickup/return via programmable EXP38 buttons. The Yealink EXP38 is ideal for receptionists, administrative assistants, call-center agents, power-users and executives who need to monitor and manage a large volume of calls on a regular basis. Features include: -
* 38 programmable keys, each with a dual-color LED * Creates up to 228 programmable keys when 6 EXT are daisy chained * Supports the functions for IP-PBX such as BLF/BLA and Intercometc
Alphatech BELL rain hood
The BRI power supply (SPEC-BNPWR1) is a power supply unit that is used to power BRI phones or other BRI devices connected to our BRI card. This is only for the BRI card in NT mode with TE devices connected to it. When the Sangoma card is in TE mode this device should not be used.
The power supply unit is powered with a molex connector (from internal/external PSU), the output of the unit will be a connector that will plug into the green connector on the Sangoma card. See image below for the location and picture of the green connector.
As cable standards keep evolving, Technicolor presents the TC7200, a EuroDOCSIS 3.0 wireless Embedded Multimedia Terminal Adapter (EMTA) gateway. With this new offer Technicolor provides cable operators with high-bandwidth data transmission and enables them to stay ahead of content delivery with gaming and video over IP.
This generation of high-speed data service solutions offers up to eight bonded downstream channels and four bonded upstream channels, allowing operators to provide their customers with four data channels and four channels for dedicated IPTV services.
The TC7200 also provides a powerful platform to port new applications such as DLNA media server and hard disk sharing.
Cyberdata VoIP Intercom + keypad 011123 (Flush mount)
Our extensive line of indoor and outdoor SIP-enabled intercoms delivers two-way communicationa and secure access control for your VoIP phone system. These devices are perfect for settings such as commercial / residential facilities, schools and universities, retail establishments, warehouse and manufacturing plantas, SMB, parking garages and shypyards and so much more.
Adapter cable--3.5mm to Quick-Disconnect Cable for iPhone
Media Gateways Digium G400
Four Span Digital T1/E1/PRI to VoIP Gateway Appliance (Europe)
The G400 is a four span T1/E1/PRI gateway that provides up to 120 concurrent calls of TDM-to-SIP, SIP-to-TDM or SIP-to-SIP media convergence. It includes integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.
Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for Asterisk connectivity. The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations. The G400 VoIP Gateway includes four software-selectable T1/E1/PRI interfaces and supports up to 120 concurrent calls. It is built to support TDM-to-SIP, SIP-to-TDM and SIP-to-SIP (transcoding) applications. In a TDM-to-SIP deployment the VoIP gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the VoIP gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.
Digium VoIP Gateways are flexible solutions that fit many communications applications. The applications listed below are the most widely used. These gateway appliances support a wide range of applications, due to the flexible configuration options and standards-based connectivity.
OptiCaller is an application for Mobile PBX with unique call optimization. It allows the user to make calls in a flexible, and above all, more cost-effective manner (average 40-95% cost savings). OptiCaller also makes it easy to manage PBX functions, e.g. call diversions and presence status, directly from the mobile phone.
The architecture is operator independent, which means that the application works regardless of mobile operator. The application has very flexible configuration options and is adaptable to suite the majority of PBX systems on the market. Configuration and deployment of clients (OTA) is easily handled by a powerful provisioning system.
OptiCaller always takes control over how each call is connected and is thus giving the user the opportunity to call in the most cost efficient way. The call can be connected in three different ways
The call method can either be predefined or decided at each call with our Always Ask functionality. It is also possible to create rules that control the call method.
OptiCaller is compatible with Asterisk and many more.
Base de carregamento
Comprimento do cabo: 90cm. Não inclui adaptador CA (900102125)
Número de produto: 900102094
The CHAT® 50 personal, USB-powered speakerphone is a mobile audio peripheral that connects to a wide variety of devices, providing crystal-clear, hands-free audio for ad-hoc conferencing or audio playback. The small form factor is sleek enough for desktop conferencing, yet rugged enough to be completely portable. Hands-free laptop and cell-phone conferencing are possible any place, any time.
ClearOne’s market-leading HDConference™ audio technology provides crystal-clear conferencing sound:
Sensitive microphone array provides 120-degree audio pickup up to 8 feet away
*Additional accessories can be purchased separately
The Skype Certified CHAT 60 speakerphone is created expressly for Skype PCs and unified communications software. Rugged and portable, the CHAT 60 is designed for ease of use and unsurpassed voice quality in the home office or on the move with laptop USB power. Perfect for personal use, teleworkers, traveling professionals or small office desktop, the CHAT 60 allows you to conference any time, any place.
Complete Care to solution Collaborate Room HD (1 year) Equipment and software manufactured by ClearOne. Repair and Replacement Service During the term of the purchased Agreement, and subject to the limitations in this Agreement, ClearOne will repair or replace the Equipment as necessary to correct any manufacturing defects in the Equipment which occurs during the usual and customary usage of the Equipment during the Service Agreement period. If ClearOne repairs your Equipment, you understand and agree that ClearOne may replace original parts with new or like new parts. Replacement parts will be functionally equivalent to the original parts. If the product is found to contain a manufacturing defect that is un-repairable, ClearOne may choose, at its sole option, to provide a new or factory certified, same or better, model as a replacement. Technical Support ClearOne will provide access to ClearOne’s Global Support Centers to assist with hardware and software product use, configuration and troubleshooting. ClearOne will use reasonable efforts to respond to you during ClearOne’s published support hours. Please refer to our website at https://www.clearone.com/customer_support for a complete listing of ClearOne Global Support Centers and their hours of operation.
Complete Care to solution Collaborate Room HD (1 year)
Complete Care to solution Collaborate Room FHD (1 year) Equipment and software manufactured by ClearOne. Repair and Replacement Service During the term of the purchased Agreement, and subject to the limitations in this Agreement, ClearOne will repair or replace the Equipment as necessary to correct any manufacturing defects in the Equipment which occurs during the usual and customary usage of the Equipment during the Service Agreement period. If ClearOne repairs your Equipment, you understand and agree that ClearOne may replace original parts with new or like new parts. Replacement parts will be functionally equivalent to the original parts. If the product is found to contain a manufacturing defect that is un-repairable, ClearOne may choose, at its sole option, to provide a new or factory certified, same or better, model as a replacement. Technical Support ClearOne will provide access to ClearOne’s Global Support Centers to assist with hardware and software product use, configuration and troubleshooting. ClearOne will use reasonable efforts to respond to you during ClearOne’s published support hours. Please refer to our website at https://www.clearone.com/customer_support for a complete listing of ClearOne Global Support Centers and their hours of operation.
Complete Care to solution Collaborate Room FHD (1 year)
COLLABORATE_Complete_Care_Service_Agreement
Media Gateways Digium G800
Eigth Span Digital T1/E1/PRI to VoIP Gateway Appliance (Europe)
The G800 is a eight span T1/E1/PRI gateway that provides up to 240 concurrent calls of TDM-to-SIP, SIP-to-TDM or SIP-to-SIP media convergence. It includes integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.
Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for Asterisk connectivity. The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations. The G800 VoIP Gateway includes four software-selectable T1/E1/PRI interfaces and supports up to 240 concurrent calls. It is built to support TDM-to-SIP, SIP-to-TDM and SIP-to-SIP (transcoding) applications. In a TDM-to-SIP deployment the VoIP gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the VoIP gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.
The beroNet 16FXS Gateway is designed to simplify the integration of legacy phone system to IP infrastructure. It enables you to connect 16 analog devices (phones or fax) to an IP network.
The beroNet 16FXS Gateway is used by firms, hostels or hospitals who need internal Analog FXS ports to connect lots of analog devices.
The Voyager Legend headset offers unmatched sound quality and user experience comfortable all day.
This headset combines three microphones with noise cancellation and wind. It incorporates voice commands and detects when located in the ear for the off-hook call automatically.
Vega100: E1/T1 Digital Gateway Sangoma Media Gateway VS0164
The Vega 100 VoIP gateways connects digital telephony equipment to IP networks. Each E1/T1 interface can be independently configured as network side or terminal side.
The Vega 100 gateway support the following signalling schemes:
All Vega gateways support SIP, H.323 & T.38 FAX.
The 100 gateway has proven interoperability with a wide range of existing telecommunications & VoIP equipment.
Identification
Operations, Maintenance & Billing
Routing & Numbering
Security & Encryption
Call Quality
Vega100: E1/T1 Digital Gateway Sangoma Media Gateway VS0157
The Vega 200 VoIP gateways connects digital telephony equipment to IP networks. Each E1/T1 interface can be independently configured as network side or terminal side.
The Vega 200 gateway support the following signalling schemes:
The Vega200 gateway has proven interoperability with a wide range of existing telecommunications & VoIP equipment.
Bytton DS (Dual SIM) Ref: BYT_LTE_D
Bytton family is an intelligent services platform which provides high data transfer rates over LTE 4G/3G+ Networks.
Suitable for mission critical applications such as multi facility corporate connectivity, professional backup and M2M applications:oil pipeline monitoring, backup routing for fiber optics, video surveillance, security applications.
Bytton LTE Intelligent Services Gateway includes a large range of communication interfaces and protocols including backward capability 4G, 3G+, 3G & 2.5G network coverage, WAN, and RS232/RS485
SMS Read / Send You can send and receive SMS messages from the Web interface, using the modem module and the one or two SIM cards used on the equipment
BENEFITS:
Technology Brief
GSM, GPRS, EDGE, UMTS, HSDPA, HSUPA, HSPA+, LTE, Ethernet, USB, RS232 and/or RS485 (optional), VPN, FXS (optional), WiFi
LTE : 800/900/1800/2100/2600 MHz or 7000 MHz
HSPA: 850/900/1900/2100 MHz HSPA
Benefits
Applicability
Applications:
Para qué sirve este producto El switch PoE de sobremesa con 8 puertos a 10/100 Mbps TL-SF1008P permite conectar entre sí dispositivos de red de un modo sencillo. Incorpora puertos compatibles tanto con Fast Ethernet a 100 Mbps como Ethernet a 10 Mbps.
Cómo funciona 4 de los 8 puertos RJ45 con negociación automática (del puerto 1 al 4) del switch soportan el modo Power over Ethernet (PoE). Estos puertos PoE detectan automáticamente la presencia de dispositivos PoE compatibles con el estándar IEEE 802.3af y les suministran alimentación eléctrica. Con la tecnología PoE, la energía eléctrica se transmite junto a los datos a través de un solo cable, lo que permite llevar la red a aquellos puntos en los que no se disponga de conexiones o tomas de corriente eléctrica. De este modo, podrá instalar de un modo permanente dispositivos tales como puntos de acceso, cámaras IP, teléfonos IP, etc.
Protección contra sobrecargas El TL-SF1008P dispone de una función de prioridad* que ayuda a proteger el equipo cuando se produce una sobrecarga en la alimentación del sistema. Si el consumo de todos los equipos PoE es mayor o igual que 53 W, se establece una prioridad de alimentación entre los puertos PoE en función de la cual el equipo interrumpe la alimentación en el puerto que posea una menor prioridad.
Función de prioridad en los puertos *Prioridad (puerto 1 > puerto 2 > puerto 3 > puerto 4): Esta función le ayuda a proteger el sistema si se produce una sobrecarga. Por ejemplo, los puertos 1, 2 y 4 están utilizando 15,4 W (el nivel máximo de alimentación por puerto es de 15,4 W) por lo que el suministro del sistema es de 46,2 W en total (el led que PoE max se ilumina en rojo). Si se conecta un dispositivo PoE que consume 10 W en el puerto 3, el equipo desconecta la alimentación del puerto 4 para protegerse. De este modo, los puertos 1 y 2 consumirán 15,4 W, el puerto 3 utilizará 10 W y no se suministrará alimentación al puerto 4.
Fácil de usar El TL-SF1008P es fácil de instalar y utilizar No requiere ningún tipo de configuración o instalación. El switch PoE de sobremesa con 8 puertos 10/100 Mbps TL-SF1008P de TP-LINK ofrece un rendimiento y calidad excepcionales por lo que resulta una excelente opción para ampliar tanto su red doméstica como la de su oficina.
The Butterfly configuration/programming interface unit is built into a USB cable.
Newly redesigned and simplified, the Sangoma B500 S/T BRI interface board delivers superior audio quality and scalability. It expands from two to four ports of BRI, with optional telco-grade hardware echo cancellation.
A single PCI Express slot hosts the connection for up to 4 ports and ensures common synchronous clocking for all channels with no signalling issues.
The B500 consists of a Remora BRI daughterboard mounted on the AFT PCIe board. The Remora BRI card has 2 sockets, each of which can accept an S/T BRI module. One S/T BRI module has two S/T four wire interfaces, which support TE or NT modes of operation.
The B500 has 4 RJ-45 ports, each port can handle one S/T four wire BRI interface with a single standard Cat-5e shielded twisted-pair cable.
Operating Systems
Windows® 2003, Windows® XP, Windows® Server 2008, Windows® Vista, Windows® 7
Production Quality
Cabo do carregador, micro USB com fonte de alimentação de parede para Spectralink Butterfly
Cable de conexion rapida para auriculares de la gama profesional de Platronic con terminación en RJ9
AirGrid M5: Revolutionary 5GHz CPE Technology Patent Pending InnerFeed™ Antenna Technology Utilizing InnerFeed technology, the new AirGrid M Series represents the evolution of outdoor broadband wireless devices. Complete antenna and radio system integration provides revolutionary cost/performance solutions to the Worldwide Broadband Industry. Robust and Simple Product Design Can be oriented to use either vertical or horizontal polarization. Mechanical design provides complete weatherproof performance. Acitivity and signal strength LED's provided for installers. Enhanced RF and Ethernet ESD/Surge protection enables prolonged operation in harshest environments. Breakthrough Wireless Performance, airMAX and AirControl Support 100+Mbps of real outdoor throughput and up to 30km+ range. AirGrid products utilize Ubiquiti's revolutionary airMAX™ TDMA protocol enabling scalable, carrier-class PtMP network performance. Additionally, AirControl™ application allows operators to centrally manage 100's of devices. AirGrid USB includes both a PoE injector and a passive adapter. Recommended ethernet cable lengths differ depending on which power supply is used: The passive adapter (POE+USB) will take 5V from any powered USB port. Maximum ethernet cable length is 65 feet (20 meters). The PoE injector (AG-POE-US) is rated at 5Vdc 2A and includes a 6-ft US power cord. Maximum ethernet cable length is 165 feet (50 meters).
Polycom KIRK power supply for 40xx handsets and repeaters
The SmartNode 10100 enables the delivery of VoIP services by bridging voice traffic between the public switched telephone network (PSTN)—based on time-division multiplexing (TDM)—and IP networks such as the Internet. Service providers are adding VoIP capabilities to their networks, whether to reduce costs when interconnecting with other carriers, to cost-effectively build out their network footprints, or simply to transport voice traffic across their IP backbones. Whether sitting at the network core or at the edge, SmartNode media gateways enable service providers to introduce VoIP into their networks while maintaining the quality and the reliability of traditional TDM networks.
TDM interfaces Service providers, whether providing local, long-distance or international voice services, are interconnected with a multitude of other providers using T1/E1/J1 links. It is critical for service providers to be able to rapidly establish new interconnections without having to always deploy new devices. SmartNode 10100 Series media gateways therefore offer flexibility and can be configured to support T1/E1/J1 interfaces.
Signaling and control protocols Just as flexibility in the selection and deployment of TDM links is a key requirement for service providers, the need to support multiple signaling protocols across various carrier partners is just as important. Each SN10100 media gateway provides support for the concurrent use of ISDN, SS7/C7, CAS (R2), SIP, and SIGTRAN signaling in the same device. The ability to provide both switching and conversion across multiple TDM and IP signaling protocols at once is paramount to enabling the operational flexibility and cost savings that drive service providers to expand their carrier relationships and converge their networks.
In parallel with the TDM and IP signaling protocols mentioned above, SN10100 devices also support the H.248 media gateway control protocol, which enables any H.248-compliant 3-party softswitch to control a media gateway. While the softswitch manages call control interactions, the SN10100 handles transmission of call media as well as any required transcoding.
Media handling Service providers will use one or more codecs on their VoIP networks according to their desire to save bandwidth, to provide a certain level of voice quality, or simply to interoperate with other VoIP devices or providers. The ability to support multiple different concurrent codecs and to allocate them in real time based on traffic is the key to delivering true network convergence.
SmartNode 10100 gateways feature extensive support for various wireline, mobile and IP telephony audio formats, delivering seamless transcoding in real-time. The media gateways ship with support for G.711, G723.1, G.726, and G.729ab right out of the box, with no additional license fee required. They also offer optional support for mobile and IP vocoders such as AMR, AMR-WB (G.722.2), GSM-FR/GSM-EFR, EVRC/QCELP, G.728, G.729eg, and iLBC. SN10100 gateways offer independent dynamic codec selection per channel. This means that it is possible to assign different vocoders to different channels, on a channel-by-channel basis. The devices can then run all of these codecs concurrently and do so with no impact on system performance.
SN10100 gateways also provide unparalleled support for Internet-based fax, also known as Fax over IP or Fax relay, using the T.38 protocol, which is used to carry fax communications over an IP network. (They also support the T.30 protocol for fax over the PSTN.)
SS7 license (Optional) The Patton SmartNode 10100 can be used with SS7 interconnection protocol, acquiring an additional license SS7:
System density SN10100 gateways feature the industry’s highest system density in a 1U form factor. Beside the capital savings achieved by purchasing less units of equipment, system density also provides operational cost savings in the form of reduced co-location fees as well as lower power and cooling costs.
Energy efficiency For many, if not most, service providers, the payoff from reducing energy use can be particularly impressive; typically, for every watt of power required to operate a device, another watt is required to cool it. The SN10100 media gateways can play a major role in reducing energy costs, with an average two-thirds less power consumption than competing products of similar capacity.
Provisioning and maintenance For network convergence efforts to contribute positively to revenue and profitability, service providers must maintain their reputation for uptime and availability during the introduction, operation, and maintenance of new services. The SN10100 offers OAM&P, an operations, administration, maintenance, provisioning (OAM&P) solution. OAM&P enables the service provider to perform the initial set-up of the SN10100 media gateway and any subsequent maintenance operations. These range from the simple, such as the collection of statistics and alarms, to the more complex, such as system configuration changes, the addition of new hardware or software components, and the application of software patches or software upgrades.
The A116 is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line – using high performance PCI Express interface, providing superior performance in critical systems all over the world. The A116 supports up to 32.8 Mbps of full duplex data throughput or 480 voice calls using 16 T1/E1/J1 spans.
With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The A116, like all cards in Sangoma’s AFT family, is field-upgradable with our unique unbreakable firmware feature.
Choose the Sangoma A116E and A116DE, equipped with world class DSP hardware to achieve carrier-grade echo cancellation and voice quality enhancement functions for telecommunication systems.
Typical A116 Applications:
The KX-A405 DECT repeater is an ideal solution for when you need to extend the range of your DECT CS (Cell Station), to cover areas where reception was previously not available.
The A405 DECT repeater extends the range in all directions, allowing for a wider area to be covered. A single cell station can register up to 6 A405 repeaters while each repeater can support 4 simultaneous transmissions channels.
Digium Rack mount ears 19" for gateways
Gateway Patton SN4324 - 24 FXS
SN4324/JS/UI
Overview The 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. Applications
Polycon 02334601
The KIRK Repeater is a building block to be used to extend the coverage area in a KIRK solution. The KIRK Repeater does not increase the number of traffic channels, however provides a larger physical spreading of the traffic channels and thereby increases the coverage area established with the KIRK Base Stations.
KIRK Repeaters are mainly used in areas with limited traffic. The KIRK Repeater is available with either 2 or 4 voice channels. It is wireless and does not need physical connection to the KIRK Wireless Server, making it very easy to install. The KIRK Repeater can be supplied with an external directional antenna, which makes it possible to create radio coverage in a remote area without cabling to the rest of the installation.
GXP2200EXT Extension Module
Grandstream GXP2200EXT is a extension module for GXP2200 phones which feature a 128x384 graphics LCD, 20 quick-dial or BLF keys with dual-color LED, 2 navigation keys, and less than 1.2W power consumption per unit.
Up to 4 GXP2200EXT for phone Grandstream GXP2200.
Compatible with:
IP Bell - VoIP SIP door entry phone station
VoIP Outdoor Intercom SIP
The CyberData SIP-enabled VoIP Outdoor Intercom is a Power-over-Ethernet (PoE 802.3af) and Voice-over-IP (VoIP) door entry device that easily connects into existing local area networks (LANs) with a single cable connection. The intercom is compatible with most SIP-based IP PBX servers that comply with the SIP RFC. Its tamper-proof design allows the unit to be mounted securely and safely.
NEW:
Newly designed with a faster processor allowing for more capabilities, and increased IP rating to IP64 for weather protection, the CyberData SIP-enabled VoIP V3 Outdoor Intercom is perfect for settings requiring two-way communication and secure access.The new Weather Shroud (pictured) can be purchased separately for even greater weather protection.
Technicolor TG784n V3 (VoIP+WiFi-n)
The Technicolor TG784n v3 is especially suited for complex network scenarios where several access technologies, such as ADSL and Fiber To The x (FTTx) co-exist. Thanks to its WAN Port Auto-Sensing feature the Technicolor TG784n v3 can automatically select among DSL, Ethernet WAN or 3G (via USB adapter) interfaces and connect without any manual operation. This means simplicity for the end user and reduced complexity for the operator.
With an improved coverage and throughput compared to any 11b/g or 1x1 11n access point, the Technicolor TG784n v3 enables real-time multimedia content streaming over the Wi-Fi network everywhere in the house. Its wireless range and coverage makes it the best wireless gateway currently on the market.
The Technicolor TG784n v3 is especially suited for complex network.
Cyberdata outdoor intercom v3 shroud
Patton SmartNode DTA SN-DTA/1BIS2V/EUI
The SmartNode DTA enables integration of ISDN network users into a local VoIP phone service, or extends an ISDN line of a PBX to a remote site over IP. It offers a simple end-user configuration interface and connects both to a PBX in point-to-point mode and an So bus in point-multipoint mode. The SN-DTA enables the connection of ISDN terminals or SOHO PBXs to a VoIP network or Internet Telephony Service. It can connect to ISDN PBXs in point-to-point mode and ISDN Terminals in point-multipoint mode (S-Bus) and offers a feature rich configuration interface. Like every SmartNode the DTA includes intelligent call routing technology with advanced features like number plan adaptations, mappings between ISDN and SIP/H.323, manipulation of call properties through regular expressions, routing calls based on time-of-day or bearer capability criteria and much more. Key ISDN services such as AOC, PARE, CLIP, COLP, etc. are mapped using industry standard methods to SIP and H.323. The SN-DTA supports SIP overlap dialing (RFC3578) for countries with dynamic dialplans. The integrated ISDN line-power feeds connected terminals in the same way as a legacy ISDN NT to enable a seamless switchover from an ISDN to VoIP access. Gateway functions use standard CODECs such as G.711, G.723, G.729, and T.38 fax as well as industry standard SIP and H.323 signaling protocols to ensure seamless connection and compatibility for all voice services. Comprehensive quality of service (QoS) features offer traffic classification and tagging options that include TOS, DiffServ and VLAN 802.1p/Q tagging.
Gateway Patton SN4120 - 1 BRI (TE) SN4120/1BIS2V/EUI
The SmartNode™ 4120 ISDN BRI PSTN Gateway converts two simultaneous phone or T.38 fax calls from SIP to ISDN BRI. It is a compact reliable standalone VoIP gateway for IP-based voice systems that delivers ISDN performance and quality. For system integrators looking to connect a VoIP or Unified Communication solution to public or private ISDN lines, the SmartNode 4120 provides unparalleled ISDN to IP feature preservation. Like any SmartNode the 4120 provides advanced ISDN functionality such as Explicit Call Transfer (ECT) support and Advice of Charge (AOC) over SIP. Patton CPEs are interoperable with most IP PBX and Unified Communication vendors such as Asterisk, 3CX, Elastix, Microsoft, IBM, Swyx, SNOM One and many others. Compared to PC-based solutions that combine IP PBX and a PCI BRI cards, the SmartNode has several advantages, including: installation without additional drivers or software, operation without ventilation or hard disk, scalable without limitation, and service integration without downtime. SmartNode gateways can be combined in a cluster for simple and flexible redundancy solutions not available with PCI BRI cards. System administrators don't have to learn ISDN as industry standard protocols like SIP do the job of connecting the PSTN to your VoIP system.
Available TWO models:
Cyberdata Conduit Speaker 011039
The conduit speaker mount is the newest option for installation of the Cyberdata VoIP Ceiling Speaker. Made of electro-galvanized steel, the easy-to-install cylindrical design gives the finished look of the mount and speaker a clear and modern appearance.
Light USB Ultra Bright (10 LED) Portable and easy to use. Plugs into any USB port (eg: your laptop, a USB charger, USB port phone ...) Elegant design. The cylinder of the lamp have bright metal finish. It has flexible arm allowing multiple positions and angle adjustment. 10 White LED lights maximum illumination. LED lighting technology with long life (approximately 50,000 hours). Especially for deaf community and videophones: USB Accessory. Can be connected to Grandstream GXV3175 videophones and GXV3140, to signal an incoming call with a flashing LED 10 simultaneous. Features:
Dimensions:
Gateway Patton SN4120 - 2 BRI (TE) SN4120/2BIS4V/EUI
The SmartNode™ 4120 ISDN BRI PSTN Gateway converts four simultaneous phone or T.38 fax calls from SIP to ISDN BRI. It is a compact reliable standalone VoIP gateway for IP-based voice systems that delivers ISDN performance and quality. For system integrators looking to connect a VoIP or Unified Communication solution to public or private ISDN lines, the SmartNode 4120 provides unparalleled ISDN to IP feature preservation. Like any SmartNode the 4120 provides advanced ISDN functionality such as Explicit Call Transfer (ECT) support and Advice of Charge (AOC) over SIP. Patton CPEs are interoperable with most IP PBX and Unified Communication vendors such as Asterisk, 3CX, Elastix, Microsoft, IBM, Swyx, SNOM One and many others. Compared to PC-based solutions that combine IP PBX and a PCI BRI cards, the SmartNode has several advantages, including: installation without additional drivers or software, operation without ventilation or hard disk, scalable without limitation, and service integration without downtime. SmartNode gateways can be combined in a cluster for simple and flexible redundancy solutions not available with PCI BRI cards. System administrators don't have to learn ISDN as industry standard protocols like SIP do the job of connecting the PSTN to your VoIP system.
Alphatech Door Phone SLIM SDP 02
can be connected to PBX FXS or analog PSTN line.
Polycom KIRK power supply for 50xx, 60xx, and 70xx handsets.
Power supply for use with:
The SmartNode™ M-ATA Micro-Analog Telephone Adapter
The SmartNode Micro Analog Telephone Adapter provides connectivity for analog phones to a home, home office or corporate LAN. Connecting to any analog phone or PBX, the SmartNode product is ay cost effective solution for small offices and telecommuters to access Internet-based telephone services and corporate intranet systems across established LAN and Internet connections like xDSL and cable modems. The M-ATA provides one Ethernet (RJ-45) port and one FXS (RJ-11) analog phone port for quick and easy interconnection to the local LAN. LEDs show at-a-glance the status of the system, LAN, WAN, and phone ports. A full suite of IP features (DHCP, NAT/PAT) are available to maximize universal connectivity. VLAN tagging and prioritization enables voice traffic to be handled before data traffic insuring higher quality voice calls. Support for PPPoE tunneling simplifies extending corporate intranet services to telecommuters. The user friendly web interface offers two levels of configuration; level one covers basic subscriber specific parameters, level two offers advanced settings for the transport network. Configuration and firmware can be downloaded from a centralized TFTP or HTTP server. The M-ATA is SIP standard compliant. Analog phones attached to the SmartNode can use advanced calling features such as call forwarding, caller ID, 3-way calling, call holding, call retrieval and call transfer. Applications Patton's Micro Analog Telephone Adapter provides seamless access to Internet telephony and data services. The M-ATA connects to any broadband access provider via a cable or xDSL modem.
*1 Bluetooth is a trademark or a registered trademark of Bluetooth SIG Inc. *2 With LCD backlight off, Bluetooth is not in use.
*1 Including Battery
Cyberdata SIP Tiembre de oficina - Modelo 011216
The Cyberdata SIP-enabled SIP Office Ringer is a Power-over- Ethernet (PoE 802.3af) and Voice-over-IP (VoIP) notification device that easily connects into existing local area networks (LANs) with a single cable connection. The SIP Office Ringer provides a user uploadable audible ring indication when part of a ring group. The SIP Office Ringer also supports direct SIP voice paging and priority-based Multicast broadcasts.
Overview
The KIRK Single Charger Stand charges the Battery Pack within the handset. The charging is temperature controlled for optimal battery lifetime.
You can use all the keys and the speakerphone while the handset is placed in the charger. The Single Charger Stand also allows for rapid charging handset for the fast-paced work environments. The power supply is ordered separately to reflect different geographical regions.
The KIRK Single Charger for the KIRK 50-Handset Series is available in two varieties:
The Charger with USB port may be used for service (upgrading firmware etc.)
Mitel RFP 36 IP (outdoor)
The indoor base station RFP 36 IP enables the complete integration of DECT radio networks into the IP infrastructure and provides 8 simultaneous call connections. It is powered either via a separate power supply unit or via Power-over-Ethernet.
Radio Fixed Parts RFP are connected directly to the LAN like a VoIP device and use the benefits of established DECT technology for radio transmission. This ensures full compatibility with cordless DECT terminals, which are available as system telephones and standard GAP terminals.
The RFP 36 IP is ready to work in outdoor enviroments (IP 65, ULV94 V0). Uses built-in dipole antennas. Thus it can be used for the parking sites or delivery areas of a company as well as for a university campus or the outside area of hospitals.
DECT
* 60 DECT channels supported for maximum use of DECT capacity * 8 simultaneous voice channels per RFP, 4 additional channels for handover * GAP standard supported * Connection handover in line with the GAP standard * DSAA authentication between base and handset * Support of DECT encryption * DECT XQ enhancement improves interferences in Reflective environments.
New in version 3:
* It's not necessary a TFTP server to init the antenna software (OMM). The units have internal flash memory to keep the OMM program, what can be updated by TFPT or USB port.
* Gigabit network interface
* CAT-iq 1.0/ Mitel Hi-Q audio technology (G726, G722 wideband audio ) avalaible with CAT-iq capable handsets (like Mitel 650c)
For more detailed information visit the product page.
*1 With LCD backlight off.
Nosso departamento de pesquisa e desenvolvimento fornece um catálogo completo de serviços de engenharia que inclui consultoria profissional e suporte técnico para projetos de telefonia e comunicações.
Você pode contratar bônus de horas ou horas únicas (10 e 50 horas) que incluem suporte por telefone e por e-mail.
Si usted quiere ser un gurú de Asterisk, administrador o desarrollador, o simplemente aprender un poco más acerca de Asterisk, Avanzada 7 ofrece la formación para satisfacer sus necesidades. Unico training autorizado Digium® en España y Portugal.
La certificación Digium Asterisk-Certificado Profesional (dCAP) es una verificación de su conocimiento de Asterisk. La certificación cubre una determinada versión estable de Asterisk. Normalmente se celebra al final del curso de formación Asterisk.
¿En qué consiste el Examen?
La prueba consiste en un examen teórico de 150 preguntas por escrito sobre Asterisk y la tecnología relacionada, y un examen práctico de laboratorio en el que se le pide que configure una PBX de acuerdo con una especificación determinada... (no se conoce hasta el momento del exámen). Nota: Realizar el curso avanzado de Asterisk no garantiza pasar el examen dCAP. Para aprobar el examen dCAP, se recomienda que usted se haya leído el libro de la editorial O'Reilly "Asterisk: El futuro de la telefonía, 2 ª edición", y tenga conocimiento efectivo de creación y mantenimiento de un servidor Asterisk, se haya familiarizado con los archivos de configuración de Asterisk de ejemplo (samples), y haya realizado el curso Astersik Advanced de Digium®. Información sobre código abierto Asterisk se puede encontrar en Asterisk.org. Para inscribirse para la certificación dCAP, debe abonar los derechos de examen.
The B410P enables Basic Rate ISDN S/T connectivity for Asterisk users outside of North America and connects eight digital voice channels (four ISDN lines for a total of eight B-channels) per card to an Asterisk server.
The B410P is a half-length, full-height universal 3.3V and 5.0V 32-bit PCI 2.2 card supporting four BRI S/T interfaces. Each of the four ports of the B410P can be independently configured for TE or NT mode, with an optional PWR400M module for supplying power to ISDN telephones. The B410P features on-board hardware echo cancellation performing 64ms or 512 taps per channel for each of the eight voice channels.
Grandstream Training Course Price Includes a Grandstream Demo Kit Worth $1000! (Voice, Video, Data & Mobility)
Course Description:
Avanzada7 Staff is Grandstream Certified with decades of experience in VoIP and IP Surveillance. Get direct access to Grandstream’s top technical personnel as they walk you through step-by-step programming of their new UCM6100 series PBX units, IP phones and IP cameras, as well as introducing you to their entire VoIP and IP Surveillance product lines. Course materials are combined with lab working on programming the equipment so you leave with confidence and a good working knowledge of the Grandstream´s products. Every student will have the opportunity to use Grandstream’s UCM PBX, VoIP handsets and IP Cameras. Upon completion of the training and lab exercises there will be a official certification test (price included). Course Outline
• Grandstream Overview • Introduction to VoIP/Benefits for VoIP • Grandstream VoIP Products • Introduction to IP PBX • Introduction to UCM6100 Series • Technical Training Using UCM6102 & GXP1405 • ** Student should bring laptop and power supply for lab work.
Program:
Demo Kit Includes:
• (1) UCM6102 IP PBX Appliance • (1) GXP2200 Enterprise Multimedia Phone for Android™ • (1) GXP1405 Small-Medium Business HD IP Phone • (1) PoE Switch (4-port) and Power Supply • (1) Durable Grandstream-Branded Rolling Carrying Case • All Interconnecting Cables and Power Supplies • STUDENT SHOULD BRING LAPTOP AND POWER SUPPLY FOR LAB WORK • VoIP & Surveillance Product Brochures Download Demo Kit Brochure
Grandstream Switchboard UCM6202
The Grandstream UCM6202 provides a solution to the communication needs of the company. It combines voice, video, data and mobility functions in one easy-to-use solution. In addition, it can be remotely managed and offers technologies such as voice, video call, videoconference, video surveillance, among others, that ensure an optimization of the communications in the line of this series UCM6200.
Fax For Asterisk
Digium's Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry's leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium's Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.
FREE 1st License:
Additionally, each open source or commercial Asterisk system is eligible to receive from Digium, a single channel of Fax For Asterisk, called Free Fax For Asterisk, for no cost. Free Fax For Asterisk is provided under license as-is, without technical support, and is available to all Asterisk users as a free, zero cost purchase from the Digium webstore. Only one channel of Free Fax For Asterisk may be used with an installation of Asterisk. If you require multiple channels of Fax capability or if you require Digium's technical support, you may purchase channels of Fax For Asterisk.
Los profesionales de oficina necesitan un sonido nítido, así como comodidad y fiabilidad durante todo el día. SupraPlus de Plantronics ofrece todo esto y mucho más, lo que convierte a este auricular en uno de los más populares en todo el mundo.
Working with the latest in wideband VoIP technology, the Plantronics SupraPlus Wideband headset delivers the highest level of audio performance even in noisy environments. The SupraPlus Wideband helps overcome the challenges posed by traditional technology, delivering heightened speech clarity and life-like fidelity. Users will experience greater satisfaction through enhanced intelligibility and reduced miscommunication. Designed for over-the-head comfort, the SupraPlus Wideband lets users choose between experiencing total-focus sound reduction (Binaural design) or conversing easily with coworkers without removing the headset (Monaural design).
The headsets have been updated with a new look, improved ergonomics, and simplified usability.
The snom HS-MM3 headset is made specifically for use with the snom 300 phone whereas the HS-MM2 headset is used for the snom 320, snom 360, and snom 370 phones.
Headsets allow their users a much wider range of movement than possible with a telephone receiver. This investment is worthwhile for anyone who spends a an office, in a home office, or in a call center - a headset offers many advantages.
The new snom headsets have one ear pad which can be worn on either ear. They are very sturdy and offer good wearing comfort.
The microphone is noise-cancelling and suppresses background noise so that you do not need to compromise on quality.
Disasters in a communication network are often very difficult to predict and there is usually a very small advance notice when a communication line goes down. beroFos provides an effective way for dealing with such unexpected events by re-routing the lines to a back-up line when undesirable changes are detected. Therefore beroNet provides bero*fos, a solution device for PBX Clustering and failover scenarios that requires a physical reconnection of analoge, BRI or PRI lines. In addition to this properties, berofos has two individually switchable powerports on the rear panel, for remote poewer on /power off or reboot the respective connected devices. The berofos can be used in two scenarios, the failover scenario or the bypass scenario.
Failover Scenario The failover scenario grants the smooth and reliable operation of two PBXs side by side, where the second one will be switched to active only after the failure of the first one in order to reduce downtime to a minimum (see figure). This scenario is not only interesting in failure situations, it could also be used in cases of maintenance, repair or upgrade of your telecommunications equipment.
Bypass Scenario The bypass scenario is interesting for customers who want to keep their old telephone system and add new functionlalities by adding a new PBX in front of the old one (eg by an Asterisk system). With the bypass szenario you have the opportunity to add a new PBX between the PSTN Lines and the outgoing Lines of the old PBX system. In the case of a failover, the PSTN Lines will transparently reconnected to the existing old PBX (see figure).
Backplane for 2 Remoras Sangoma A500.
Adding an additional Remora card to your existing A500 allows you to add an additional 6 BRI ports, by installing one to three BRI modules. Each A500 can support up to 4 cards, including the base PCI card, for a total of 12 available ports.
Backplane Sangoma 3 conectores A500
Cada A500 admite hasta cuatro tarjetas rémora, sobre una única tarjeta PCI de base, llegando a un total de 12 slots disponibles (=24 Puertos RDSI).
Disponibles backplane de 2, 3 y 4 tarjetas.
Rather than the 64kbit/s required for a standard, uncompressed G.711 PCM audio data stream, the G.729 codec compresses the payload to 8kbit/s. Bandwidth calculations for a VoIP call should consider signaling and packet overhead as well, which varies according to network topology. In a typical Ethernet environment and utilizing the SIP or IAX signaling protocols, a G.711 call will consume about 87.2kbit/s while a typical G.729 compressed call will consume about 31.2kbit/s.
A practical example is the number of calls that may be carried across a standard 1.5 megabit/s T1 link. When using uncompressed G.711 audio, one can expect 18 concurrent calls across a T1. And, when using G.729 compression and Digium's IAX2 Trunking, instead of SIP, signaling protocol, one can expect about 140 concurrent calls across the same link.
Digium's implementation of the G.729 Codec in software allows Asterisk to transcode (compress and decompress) audio to and from formats other than G.729. Many business-class IP telephones and VoIP gateways include support for G.729. With the Digium G.729 Codec for Asterisk, those devices can now exchange calls with Asterisk directly.
Without the capability to transcode G.729, Asterisk can only pass-through G.729 data between endpoints. This means that basic station to station calling can be made to work, but the advanced PBX features of Asterisk such as Call Conferences, DTMF digit collection, Call Recording and more will not work without Digium's licensed G.729 Codec.
Multiple versions of G.729 are defined according to industry standards. Asterisk, and Digium's G.729 implementation support G.729 Annex A, or G.729a. Aster isk and Digium's G.729 implementation do not support G.729 Annex B, or G.729b.Digium's software G.729 Codec utilizes the power of the host system's CPU to perform its transformations. Therefore, the transcoding capacity, in terms of simultaneous channels/transcodes, is determined by the performance of the host server. Digium's internal testing indicates that 60 concurrent G.729 calls/transcodes require a system equivalent to a dual Intel Xeon at 1.8GHz. Further testing indicates that 80 concurrent G.729 calls/transcodes require something equivalent to a dual Intel Xeon at 2.8GHz.
Digium's G.729 Codec for Asterisk is licensed on a per-channel basis. A channel is defined as a single connection from an endpoint to an Asterisk applicatio n, or a bi-directional call between two endpoints attached to Asterisk. Customers may use the licensed G.729 Codec in conjunction with Asterisk and any combination of Digi um telephony interface boards.
The G.729 Codec is provided with support from Digium's Technical Support organization for Linux x86 and x86_64 environments. Digium also provides builds for other platforms, but without support.
Backplane for 4 Remoras Sangoma A500.
A200BP2: 2 Connector Backplane (up to 4 Ports)
Backplane for 2 Remoras Sangoma A200.
Adding additional Remoras card to your existing A200 allows you add up 24 analog ports (6 rémora, with 2 slots/each up 12 slots = 24 FXS/FXO ports. Each A200 can support up to 6 cards, sharing base PCI card, for a total of 24 available ports.
Backplane with 2, 3, 4, 5 & 6 connector availables.
A200BP3: 3 Connector Backplane
Backplane for 3 Remoras Sangoma A200.
O comutador FS108P fornece energia e dados a partir de um único ponto, usando PoE em um único cabo Cat-5. As oito portas Ethernet podem ser usadas para qualquer link de 10/100 Mbps e quatro dessas portas podem fornecer energia padrão da indústria IEEE 802.3af.
O avançado algoritmo de detecção automática fornece energia apenas para os dispositivos finais 802.3af, portanto, você não deve se preocupar em danificar o equipmento proprietário PoE ou não PoE.
Além disso, interrompa a energia quando os dispositivos PoE estiverem desconectados. Fácil e confiável, o FS108P a um preço acessível oferece PoE para qualquer pequena empresa.
Easy to set up and use, the Cisco 300 Series Switches provide the ideal combination of affordability and capabilities for small businesses, and help create a more efficient, better-connected workforce.
The Cisco 300 Series, part of the Cisco Small Business line of network solutions, is a portfolio of managed switches that provides a powerful, reliable foundation for small business networks.
Powerful Features and Performance
All Cisco 300 Series Switches support the advanced security management capabilities and network features needed to support business-class data, voice, security, and wireless services. These switches support long-term investments and advanced features, such as quality of service, Layer 3 static routing, and IPv6.
Cisco 300 Series Switches offer:
A200BP4: 4 Connector Backplane
Backplane for 4 Remoras Sangoma A200.
Digium D40 Phone: The Only Phones Built Specifically For Asterisk
Digium's family of IP Phones are the first on the market built specifically for use with Asterisk and Asterisk-based systems. All models include HD audio and plug-and-play deployment at a price that fits any budget. With multiple line appearances, context-aware soft keys, and advanced applications that integrate directly with Asterisk features, the Digium phones offer a better user experience than any other phone on the market.
Digium D40 Features:
Digium IP Phones
Asterisk Phone Features Smart Software
Access to information is the key to productivity in today's business environment. The integrated applications that come standard with all Digium phones put critical information at your fingertips. With voicemail, call log, contacts, phone status, user presence, and parking, the Digium phones provide simple, intuitive access to a wealth of information, saving valuable time.
Simplified Provisioning
Standards-based IP phones have a reputation for being difficult to install and configure. Most systems require changes to network configurations or additional components to facilitate deployment. Digium phones support plug-and-play provisioning. Simply plug in the phone and it will automatically discover Asterisk systems on the network. Select the user you want to assign to the phone and the proper configuration is instantly loaded. For larger deployments you can pre-assign phones by tying a MAC address to an Asterisk user. It's that simple.
Custom Applications
Most desktop phones come with a fixed feature set that is determined exclusively by the manufacturer. Digium phones are different. All models include the Digium app engine, an innovative application framework that makes it remarkably simple to add new functionality or tailor the standard applications to your needs. All of the productivity applications that ship with the phones are written using the app engine SDK. Documentation for the app engine is coming soon, opening up the phones to third party development.
The H8 Series are versatile devices used for connecting to the PSTN. This is accomplished with analog lines connected to trunk (FXO) interfaces, analog handsets connected to station (FXS) interfaces, and digital lines connected to ISDN BRI interfaces. The H8 Series supports all three types of connectivity on a single card through the use of Digium's specialized telephony modules for FXO, FXS, and BRI. Used in conjunction with Asterisk®, the H8-series cards can deliver a wide range of voice applications including IP PBX, VoIP Gateww ay, IVR platform, Conference Bridge, Voicemail, and more.
Like Digium's analog and single-port PRI cards, the H8 cards utilize Digium's VoiceBus™ technology. VoiceBus technology allows the H8 cards to use an industry standard bus-mastering PCI interface, as found in millions of PCs worldwide, to maximize system compatibility and eliminate system conflicts. Additionally, the H8 cards can improve voice quality in troublesome echo situations with its support for Digium's G.168-compliant 128ms cancelling VPMADT032.
The H8 Series are half-length, full-height cards supporting Digium's existing single-port FXS (S110M) and FXO (X100M) modules as well as Digium's existing quad-port FXS (S400M) and FXO (X400M) modules. New with the release of the H8 cards is Digium's B400M four-port EuroISDN S/T module. The B400M sets a new standard for BRI connectivity in the Asterisk market with its support for software-selectable mode (NT or TE) and line termination. The B400M requires no jumpers for operation, regardless of mode or termination.
The H8 is available in universal 3.3/5.0V PCI format as HA8 and in PCI-Express x1 format as HB8.
A200BP5: 5 Connector Backplane
Backplane for 5 Remoras Sangoma A200.
The FX0 module allows the TDM400P card to termiante analog telephone lines (POTS). Because of the modular design, a user can activate additional ports at any time with more FXS or FX0 daughter cards.
The FX0 module passes all the call features any standard analog telelphone line will support.
Assuming a basic knowledge of PHP, XML, JavaScript and MySQL, this book will help you understand how the heart of AJAX beats and how the constituent technologies work together. After teaching the foundations, the book will walk you through numerous real-world case studies covering tasks you'll be likely to need for your own applications:
- Server-enabled form-validation page
- Online chat collaboration tool
- Customized type-ahead text entry solution
- Real-time charting using SVG
- Database-enabled, editable and customizable data grid
- RSS aggregator application
- A server-managed sortable list with drag&drop support using the script.aculo.us JavaScript toolkit
The appendices guide you through installing your working environment, using powerful tools that enable debugging, improving, and profiling your code, working with XSLT and XPath.
From the Author, Cristian Darie
"AJAX and PHP: Building Responsive Web Applications is mainly a book for beginners, but when designing its contents we tried to find the ideal blend of topics that would help both novice and experienced web developers make a big step forward. One customer was very kind to let us know, through a review, that we succeeded:
"The theory behind all the technologies used is very clearly explained, without boring you with details about obvious things. Right from the first chapter you start learning by examples. The examples can be easily adapted to many web projects and they cover stuff that is both useful and fun."
TOPEX MobiLink IP is a GSM/UMTS small capacity gateway with VoIP interfaces. Its main functionality is to interconnect IP PBX or hibrid PBX with mobile networks. With TOPEX MobiLink IP you make significant savings on calls from IP to cellular networks and backwards.
* GSM/UMTS gateway with VoIP interfaces
* Interconnection with IP-PBXs based on SIP
* Cost reduction and savings through LCR
* Advanced call-routing engine
A200BP6: 4 Connector Backplane
Backplane for 6 Remoras Sangoma A200.
Using Digium's Asterisk® software and standard PC hardware, one can create a telephony environment that includes all of the sophisticated features of a high-end business telephone system.
Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus Technology, the AEX410 eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additiona l AEX410 or other Digium analog interface cards.
The FXS module allows the TDM400P card to terminate analog telephones. Because of the modular design, a user can activate additional ports at any time with more FXS or FX0 daughter cards.
The FXS module passes all the call features any standard analog telelphone will support.
The B400M sets a new standard for BRI connectivity in the Asterisk market with its support for software-selectable mode (NT or TE) and line termination. The B400M requires no jumpers for operation, regardless of mode or termination. Digium B400M BRI Module 4 BRI Ports (8 Voice Channels) • EuroISDN S/T • NT - PTP • TE - PTP / TPMP Up to 2 B400M per H8 Card • 8 BRI Ports on one card (16 Voice Channels) Software Configurable; No Jumpers! • SET TE or NT mode in Software • SET Termination in software EAN Code 797734475845 Price € 288.70 Add Your Tags: Add Tags Use spaces to separate tags. Use single quotes (') for phrases.
Plantronics Audio 400 DSP
O Plantronics 400 DSP possui um microfone com cancelamento de ruido e é ideal para reconhecimento de voz, telefonia pela Internet e videoconferência. Além disso, o processamento de signal digital (DSP) oferece até 35% de redução no ruído de fundo e mantém a transmissão de voz clara e nítida.
Sangoma Fax Synch Cable
Sangoma Fax Synch Cable hooks up to a two pin spot on the card. The Sangoma cable will sync up the A200 and A400 cards with a digital card so that faxing is easier and has about a 99 percent success rate.
The FX0 module allows the TDM800P and TDM2400P card to termiante analog telephone lines (POTS). Because of the modular design, a user can activate additional ports at any time with more FXS or FX0 daughter cards.
La Serie Jabra GN2000 le permite escuchar cada detalle, a la primera. Para telefonía IP de ordenador, los Jabra GN2000 IP y Jabra GN2000 USB soportan toda la amplitud de banda de la red IP para aplicaciones como reconocimiento de voz, llamada de voz sobre IP y de formación por ordenador. Para las aplicaciones RTPC, la Serie Jabra GN2000 son auriculares perfectos para entornos concurridos y ruidosos, mediante el uso de toda la frecuencia de la red telefónica, asegurando una calidad de sonido excepcional en cada llamada.
Desde su sólida composición de goma y plástico hasta los pivotes flexibles y los auriculares verdaderamente resistentes, cada detalle de la Serie Jabra GN2000 ha sido diseñado para su durabilidad, haciéndolo ideal incluso para los centros de llamadas más exigentes.
The FXS module allows the TDM800P and TDM2400P card to terminate analog telephones. Because of the modular design, a user can activate additional ports at any time with more FXS or FX0 daughter cards.
Mitel SIP Power Supply for telephones Astra (except model 6730i)
Digium announces the release of a new product that will revolutionize T1/E1 communications around the world. Our new card, TE405P, can be a quad E1 or T1 selectable per card or per port. Now you can do both signaling formats in a single card. Also, with our new bus mastering design, there are significant performance improvements over older designs. We strive to make our hardware backward compatible with existing Asterisk applications, so this card is a drop-in replacement for legacy quad cards. The TE405P has been approved for use in Australia, Europe, and the United States.
The TE405P supports a 5v PCI slot only - typically available on newer motherboards and in 64-bit PCI bus architectures. The TE410P supports a 3.3v PCI slot only.
Now, Digium is pleased to make available the VPMOCT032 echo cancellation module. This module, which offers 128ms (1024 taps) of echo cancellation across all of its channels and provides the same toll-quality G.168 compliant echo cancellation found in Digium's HPEC software-based commercial echo canceller. The VPMOCT032 is immune from any system CPU spikes that might otherwise affect a software-based solution.
Power Supply for Grandstream telephones.
VPMOCT128M Módulo Cancelación de Eco Hardware basado en DSP, proporciona el algoritmo G.168 que ha marcado un punto de referencia para la cancelación de eco y realiza 128ms (1024 taps) de cancelación de eco en TODOS los 120 canales del modo primarios E.
• Hasta 128 canales. • 128ms (1024 taps) por canal.
Este módulo trabaja con las siguientes tarjetas: • TE410P (bundled como TE412P) • TE405P (bundled como TE407P) • TE420 (bundled como TE420B)
The TC400B is a bundle of the half-length, low-profile PCI-2.2 compliant TC400P base card and the TC400M voice processing module. The TC400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio as would otherwise be processed by Asterisk in software. Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TC400B not only relieves the CPU of this duty, freeing it up to handle other tasks or to complete additional call processing; but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not previously possible. The TC400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit) into u-law or a-law; or, compresses u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TC400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TC400B does not require additional licensing fees for the use of these codecs nor does it require the registration process associated with Digium's software-based G.729a codec licensing.
The Ceiling Mount Bracket enables the various CyberData VoIP Ceiling Speakers for 24-inch wide ceiling tile mounting.
The TCE400B is a bundle of the half-length, low-profile PCI-Express x1 TCE400P base card and the TC400M voice processing module. The TCE400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio as would otherwise be processed by Asterisk in software.
Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TCE400B not only relieves the CPU of this duty, freeing it up to handle other tasks or complete additional call processing, but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not otherwise possible.
The TCE400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit/6.0kbit) into G.711 u-law or a-law; or compresses G.711 u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TCE400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TCE400B does not require additional licensing fees for the use of these codecs nor does it require the registration process association with Digium's software-based G.729a codec licensing.
The Konftel 300IP is a flexible SIP-based conference phone that is ideal for companies using IP telephony. It is equipped with Konftel’s patented audio technology OmniSound® 2.0 with wideband to ensure a clear and natural sound. The stylishly designed Konftel 300IP is packed with intelligent functions to make teleconferences more efficient even in larger conferences.
With the Konftel 300IP your company will have a top-quality conference phone that combines all the benefits of IP telephony with innovative new functions.
Lifetime Access is a membership that allows the student to remain enrolled in the course permanently and have access to updated material.
Included:
Not included:
The student will have access to the content available on the academic platform and to the exclusive mailing list to review actual questions and answers. The mailing list is moderated by official Elastix instructors.
With this membership, the student makes a long-term investment in training.
Digium TE820F (8PRI) PCI express
Digium octal span digital cards are high-performance, cost effective digital telephony interface devices with the power to seamlessly interconnects traditional telephony systems with Voice over IP technologies.
Digium’s TE820 Octal-Span digital card is built exclusively for high-density Asterisk® applications that require high-performance, cost effective digital connectivity. The TE820 includes eight independently software-selectable digital telephony interfaces, supporting up to 192 channels (in T1/J1 mode) or 240 channels (in E1 mode) — the highest single-card port density available for use with Asterisk. The TE820 seamlessly interconnects traditional telephony systems with Voice over IP (VoIP) technologies at the lowest per-port price on the market today.
The TE820 card supports industry standard telephony protocols, including multiple variants of Primary Rate ISDN. Each span can be configured as either CPE or network for optimal flexibility. The optional VPMOCT256 hardware echo cancellation module, based on the industry-leading Octasic chipset, offloads the task of echo cancellation from the CPU, increasing overall system performance and call quality.
The TE820 card has been designed to be fully compatible with existing software applications and is fully integrated with Digium’s Asterisk software. The open source drivers for Digium cards support an application programming interface (API) for custom application development. The combination of Digium hardware and Asterisk software provides a cost effective platform for building numerous communications solutions, from PBX system and VoIP gateways to call centers and complete unified communications suites. Digium solutions are paving the way for a new generation of worldwide communications.
The Sangoma A500 S/T BRI interface card delivers superior audio quality and scalability. Expand from two to twenty-four ports of BRI
* without hardware echo cancellation.
A single PCI or PCI Express slot hosts the connection for up to 24 ports and ensures common synchronous clocking for all channels with no signaling issues. The card is 100% software configurable.
• From 2 to 24 ports are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers — simply invert the colour-coded module
• Support for Asterisk®, FreeSwitch™, and Yate as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications
• Single synchronous PCI and PCI Express interface for all 24 BRI interfaces
• Six ports per RemoraTM card
• Dimensions: 2U Form factor: 187 mm x 55 mm for use in restricted chassis; includes high quality, tested 2 m 8-pin RJ45 port splitter cables and short 2U compatible mounting clips for installation in 2U rackmount servers
• 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention
• Autosense compatibility with 5 V and 3.3 V PCI busses
Power supply for Linksys terminals.
FONESTAR FE-1310 Altavoz exponencial de baja impedancia, 30 W máximo
The GXW400x FXS Series is an ideal solution for businesses looking to connect one or more lines of a traditional PBX to a VOIP phone system or provider. The GXW400x features 4 or 8-port FXS interfaces for analog telephones, dual 10M/100M network ports with integrated router, PSTN life line in case of power failure, and an RS232 serial port for administration.
The Konftel 300 is packed with smart features and has been developed for flexible application. You can record your calls onto an SD memory card and the line selector mode allows you to switch between and combine three connectivity technologies, analogue, mobile phone and USB. The conference guide is a useful feature if you make regular calls to the same group as it enables you to set up multiparty calls and save call groups.
he Konftel 300 is also ideal in larger contexts as you have the option of adding expansion microphones, a wireless headset and an existing PA system too. It goes without saying that it delivers ultimate sound quality, based on OmniSound ® 2.0 , the new generation of Konftel’s crystal clear audio technology.
The Sangoma A500 S/T BRI interface card delivers superior audio quality and scalability. Expand from two to twenty-four ports of BRI with optional telco-grade hardware echo cancellation.
Adding an additional Remora card to your existing A500 allows you to add an additional 6 BRI ports, by installing one to three BRI modules. Each A500 can support up to 4 cards, including the base PCI card, for a total of 12 available ports. This card connects to your existing A500 through the use of an external bus connector, which attaches to the rear of the card. It is therefore recommended that you install these cards next to each other in the chassis.
NOTE: When ordering this card to add to your existing A500, please ensure you select the right bus connector for the total number of cards, including the base PCI card, you expect to install in your system. For example, if you plan on having 12 ports in total, you will need the "2 Card Bus Connector", or for 24 ports, you will need the "4 Card Bus Connector".
In addition to the corded snom headsets, wireless headsets are very popular in professional business environments like, for example, call centers.
The snom Headset Adapter, for the control of wireless headsets is the bridge between professional VoIP telephony and professional wireless headsets.
The snom Headset Adapter has an EHS interface conforming to vendor specific as well as the DHSG standard which enables the electronic reception of calls on the headset itself.
When the phone receives an incoming call the original ringing tone is signalled in the headset and the call can be answered and terminated on the headset.The snom Headset Adapter has been constructed specifically for the snom 320, 360, 370 and 820 phones.
Whether it's business usage in large or small enterprises or in call centers - the newly won freedom of movement will pay off immediately.
Main features:
- Complete freedom of movement
- Signaling of ringtone
- Call acceptance on headset
- Call termination on headset
- Perfect integration of firmware
- vendor specific EHS protocols
- DHSG Standard
- No additional power supply required
- Easy to connect
Expansion microphones for Konftel 300 series
Expansion microphones (1 pair). Increase the range from 30 to 70 m2 (up to 750 sq ft). Connects to the Konftel 300IP with enclosed cables, 1.5 or 2.5 metres.
The GXW FXO IP Analog Gateway 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 multiple locations with up to 8 PSTN lines, to an IP PBX system, or with an existing traditional phone system.
La licencia permitirá aumentar las funcionalidades de su equipo Quadro con la capacidad de almacenar las llamadas que se cursen en el sistema.
Estas llamadas podrán ser grabadas automáticamente o a voluntad del usuario; almacenadas temporalmente en el equipo o enviadas vía FTP a otra hubicación.
Expansion microphones for Konftel 60W/200W
Expansion microphones (1 pair). Increases the range up to 70 square meters (750 sq ft). Connection cables in two lengths, 1.5 meters (4 ft) and 2.5 meters (8 ft) included. Color: Deep sky-blue.
Call Recording is a powerful feature allowing the system to record all calls made from and to the IP extensions of the PBX. This allows a user to record selected calls both automatically and by special request from the Web GUI or directly from the phone. The recordings could be stored either on the IP PBX (and be reviewed on the Quadro) or be uploaded to an external file storage for further processing. Call Recording is a purchasable feature priced per recording port and sold in groups of ports available on the QuadroM IP PBX products, including 8L, 12Li, 26x, 26xi and 32x.
QuadroM32x: Four 8 port licenses can be purchased for a total of 32 recording ports.
Recording can be set to record all calls or restricted based on called/caller party number or based on the digits dialed.
Record calls automatically or after pressing the Record button on the handset. Recording status displayed on Aastra & snom phones, as well as displayed on Quadro GUI.
Recorded files: .wav files using G.711
Saved, viewed or played back locally on the Quadro GUI
Saved, viewed or played back on an ftp server Optionally prompt for password before playback
Using Digium's Asterisk software and standard PC hardware, one can create a telephony environment that includes all of the sophisticated features of a high-end business telephone system.
Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus™ technology, the AEX2400 eliminates the requirement for separate channel bank and T1 interface cards, with industry-leading performance and price. The quad trunk and quad station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional AEX2400 interface cards.
Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a telephony environment that includes all the sophisticated features of a high-end business telephone system.
Using an industry-standard bursting, bus-mastering PCI interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus™ technology, the TDM2400P eliminates the requirement for separate channel bank and T1 interface cards, at an industry-leading price. The quad-FXO and quad-FXS modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both FXO and FXS interfaces. Scaling of this solution is accomplished by adding additional TDM2400P cards.
The CyberData Wall Mount Adaptor enables the various CyberData Ceiling Speakers for wall mounting.
The Barge-in Feature Pack is comprised of three independent features designed to help supervisors or managers of a call center or in an office environment monitor and coach their employees. These features are tools supervisors can use to participate in conversations between employees/agents and customers. In addition, supervisors can monitor their employees’ performance or customers’ behavior. The supervisor can coach an employee while he/she is en-gaged with a customer on a phone call without the customer knowing, or the supervisor can participate in a three-way call and assist both at the same time or just monitor the call of an employee and customer
Silent Monitoring Feature This feature allows the supervisor to listen to a call between an internal extension (employee) and an external call (the customer).
Agent Whisper Feature With the Agent Whisper feature, the supervisor can dial *92+extension number and listen to the conversation but only speak and be heard by the internal extension (employee) not the external call (the customer).
Barge In A supervisor will be able to join an established call between the employee and customer and have a three-way call.
Administrative Assistant To further enhance the abilities of an Administrative Assistant, the Barge-In Feature Pack adds some very interesting communication options. For example, if the executive would like the assistant to take notes during a call, but ensure they cannot be heard, silent monitoring would be used. Whisper could also be used if the assistant needed to interrupt the executive if there was a more urgent situation.
Panel 16 port dual coaxial (BNC/1.6-5.6) to RJ45 twisted pair, specially designed for telecommunications operators
Balun stands for: Balanced-Unbalanced and the term is used to identify adapters which connect twisted pair balanced 120 ohm lines, such us ethernet cables, to coaxial 75 ohm unbalanced lines (as many operators provide ISDN primary rate lines).
Siemens coaxial connector is the most common in the market, so you should only check connector sex used by the operator to choose the opposite one in the balun.
Standard hunt groups and simultaneous call distribution are not sufficient for the demands of true call center environments. Advanced methods of call distribution are required, such as Skills Based Routing or Least Active Agent. Grouping of the agents is also a key advantage, allowing call centers to logically group resources together to clearly define responsibilities and expertise of the agents. Automatic Call Distribution (ACD) is a purchasable feature only available on the QuadroM IP PBX products, including 8L, 12Li, 26x, 26xi and 32x.
The SIP-enabled IP Intercom with Keypad is a Power over Ethernet (PoE 802.3af) and VoIP two-way communication and door entry device. Equipped with a 12-key keypad, the Intercom also has programmable speed dial and secure doorlock access through the keypad for up to 20 pre-programmed users.
The 12-key keypad Intercom also has programmable speed dial and works well for any private or public sector entity requiring mass notification such as medical offices and hospitals, railway stations and ports, manufacturing warehouses and more.
The VoIP Intercom with Keypad can be configured as a Night Ringer with two SIP extensions. One extension can be assigned to a page group for auto answer paging. The second extension can be assigned to a “First-to-Answer” Night Ring group with IP phones. An audio ring file is activated when the second SIP extension of the Intercom is dialed. If any of the IP phones in the ring group is answered (or if the caller hangs up), the Intercom stops ringing.
Standard night ringer function can be utilized.
Features: ? SIP or multicast compatible ? Night Ringer function ? Dual speed 10/100 Mbps ? POE 802.3af ? Network firmware upgradeable ? Network volume adjustable ? Network microphone sensitivity adjust ? Power-over-ethernet enabled ? 12-key kepad ? Dry contact relay
Note: The VoIP Intercom with Keypad is available in either the Wall-Mounted version (011113) or the Flush-Mounted version (011123).
Conector adapter 15 cm. Male-Female for BALUN
The CyberData Clock Kit enables the CyberData V2 Ceiling Speaker for wall mounting.
The Clock Kit enables the CyberData V2 VoIP Ceiling Speaker to be upgraded to a highly visible public address device with a time display. The large display characters are easily read and adjust for ambient light conditions. Time sync is performed by NTP and a built-in real-time clock. Power is supplied by the CyberData V2 Speaker from a single PoE connection.
The VoIP Clock Kit consists of the VoIP V2 Ceiling Speaker and either the Clock Kit Wall Mount Adapter or the Clock Kit Flush Mount Adapter.
The Mediatrix® 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec’s simultaneously on each analog voice ports, saving valuable bandwidth. The Mediatrix 4100 Series also offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling and media transmission aspects.
In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
The 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
Main Advantages: Interoperable with equipment from leading industry vendors, major savings in long-distance and PSTN access fees, re-use of existing infrastructure such as analog phones and faxes, simplified wiring = savings in ADDS/MOVES/CHANGES, compliancy to current legacy standards increases compatibility with actual PBX or key systems, access to essential PSTN emergency calls capabilities even during power outages.
The 4102 is available in two models: the 4102 and 4102S.
Security-Ready VoIP Solution
The Mediatrix 4102S supports SIP over TLS, SRTP and SNMP, providing a totally “security-ready” product, which interfaces seamlessly with the full Mediatrix portfolio of products in secure networks.
4100 Series are available in 2, 4, 8, 16 and 24 FXS ports and soportted TR-069 operation
ClearOne Standard Definition Room Video Conferencing
The Right Mix of Quality and Features
Incorporating innovative technologies developed by pioneers in the worlds of audio and video conferencing, the COLLABORATE Room SD, like other systems from the COLLABORATE Room series, offers ease of operation and the right mix of quality and features.
Top-rated echo/noise cancellation for crisp, clean audio every time
Daisy-chainable HDConference® microphone array for up to 6 microphones, guaranteeing 360-degree coverage to ensure all participants are heard
Recording – on internal & external HDD for playback and archiving of important meetings
Streaming – unlimited numbers of people can view a session (great for webinars and remote training)
Soft-codec videoconferencing – enabling flexibility, easy upgrades, future-proof scalability, low TCO (total cost of ownership), and openness to the world of unified communications
Unbeatable Return on Investment (ROI) COLLABORATE Room SD is a full-featured system at an unbeatable price. With the lowest cost in its class, its high-quality audio and video lets you “be there” without the need to pay for travel. Wherever you are, the multi-purpose design and ease-of-use of the system contribute to workplace efficiency.
System Includes:
COLLABORATE Room SD Codec
Upgradable up to 9-way MCU
Recording and Streaming
SD PTZ Camera
HD Microphone Array (daisy-chainable to attach additional mic array)
Wireless Remote Control
Power supply and cables
Ampliación de garantía a 3 años para AA60
The PA interface box connects the conference phone with an existing PA system. To match several types of situations and equipment, there are some settings available in the Konftel 300/300IP menu.
Parts included:
-Interface box
-Connection cable, 2.5 metres
Output specification:
Connector RCA
Impedance 600Ω
Output level 200mV RMS
Input specification:
Impedance 5kΩ
Input level 300mV RMS
Konftel 250 is an excellent choice for holding telephone conferences with great sound quality. It’s the obvious choice when you have an analogue line – high performance, smart features and expandable too. The Konftel 250 is equipped with Konftel’s patented OmniSound® 2.0 audio technology, for crystal-clear sound. Save your contacts in the phone book and use the conference guide to easily set up multi-party calls or pre-programmed group calls. Konftel 250 also has a built-in recording function that enables you to record your calls on an SD memory card. With its modern Scandinavian design, Konftel 250 will be a welcome addition to any conference room. It’s also ideal in larger settings with the addition of optional expansion microphones. Hold productive telephone conference meetings that not only save time but the environment as well by cutting travel expenses.
Alphatech IPDP01C antivandal (SIP 1 botón + cámara )
Este Video-Portero Automático o Video-Interfono VoIP logrará satisfacer sus necesidades de comunicación con las personas en la puerta principal del edificio o la entrada de la empresa, o la puerta de casa familiar. La universalidad se encuentra en la posibilidad de conectar este dispositivo a una red Ethernet, una centralita VoIP o registrada con su servidor SIP a través de Internet.
Terminación en Zama para uso en situaciones donde se requiere una protección adicional frente vandalismo.
- Solución compacta con botón de llamada, altavoz/micro y cámara de color VGA con autofocus.
- Iluminación automática con LED blancos mediante sensor de luz (ajustable).
- Hasta 64 x 2 grupos memorias para extensiones programadas (hasta 16 dígitos), usados de dos formas:
- Audio G711, GSM, G726-23. Video H263 en CIF hasta 5 fps. DTMF RFC2833 o SIP INFO.
IP Door Phone Intercom (IPDP00)
VoIP Door phone is a complete door entry solution based on IP communication.
The connection via SIP P2P protocol to different IP systems or LAN makes it very attractive for all company sizes. The support of video as well as audio via SIP gives many opportunities of practical use - in private sector and also for security purposes. Programming, maintenance and firmware upgrade is done through included web server. The unit is reachable from any place around the world easily.
Benefits · Modular system allows to connect 1 to 64 buttons · Two 16-digit numbers (IP address) with each button · Possible to connect two independent locks for door opening · Two codes for hanging up the unit from telephone · Two codes for door opening from telephone · Six code locks (password from buttons at the door) · Possibility to connect a numerical keyboard (can include 4-18 standard buttons) · Integrated heating of printed circuit · Ethernet – 10/100Mb with standard 10BaseT a 100BaseTx · Web server for remote configuration – BOA · Operating system – Linux 2.6 · SIP connection P2P or PBX network system · WEB – firmware upgradeable · WEB – interface for control and setup parameters
Models · IPDP-00 – basic module without button (only audio) · IPDP-00C – basic module with integrated camera · IPNC1,C2,C4 – additional button module with 1,2, or 4 buttons · IPDP-K – additional module with numeral keypad
Accessories · Cases for surface mounted · Cases for embedded mounted · Module with labels for companies names
IP Video Door Phone Intercom IPDP
Benefits · Modular system allows to connect 1 to 64 buttons · Two 16-digit numbers (IP address) with each button · Possible to connect two independent locks for door opening · Two codes for hanging up the unit from telephone · Two codes for door opening from telephone · Six code locks (password from buttons at the door) · Possibility to connect a numerical keyboard (can include 4-18 standard buttons) · Integrated heating of printed circuit · Permanent lighting through visiting cards · Included color camera · Ethernet – 10/100Mb with standard 10BaseT a 100BaseTx · Web server for remote configuration – BOA · Operating system – Linux 2.6 · SIP connection P2P or PBX network system · WEB – firmware upgradeable · WEB – interface for control and setup parameters
NetBorder SS7 VoIP gateway appliance for 4 T1/E1 ports with transcoding (1U) [Part.Number ss7-nsg-ap04]
NetBorder SS7 VoIP gateway appliance 4 T1/E1 and powerful transcoding capabilities available in a 1U box: ss7-nsg-ap04
The serie NetBorder SS7 VoIP gateway appliance is a cost-effective, turnkey solution offering from 4 to 256 T1/E1 lines at a single point code, and powerful transcoding capabilities. Up to 8 T1/E1 lines are available in a 1U box, up to 32 lines in a 2U box.
This product is ideal for applications such as connecting a private branch exchange to the legacy telephone network or providing multiple points of presence to a VoIP network.
The Sangoma SS7 Media Gateway provides full call control routing for SS7 traffic without the need for third party media gateway controllers or protocol converters. Full inter-working is supported across all VoIP and TDM protocols simultaneously, allowing this single multi-protocol TDM to VOIP gateway to be deployed interconnecting differing networks.
The compact, all-in-one design reduces footprint and eliminates the need to source multiple network components to handle media, signalling and routing.
SIGTRAN and MEGACO allows a distributed solution across multiple points of presence where SS7 Interconnect is required.
SNMP & Radius allows monitoring and management of NSG via both of these industry standards. A GUI provides convenient access to most configuration, monitoring and management functions, while a command line interface provides full access to management functions with a minimum of bandwidth consumption.
Effortless connection between PSTN SS7/TDM and VoIP networks
Producto Certificado por el fabricante (SANGOMA) para interconexión a la red de España con los estandar de homologación E1 SS7.
Alcatel IP 200 IP Phone Post-screen SIP, PoE and headphone jack. The IP200 SIP Phone Alcatel Temporis is a position with many economic caractéritiques (screen 2 lines, 2 SIP accounts management, headphone jack with dedicated button, self-supplied PoE ...). interoperability Alcatel Temporis IP200 SIP is a position with great interoperability, which allows it to easily fit behind most VoIP providers and IP PBX market. Effective management of calls Mute, hold, transfer, call forwarding, call transfer incoming-hook, redial, conference in 3 (all of these phone services must be validated with your operator).
Technical features:
Switchvox is Digium Unified Platform that not only offers IP Telephony to your enterprise, recording calls is included, video, fax, chat, presence, call queues, callcenter , mobility (iphone, blackberry) and many more features.
AA60 Version has 10 sip users licenses but you could upgrade up to 20.
Concurrent calls are 10.
Is offered in rackeable 1U Unit
Your could customize connectivity E1/T1, analog and sip users supported. It could autoprovision Polycom Phones .
You could check more features in
en https://www.digium.com/en/products/switchvox/
NetBorder SS7 VoIP gateway appliance for 8 T1/E1 ports with transcoding (1U) [Part.Number ss7-nsg-ap08]
NetBorder SS7 VoIP gateway appliance 8 T1/E1 and powerful transcoding capabilities available in a 1U box: ss7-nsg-ap08
Ampliación de 1 usuario para centralitas AA305 ,AA355, AA60 y AA65
Ampliación de 5 usuarios para centralitas AA305 ,AA355, AA60 y AA65
Ampliación de 25 usuarios para centralitas AA305 ,AA355, AA60 y AA65
NetBorder SS7 VoIP gateway appliance for 16 T1/E1 ports with transcoding (1U) [Part.Number ss7-nsg-ap16]
NetBorder SS7 VoIP gateway appliance 16 T1/E1 and powerful transcoding capabilities available in a 1U box: ss7-nsg-ap16
Ampliación hasta 100 usuarios para centralitas AA305 y 355
SIP-based Call Button This device is mounted under a desk or in a discrete location. If an event takes place, the user depresses the button. The Call Button automatically makes a call to a pre-set phone or extension number. When the called number answers, the Call Button plays and repeats a stored audio file. This stored audio file is uploaded by the administrator to meet the needs to the installed location.
Features * User downloadable message up to 80 seconds * Single button call to pre-set number * Continuous repeat of message * Call progress light * Event-controlled relay * Tamper sensor * Web-based setup * PoE-powered Use areas include: * Classrooms * Banks or financial institutions * Court rooms * Front lobby reception areas
NetBorder SS7 VoIP gateway appliance for 32 T1/E1 ports with transcoding (2U) [Part.Number ss7-nsg-ap32]
NetBorder SS7 VoIP gateway appliance 32 T1/E1 and powerful transcoding capabilities available in a 1U box: ss7-nsg-ap32
Con este cable podrá conectar su auricular de gama profesionar con conector QD a su PC sin necesidad de cambiar el auricular.
Simplemente cambie el adaptador.
Módulo BF2E1 para Berofix de dos puertos PRI (60 líneas) El bf2E1 es módulo de dos PRI (1E1=30 lineas) de RDSI para las tarjetas berofix. Puede ser configurado individualmente por puerto a NT/TE (es posible que necesite un cable cruzado opcional disponible ("bnE1Crosscable"). Terminación de línea (120/75 ohmios) seleccionable para cada puerto. Características: • 2 puertos PRI/E1 • Cada puerto puede ser configurado individualmente modo TE/NT.
Media Gateways Digium G100
One Span Digital T1/E1/PRI to VoIP Gateway Appliance (Europe)
The G100 is a single span T1/E1/PRI gateway that provides up to 30 concurrent calls of TDM-to-SIP, SIP-to-TDM or SIP-to-SIP media convergence. It includes integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.
Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for Asterisk connectivity. The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations. The Digium G100 VoIP Gateway includes a single software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. It is built to support TDM-to-SIP, SIP-to-TDM and SIP-to-SIP (transcoding) applications. In a TDM-to-SIP deployment the VoIP gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the VoIP gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.
Panasonic KX-UT113
Basic SIP Phone with 3 Line Backlit LCD Display
The Panasonic KX-UT range of SIP telephony terminals enhance personal communications through excellent HD quality audio on every phone, combined with easy access to powerful supportive features and applications. The range - from standard phones, offi ce key-sets, executive terminals and touch-screen Smart Desk application phones - addresses all requirements.
Panasonic’s reputation for design, quality, reliability and care for the environment, ensures an exceptional user experience wherever the terminals are deployed - as part of a “cloud based” service or with an IP PBX - in a business environment or in the home.
Panasonic Basic SIP Phone
beroNet offers the bero*fix card series, a powerful and flexible hardware solution to connect ISDN and GSM infrastructures to any SIP based VoIP system. It has a modular concept and supports all relevant VoIP technologies like Echo Cancellation, Codec Translation, Secure RTP and FoIP. The card is designed to run on every operating system. It will be installed as a normal network adaptor and every communication to and from the card will be done over ethernet.
The bero*fix base board is available for PCI or PCI Express card with different channel densities, depending on your usage:
• bero*fix 400 (4-16 channels) • bero*fix 1600 (16-64 channels) • bero*fix 6400 (64-120 channels)
Available modules to connect ISDN are : • BF4S0, 4xS0 • BF1E1, 1xE1 • BF2E1, 2xE1
Analog(FXS/FXO) and GSMN modules will be coming asap
The base board is able to carry 2 modules. The following combinations are available: • bero*fix 400 + 4xS0 (1xBF4S0) • bero*fix 1600 + 4xS0 + 1xE1 (1xBF4S0+1xBF1E1) • bero*fix 6400 + 4xS0 + 2xE1 (1xBF4S0+1xBF2E1) • bero*fix 400 + 8xS0 (2xBF4S0) • bero*fix 1600 + 8xS0 (2xBF4S0) • bero*fix 1600 + 1xE1 (1xBF1E1) • bero*fix 6400 + 2xE1 (1xBF2E2) • bero*fix 6400 + 3xE1 (BF1E1 + BF2E2) • bero*fix 6400 + 4xE1 (2xBF2E1)
Incorporating innovative technologies developed by pioneers in the worlds of audio and video conferencing, the COLLABORATE Room FHD, like other systems from the COLLABORATE Room series, offers ease of operation and the right mix of quality and features.
Soft-codec-based video conferencing – enabling flexibility, easy upgrades, future-proof scalability, low TCO (total cost of ownership),and openness to the world of unified communications
Unbeatable Return on Investment (ROI)
COLLABORATE Room FHD is a full-featured system at an unbeatable price. With the lowest cost in its class, its high-quality audio and video lets you “be there” without the need to pay for travel. Wherever you are, the multi-purpose design and ease-of-use of the system contribute to workplace efficiency.
COLLABORATE Room FHD Codec
PTZ & Full-HD (1080p) Camera
Panasonic KX-UT133
Media Gateways Digium G200
Dual Span Digital T1/E1/PRI to VoIP Gateway Appliance (Europe)
The G200 is a dual span T1/E1/PRI gateway that provides up to 60 concurrent calls of TDM-to-SIP, SIP-to-TDM or SIP-to-SIP media convergence. It includes integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.
Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for Asterisk connectivity. The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations. The G200 VoIP Gateway includes two software-selectable T1/E1/PRI interfaces and supports up to 60 concurrent calls. It is built to support TDM-to-SIP, SIP-to-TDM and SIP-to-SIP (transcoding) applications. In a TDM-to-SIP deployment the VoIP gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the VoIP gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.
Panasonic KX-UT123
Inyector PoE TL-POE150S
The PoE Injector TL-POE150S fully complies with IEEE 802.3af standard, and can work with all IEEE 802.3af PoE compliant PDs (Powered Devices) or PoE Receiver Adapters, such as TP-LINK’s TL-POE10R, or other equivalent product, to expand your network to where there are no power line or outlet, where you wish to fix device such as AP, IP Camera or IP Phone, etc.
Sangoma A101 1E1 PCI (A101)
The Sangoma A101 card is a single port T1/E1/J1 card to Voice and Data. Like all cards in the Sangoma Advanced Flexible Telecommunications product line, the Sangoma A101 is equipped with "crash proof", field upgradeable firmware.
Diagnostic Tools: WANPIPEMON, SNMP, system logs
Warranty: Lifetime warranty on parts and labour.
Production Quality: ISO 9002
Sangoma A101 1E1 PCI Express (A101E)
Warrant: Lifetime warranty on parts and labour.
IP Door Phone Intercom IPDP Antivandal (IPDP-00 AV)
The connection via SIP P2P protocol to different IP systems or LAN makes it very attractive for all company sizes. Audio via SIP gives many opportunities of practical use - in private sector and also for security purposes. Programming, maintenance and firmware upgrade is done through included web server. The unit is reachable from any place around the world easily.
Panasonic KX-UT136
Basic SIP Phone with 6 Line Backlit LCD Display
Sangoma A101 1E1 PCI + Echo Cancel (A101DE)
DSP Hardware Echo Canceller Daughterboard
Sangoma A101 1E1 PCI Express + Echo Cancel (A101DE)
A102 Dual Voice and Data Board Two-Port T1/E1/J1
The Sangoma A102 Card: Two ports of optimized voice and data over T1, E1, and J1 available with Telco-grade hardware echo cancellation.
Part of Sangoma's award-winning Advanced Flexible Telecommunications product line, the A102 uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world. The A102 supports up to 60 voice calls or 4.096 Mbps of full-duplex data throughput over two T1, E1, or J1 lines.
La gama de tarjetas Berofix de Beronet son una solución potente y flexible para integrar líneas RDSI (BRI/PRI) , analógicas (FXO/FXS) y GSM con cualquier sistema VoIP basado en SIP. Berofix no es el típico media gateway SIP ni una tarjeta estándar PCI/PCIe para la que necesite drivers propietarios, por lo que la llamamos "tarjeta gateway". Gracias a su especial diseño hardware, el sistema operativo detecta la tarjeta como una tarjeta de red clásica. Todos los drivers necesarios serán automáticamente cargados por el SO. Por tanto, Berofix es independiente del sistema operativo y se puede usar con Linux/Unix, Windows y MAC (testeado con Windows y Linux). Berofix está basado en un concepto modular y soporta las siguientes caracterísiticas de procesamiento de voz basadas en DSP:
- Cancelación de eco G.168/G.165 con detección de cambio del itinerario del eco de hasta 128ms
- Traducción de códecs: G.723.1, G.729a, G.726, G.711u/a
- Generación y detección DTMF
- Fax T.38 (V.27,V.29 y V.17)
- SIP over TCP con SRTP y TLS (disponible en Q2 2011)
- DSS1, EuroISDN
- Funcionalidades básicas de Q.SIG
La interconexión de tarjetas Berofix mediante Bus PCM permite bridging hardware para transmisión transparente de voz, datos y fax via un cable para bus PCM opcional. (Disponible Q4 2010)
La serie Berofix está compuesta por una placa base y módulos de línea que se insertan en la placa base.
Cada placa base admite 2 módulos de línea y está disponible en formatos PCI, PCIe o como caja externa. Densidades disponibles:
berofix 400 (4-16 canales)
berofix 1600 (16-64 canales)
berofix 6400 (64-120 canales)
Estas placas base se pueden equipar con los siguientes módulos de línea:
bf4S0, módulo de 4 puertos BRI
bf1E1, módulo de 1 puerto PRI
bf2E1, módulo de 2 puertos PRI
bf2S02FXS, módulo de 2 BRI y 2 FXS (disponible Q4 2010)
bf4FXO, módulo de 4 FXO (disponible Q4 2010)
bf4FXS, módulo de 4 FXS (disponible Q4 2010)
bf2GSM,módulo de 2 GSM (disponible Q1 2011)
Como complemento a placas base y módulos de línea, están disponibles los siguientes accesorios:
- bf4Bridge (Para usar todas las bocas RJ-45 de una placa base con un bf4S0, bf2S02FXS, bf4FXO o bf4FXS)
- Divisores UTP para extraer dos conectores de 4 hilos de cada conector de 8
- Cable bnPCM (para conectar placas base)
- bnE1Crosscable (para conectar puertos E1 con otros sistemas)
- bn19Bracket 19" (bracket para montar en rack las berofix boxes)
Gracias al concepto modular de berofix podrá mezclar libremente líneas analógicas, digitales y GSM en una única placa base.
Complete Care to solution Collaborate Room SD (1 year) Equipment and software manufactured by ClearOne. Repair and Replacement Service During the term of the purchased Agreement, and subject to the limitations in this Agreement, ClearOne will repair or replace the Equipment as necessary to correct any manufacturing defects in the Equipment which occurs during the usual and customary usage of the Equipment during the Service Agreement period. If ClearOne repairs your Equipment, you understand and agree that ClearOne may replace original parts with new or like new parts. Replacement parts will be functionally equivalent to the original parts. If the product is found to contain a manufacturing defect that is un-repairable, ClearOne may choose, at its sole option, to provide a new or factory certified, same or better, model as a replacement. Technical Support ClearOne will provide access to ClearOne’s Global Support Centers to assist with hardware and software product use, configuration and troubleshooting. ClearOne will use reasonable efforts to respond to you during ClearOne’s published support hours. Please refer to our website at https://www.clearone.com/customer_support for a complete listing of ClearOne Global Support Centers and their hours of operation.
Complete Care to solution Collaborate Room SD (1 year)
Um micro computador com uma solução poderosa
O design compacto do microUCS juntamente com Elastix ® como uma solução de comunicações unificadas, o torna um PBX IP poderoso que lhe permite monitorar as comunicações em seu escritório ou em casa de qualquer lugar. Fácil de instalar, administração poderoso como só ofertas Elastix ® e seu baixo consumo de energia, tornar-se uma solução eficaz muito seguro e custo.
O aparelho microUCS, vem pré-carregado com o Elastix, uma solução de comunicações unificadas Open Source que integra as melhores ferramentas disponíveis para sistemas IP-PBX em um fácil de usar interface de administração Algumas das principais características são:. Atendentes automáticos, correio de voz, dial- por nome Directory, música em espera, vários idiomas, Grupos de Trabalho, Fila de Chamadas, Chamada em Espera, exibição Caller, Teleconferência, Chamada em Espera / Transferência, estacionamento de chamadas, captura de chamadas, detecção automática de fax, Gerenciamento Remoto, ramais remotos, e muito mais.
Pequeno mas robusto
Apesar de ser um dispositivo pequeno, portátil e muito fácil de manusear, o microUCS possui uma caixa metálica robusta com a rigidez necessária para proteger contra colisões e choques elétricos *.
Muito baixo consumo de energia
Com apenas 3W de consumo nominal de energia, o microUCS oferece uma economia significativa nos custos de energia e operacionais, contribuindo para a conservação do meio ambiente.
Fácil de usar, sem custos de licenciamento
Fácil de implantar e gerenciar, graças a Elastix como o software pré-instalado poderoso, que não vai gerar custos de licenciamento.
Gerenciamento remoto
Graças ao software pré-instalado poderoso Elastix, você terá todo o seu fluido de comunicação empresarial e sob o controle de qualquer lugar.
Baixe aqui a versão mais recente do firmware para este dispositivo.
A102 Dual Voice and Data Board Two-Port T1/E1/J1 (PCI & Eco Cancel DSP)
Optional DSP Hardware Echo Canceller Daughterboard
Sangoma Netborder SS7 Media Gateway License up to 4E1/T1
SS7 to VoIP Media Gateway Software
Sangoma’s NetBorder SS7 to VoIP Gateway software provides full-featured, carrier-class VoIP deployments while leveraging the flexibility of standard computing platforms and operating systems. The software version of this product provides maximum flexibility for developing new or enhanced gateway products.
The NetBorder SS7 to VoIP Gateway allows telecom service providers to introduce VoIP in their networks in the most cost-effective and flexible way. This is simply accomplished by combining the software with Sangoma’s award-winning digital T1/E1 and transcoding boards on standard computing servers. The combination works as a full-fledged SS7 to VoIP gateway, with the flexibility and expandability of software.
VoIP to PSTN SS7/TDM network interworking platform
Flexibility
The solution supports up to 32 T1/E1 per server. For larger installations (up to 256 T1/E1), distribution across multiple servers provide maximum flexibility to support growth.
O Elastix Appliance ELX Series ® oferece a mesma Elastix ® software que você usou com produtores primários Hardware Telefonia IP. Certamente é a melhor alternativa disponível no mercado para empresas de médio pequeñay indústria.
Design Compacto
O design dos nossos aparelhos é simples e compacto tornando-os portáteis e fáceis de manter. Com caixa de metal 1U, 2U 1.5U e ter acessibilidade para expansões usando PCI ou USB usando ElastibankTM bancos de canais. Todos os nossos aparelhos são rack permitindo a integração rápida à sua infra-estrutura de rede estabelecida.
Nossos aparelhos são projetados para consumir a menor quantidade de energia normalmente exigido. Por um lado, isso permite uma poupança energyand das despesas operacionais sobre a outra equipe e contribuir para a conservação do meio ambiente.
Intetgración Capacidade Digital e Analógica
O Elastix Appliance ELX Series ® integrado analógico ou cartões digitais (FXO / FXS, E1/T1), de acordo com sua exigência. Estamos disponíveis para aconselhamento antes de encomendar uma implementação para que você explorar a funcionalidade do aparelho ao máximo. Todo o hardware vem pré-configurado e testado a partir de nosso armazém.
* Representa uma estimativa em um cenário básico
A104 Dual Voice and Data Board Four-Port T1/E1/J1
The Sangoma A104 Card: Four ports of optimized voice and data over T1, E1, and J1 available with Telco-grade hardware echo cancellation.
Part of Sangoma's award-winning Advanced Flexible Telecommunications product line, the A104 uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world. The A104 supports up to 120 voice calls and full-duplex data throughput over 4 T1, E1, or J1 lines.
Sangoma Netborder SS7 Media Gateway License up to 8E1/T1
Dimensiones (Largo x Alto x Profundo): 158 mm x 151 mm x 47 mm.
Sangoma Netborder SS7 Media Gateway License up to 16 E1/T1
Sangoma A108 Octal Voice and Data Board Eight-Port T1/E1/J1 A108
The A108 is part of Sangoma's family of Advanced Flexible Telecommunications hardware product line—it uses the same high-performance PCI interface that is providing superior performance in critical systems all over the world. The A108 supports up to 16.4 Mbps of full duplex data throughput or up to 240 voice calls using up to eight T1, E1, or J1 lines. With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The A108, like all cards in Sangoma's AFT family, is field upgradable with crash-proof firmware. Choose the Sangoma A108D and A108DE, equiped with world class DSP hardware to achieve carrier-grade Echo Cancellation and Voice Quality Enhancement functions for your telephone systems.
Ampliación de garantía a 3 años para AA350
Sangoma A108 Octal Voice and Data Board Eight-Port T1/E1/J1 A108E
The A108 is part of Sangoma's family of Advanced Flexible Telecommunications hardware product line—it uses the same high-performance PCI Express interface that is providing superior performance in critical systems all over the world. The A108 supports up to 16.4 Mbps of full duplex data throughput or up to 240 voice calls using up to eight T1, E1, or J1 lines. With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The A108, like all cards in Sangoma's AFT family, is field upgradable with crash-proof firmware. Choose the Sangoma A108D and A108DE, equiped with world class DSP hardware to achieve carrier-grade Echo Cancellation and Voice Quality Enhancement functions for your telephone systems.
Sangoma Netborder SS7 Media Gateway License up to 32 E1/T1
Sangoma A108 Octal Voice and Data Board Eight-Port T1/E1/J1 A108D
DSP Hardware Echo Canceller Daughterboard (included)
Sangoma A108 Octal Voice and Data Board Eight-Port T1/E1/J1 A108DE
Splitter-t1-e1-sangoma-a108 (cable)
Ubiquiti NanoStationM5: 5 GHz 2x2 MIMO AirMax TDMA Station
The original NanoStation set the bar for the world's first low-cost and efficiently designed outdoor broadband CPE. The new NanoStation M and NanoStation Loco M take the same concept to the future with new redesigned sleek and elegant form-factors.
150+ Mpbs real outdoor throughput and up to 15km+ range. Featuring 2x2 MIMO technology, the new NanoStation links significantly faster and farther than ever before.
Next Gen Antenna Design: New antenna array designs featuring 16dBi dual-polarity gain at 5GHz and 11dBi dual-polarity 11dBi at 2.4GHz. Both with optimized cross-polarity isolation and in a compact form-factor.
Dual Ethernet Connectivity: Secondary ethernet port with software enabled POE output for seamless IP Video integration.
Intelligent POE design: Remote hardware reset circuitry of NanoStation M allows for device to be reset remotely from power supply location. In addition, any NanoStation can easily become 802.3af 48V compliant through use of Instant 802.3af adapter.
AirOS V: Version 5 of Ubiquiti's AirOS builds upon the market leading intuitive user-interface loaded with advanced wireless configurations and routing functionality.
Specification:
Operating Frecuency 5470MHz-5825MHz
El modelo 6753i de Mitel es un terminal SIP de gama básica que proporciona un gran número de prestaciones y funciones basadas en los principales estándares del mercado. Posee un diseño atractivo y elegante, display LCD de 3 líneas, interfaz XML para el desarrollo de aplicaciones a medida y soporta hasta 9 líneas configurables de forma independiente. El 6753i es compatible con las principales plataformas de Telefonía IP del mercado
Prestaciones
• Navegador XML
El terminal 6753i dispone de un navegador XML que permite el desarrollo de servicios y aplicaciones a medida. Para utilizar esta prestación, es necesario utilizar la guía de desarrollo XML disponible en esta página. Esta prestación proporciona al terminal 6753i un potencial ilimitado para cubrir todas las necesidades del cliente a través de aplicaciones que interactúen con el teclado y la pantalla del terminal.
• Gestión avanzada de llamadas
Las capacidades de almacenamiento, agenda privada, registro de llamadas, lista de rellamadas, teclas programables y la amplia información mostrada en pantalla, hacen que el 6753i pueda ser gestionado con la simple pulsación de una tecla.
• Calidad de audio
Todos los terminals SIP de la Serie 67xxi poseen un altavoz manos libres full-duplex de excelente calidad. Los años de experiencia que Mitel tiene en el sector de las telecomunicaciones y la tecnología utilizada, garantizan la calidad de sonido en todos los productos desarrollados.
• Facilidad de instalación
La instalación, configuración y gestión de todos los terminales SIP de Mitel, se realizan de una forma muy sencilla, lo que permite un ahorro de tiempo y dinero a las empresas. Toda la Serie 67xxi dispone de un mini switch de dos puertos ethernet con posibilidad de alimentación del terminal a través de la red de datos (PoE estándar IEEE.802.3af).
The Mitel 6730i, a new member of the carrier-grade, open-standards based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.
Supported by a host of Mitel configuration options, XML development tools and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.
SANGOMA A200: 2-24 Port Expandable Analog Voice Card
The A200 Series of Analog Voice Cards deliver superior audio quality in a compact 2U card that expands from two to twenty-four ports using a single interface slot.
To reach this configuration, simply add additional Remora™ cards to any base card in the A200 Series. A single PCI or PCI Express slot hosts the connection and ensures common synchronous clocking for all channels.
Sangoma cards guarantee error-free faxing and click-free audio on analog-digital links.
Get the Fax on Sangoma Hardware
Mejore la gestión de llamadas en la Serie 6750i con los teclados de ampliación
El módulo de expansión M670i ha sido diseñado para incrementar la potencia y flexibilidad de los terminales SIP de la Serie 675xi. Es posible conectar hasta tres módulos M670i en los modelos 6755i y 6757i compartiendo la alimentación y la señalización con el terminal, sin necesidad de un cableado adicional. Es una opción ideal para operadoras, secretarias, agentes de centros de llamadas o directivos que necesitan monitorizar y gestionar un gran volumen de llamadas.
• Teclas configurables
El M670i está equipado con varios LED de estado y soporta una gran variedad de funciones programables, incluyendo teclas de línea, marcación rápida, indicadores de estado de ocupación (BLF), servicio no molestar (DND), etc.
Al compartir la alimentación y la señalización con los terminales de la Serie 6750i, el modulo M670i puede ser instalado sin necesidad de realizar un cableado adicional. La configuración se realiza por medio de la interfaz web del terminal o a través de ficheros de configuración.
• Expansión
Es posible conectar hasta 3 módulos M670i en los modelos 6753i, 6755i y 6757i de la Serie 6750i, ofreciendo hasta 108 teclas adicionales. Es posible empezar con un único módulo e ir incrementando el número de teclados según crecen las necesidades del cliente
For optimum echo-cancelled voice quality and enhancement choose the A200D Model with DSP daughterboard Hardware Echo Canceller. One DSP echo canceller on the A200D Base Card will support Telco-grade hardware echo cancellation on all channels up to the card's maximum configuration in 24 ports with no additional CPU load.
Optional DSP Echo Canceller Daughterboard on the A200D
Módulo Sangoma 2 FXO A200 (A200-FXO)
Module analog with 2 lines FXO compatible with all Sangoma cards A200
Módulo Sangoma 2 FXS A200 (A200-FXS)
Module analog with 2 extensions FXS compatible with all Sangoma cards A200
Tarjeta Sangoma rémora adicional A200
Adding an additional Remora card to your existing A200 allows you to add an additional 2 FXS/FXO ports, by installing one to two A200 modules. Each A200 can support up to 6 cards, including the base PCI card, for a total of 12 slots or 24 FXS/FXO available ports. This card connects to your existing A200 through the use of an external bus connector, which attaches to the rear of the card. It is therefore recommended that you install these cards next to each other in the chassis.
Sangoma A400: 2 - 24 Port Expandable Analog Voice Card
The A400 Series of analog voice cards deliver superior audio quality in a 2U card that expands from two to twenty-four ports using a single interface slot.
For optimum echo-cancelled voice quality and enhancement, choose the A400D Model with DSP Daughterboard Hardware Echo Canceller. One DSP echo canceller on the A400D Base Card will support Telco-grade hardware echo cancellation on all channels, up to the card's maximum configuration in 24 ports with no additional CPU load.
To reach this configuration, simply add additional Remora™ cards to any base card in the A400 Series. A single PCI or PCI Express slot hosts the connection and ensures common synchronous clocking for all channels.
Like all the cards in the award-winning Sangoma AFT Series Product Line, the A400 and Remora™ system's architecture is shared with Sangoma's A101, A102, A104, A108 and A200 cards, ensuring common 3.3 V or 5 V, high performance, universal PCI or PCI Express compatibility, and crash-proof field upgradable firmware to take advantage of enhancements, as they become available.
Architecture
Sangoma A400: 2 - 24 Port Expandable Analog Voice Cardon PCI-e
Like all the cards in the award-winning Sangoma AFT Series Product Line, the A400 and Remora system's architecture is shared with Sangoma's A101, A102, A104, A108 and A200 cards, ensuring common 3.3 V or 5 V, high performance, universal PCI or PCI Express compatibility, and crash-proof field upgradable firmware to take advantage of enhancements, as they become available.
Sangoma A400D: 2 - 24 Port Expandable Analog Voice Card with E.C.
To reach this configuration, simply add additional Remora cards to any base card in the A400 Series. A single PCI or PCI Express slot hosts the connection and ensures common synchronous clocking for all channels.
Sangoma A400DE: 2 - 24 Port Expandable Analog Voice Card PCIe with E.C.
Sangoma Remora™ A400RA: Expandable Analog Voice Card up 24 Ports.
Add a additional Remora™ card to A400 and connect backplane. Ensures common synchronous clocking for all channels.
Módulo de expansão para o SoundPoint IP650
Módulo de expansão com tela e 14 teclas
Epygi QX50 IP PBX with 16 SIP extensions, 2 FXO & 2 FXS (up to 50 SIP extensions & 16 simultaneous calls)
The QX50 IP PBX is designed for your small office or self-sufficient branch with as many as 50 IP devices. This solution offers two FXO analog PSTN connections, two FXS analog station ports and supports up to 16 IP phones. The QX50 allows you to make up to 16 simultaneous IP calls. Also, this phone system is equipped with a built-in router with LAN and WAN ports to manage your office's data network.
The QX50's analog phone network connection ports make it possible to connect fax machines or retail point of sale stations; however, your everyday calls are routed through IP phones in order for your company to take advantage of the cost savings and breadth of VoIP features. For small business environments that are looking to migrate from an analog Key System to an IP PBX, the QX50 and its Call Park enhancements can emulate Key System features, yet still allow you to take advantage of PBX functions.
When rack-mounted and paired with an Epygi QX Gateway, power redundancy provides added protection.
You can use in combination with QX Gateways:
New Epygi QX Line
Our new QX line consists of three IP PBXs and four Gateways. The QX products are more compact, fully rack-mountable and contained in a metal enclosure. The new products are meant to mix and match for a fully customizable system to fit every consumer's specific needs. With the purchase of a rack-mounting kit, the units also come with two DC power cables for power redundancy.
IP PBX appliance UCM6510
IP PBX appliance UCM6510, powered on the popular open source Asterisk based software platform, support enterprise level high calling volume (up to 2000 users, 200 simultaneous call, and 8 conference bridges with up to 64 attendees). It also features a number of improved reliability designs such as redundant power supplies, redundant Gigabit network ports with support for hot - standby high availability operation (pending software upgrade), 2 FXS and 2 FXO ports, integrated PoE, USB, SD, unlimited SIP trunk accounts with flexible call routing control, large 32 GB onboard Flash memory for voicemail, electronic fax, call recording, personalized music-on-hold, and virtually unlimited peering for multisite deployment or backup.
The UCM6510 is an innovative IP PBX appliance for E1/T1/J1 networks that brings enterprise-grade Unified Communications and security protection to enterprises, small-to-medium businesses (SMBs), retail environments and residential settings in an easy-to-manage fashion. Powered by an advanced hardware platform and revolutionary software functionalities, the UCM6510 offers a breakthrough turnkey solution for converged voice, video, data, fax, security surveillance, and mobility applications out of the box without any extra license fees or recurring costs.
Patton’s SmartNode Open Gateway Appliance (SNOGA) enables industry peers or integrators to simply buy the unit and install their software of choice, or partner with Patton to create a custom OEM solution for any number of applications. With the ability to run any IPPBX, soft switch, router, SBC, call recorder, or call accounting software, alongside any one of Patton’s robust Gateways, the SNOGA opens the door to endless possibilities and customer satisfaction, every time.
The Open Gateway Appliance is a convenient “one box” solution. With the built-in VoIP Gateway, the Open Gateway Appliance eliminates the interoperability obstacles for companies that want to keep their legacy equipment including PBXs, phones, fax equipment and POTS lines. Configuration options include a range of analog FXS/FXO and/or ISDN BRI/PRI gateways coupled with the industry standard miniITX PC board.
The starter pack includes these components:
1 Gateway with a Hybrid Module 2BRI/2FXS (BF4002S02FXS Box)
1 Appliance (L) with a Hybrid Module 2BRI/2FXS ( BNTA_2S02FXS_L)
1 Beronet Cloud account
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BERONET APPLIANCE
The beroNet Telephony Appliance is the ideal platform for customers and technology integrators, looking for a reliable hardware solution, with integrated ISDNa, Analog or GSM connectivity, in this STARTER PACK a hybrid module BRI/FXS is included.
The beroNet Telephony Appliance’s elegant design delivers superior energy efficient properties and is designed specifically for the rigorous 24/7 uptime demand of modern telephony systems. With a power consumption of less than 24W during normal operation, over 200.- EUR can be saved in electricity costs per year.
With its fanless design and harmonized components the beroNet Telephony Appliance is the suitable solution for all kinds of telephony projects. The beroNet Telephony Appliance is designed for telephony applications in the SOHO and SME markets and is perfectly suited for businesses with up to 60 concurrent calls.
Choose your favorite telephony software. The beroNet Telephony Appliance has been extensively tested with Linux and Windows operating systems and is compatible with all of these software solutions:
The Appliance beronet includes a USB pendribe which includes the executable from the most important software PBXL (3CX, Elastix, Asterisk...), each integrator must choose which of those PBX wanted to install in the appliance.*
*A software PBX must be installed in order to use the appliance Beronet.
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BEROFIX BOX
The berofix box is a powerful and flexible hardware solution to connect ISDN (BRI, PRI) and GSM lines to any SIP based VoIP system. The berofix box is the external aternative to the cards. Due to the modular concept you can mix PRI, BRI and GSM ports, in this STARTER PACK a hybrid Module (BRI/FXS) is included.
NOTE: Only one starter pack per integrator is included. Promotional price
1 port of FXS and 4 ports of FXO. Introductory series optimized for small-budget installations. With B-Series hardware, Sangoma’s premium audio and engineering quality is even more affordable.
For small, stable systems, when the modular design, single-slot expansion and port-customization power of Sangoma’s A200 and A400 series is not required. The Sangoma B600 voice cards deliver customers a substantial cost savings, coupled with the “because it must work” quality and the Wanpipe® high-end system diagnostics, drivers, and utilities for which Sangoma is known worldwide.
Your customers can achieve optimum echo-cancelled voice quality and enhancement by choosing models withTelco-Grade DSP Hardware Echo Cancellation. One DSP echo canceller on the B600D or B600DE voice card provides Telco-grade hardware echo cancellation on all channels with no additional CPU load and minimal additional cost.
D100: 30-400 Sessions
Voice Transcoding Boards
The industry’s first affordable PCI or PCI Express voice transcoding card — designed for optimum voice quality for low to high density system!
Overview:
IP telephony applications commonly require the use of multiple voice codecs, used to digitally compress voice signals and save on bandwidth. Voice signals from the Public Switched Telephone Network (PSTN) come in the form of the G.711 codec, but the VoIP terminal equipment and networks can support a variety of different voice codecs, such as G.729, G.726, AMR, G.723.1, G.722, iLBC, etc. The VoIP infrastructure needs the capability to mediate between endpoints supporting different codecs, but this functionality requires digital signal processing tasks that are often costly and resource-intensive, and can affect the quality of the voice signals, if it introduces too much latency and delay.
The D100 card, available in PCI and PCI Express form factors, converts simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows up to 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality¹. All codecs are fully indemnified; no additional licensing is required for their use².
The D100 works with both Asterisk and FreeSWITCH. With compatible drivers, these applications can use the D100 cards as seamless voice transcoding resources.
Sample Applications:
¹ Total port counts performance varies from codec to codec use and on the traffic mix at any given time.
² Except for AMR and AMR-WB
The beroNet Bridge (bfBridge) can be used to access all 4 RJ45 outputs of a berofix baseboard with only one bf4S0,bf4FXS,bf4FXO or bf2S02FXS plugged on the baseboard.
Use o comparador e escolha o produto que melhor se adapte a você !
Robusto y resistente, configurable con M700 y M300
DECT and desktop phone
Compatível com Panasonic HDV130
2 lines, LCD 2,3'', HD sound, 2 LAN ports
Wireless extension TPA60
Intelligent wireless IP Phone System
The new GXP1628 is based on Linux OS and includes 2 ...
Addcom USB Cord - QD to USB with DSP (ADDQD-76)
Sentinel es una plataforma de servicios múltiples VoIP con la capacidad ...
Digium's quad span digital interface cards: 4 T1/E1 trunks to ...
Tarjeta Digium TE235 Dual span digital cards in PCI-Express compatible with ...
Grandstream GXP-1625 (2 SIP, 3 XML, 2 ETHERNET + PoE, LCD ...
KX-A406 DECT repeater is an ideal solution to extend the range ...
The beroNet 1 PRI Gateway (S2M, E1)
GXP2170 executive IP phone with 6 SIP accounts, 2 Gigabit pots, ...
Android Enterprise Conference Phone
Elastix High Availability Add-On
PoE 802.3af, SIP, Full-Dúplex, 10/100Mbps
Videoconferencing Solution - GVC3200
The new GXP1630 is based on Linux OS and includes 3 ...
Alphatech IP BOLD door stations offer two relay contacts, PoE, full ...
Rain hood for Alphatech surface roof.
Grandstream GXP-1620 (2 SIP, 3 XML, 2 ETHERNET, LCD 132x48 backlit ...
Grandstream GXP-1400 (1 SIP, 3 XML, 2 ETHERNET, LCD 128/40, PoE, ...
Teléfono DECT Snom M25 con Pantalla Color, Jack 3.5mm de Headset ...
Teléfono IP Snom 715: Simplemente funcional
Bero*fix box up to 16 channels
Ecrã LCD a cores de 4,3 "e 430x270px
Epygi QXFXS24 Gateway is the perfect solution for use existing analog ...
snom Vision - the expansion module for the snom 8XX & ...
Epygi QX rack mounting kit
Lightest DE CT™ headset on the market
The snom 760 IP phone: High-level functionality coupled with a multitude ...
La GXV3611IR_HD es un cámara IP infrarroja (IR)
DECT Solution: M300 + M25
Expand room coverage by CPE80
Session Border Controller SMB
Electronic Hook Switch APV-62 for telephone GXP2124 and handset cs540
Compatible com T29, T46 e T48
Metal Grip for NLX4000
+ 18 programmable keys | You can connect up to 3 ...
Tela retroiluminada
Gateway Grandstream ATA HT802 (2 FXS+1 ETH)
1 SIP, PoE and 3-way audio conferencing
G.722, Full Duplex or Half Duplex, autoprovisioning, PoE
SIP loudspeaker for ceiling mounting | 2 SIP extensions
SIP (RFC3261) compatible | G.722, PoE 802.3af or 802.3AT
Personal conference system
Compatível com Yealink T48G
HD sound, hands-free, color display and jack 3.5mm
DECT handset (Compatible with Base Station DP750)
Color display, 12 BLF , gigabit and bluetooth compatibility
2 detachable microphones DECT
SIP/Android Video conferencing Solution Full-HD
Grandstream IP Phone GXP2135
PoE 802.3af, improved echo-acoustic canceller, full-duplex, SRST
Base Station M300 DECT for M25 or M65
Audioconference SIP for meeting business
DECT sem fio - SIP, analógico ou GAP, até 20 participantes
Konftel IP DECT 10 base station
Compatible with Yealink phones: SIP-T27P, SIP-T27G and SIP-T29G
Survival Branch Appliance
Compatible com A4B
Base Station (It supports up to 5 handset DP720)
Yealink wireless solution | Handset + Base Station
Gigabit, pantalla de alta resolución y teclas personalizables
12 SIP, Gigabit, 12 customizable keys LED and a 2º screen
4 SIP identities | Gigabit
4 SIP identities | 4 context sensitive keys and 5 configurable ...
Tela colorida, 128x160
Cable RJ21 plano a libre 10m
Compatível com T27P e T29G
Compatível com Yealink (T21P,T23P,T23G)
Compatible com A8B
The new videophone Yealink SIP VP-T49G HD incorporates new features that ...
1U - 500 calls - DualPSU
4 channels DECT coverage extender for multicell DECT Spectralink systems
Vega Enterprise SBC VM/Software: Secure, Interoperable, Flexible and Durable VM-Ready eSBC
El Konftel 300Wx con tecnología DECT permite establecer conferencias donde y ...
Vega400 E1/T1: E1/T1 Digital Gateway + 60 VoIP calls license
Vega400 E1/T1: E1/T1 Digital Gateway + 30 VoIP calls license
The ultra professional corded phone
Alcatel Temporis 580 (white) with handsfree, big display and headset mode.
Alcatel Temporis 580 with handsfree, big display and headset mode.
Temporis 380: 10 direct memory keys, handsfree, key in headset mode, ...
Alcatel Temporis 180: The professional telephone that keeps things simple.
Upgrade de 30 canales para el gateway vega 400 de sangoma
1U - 250 calls - DualPSU
3 x RJ45 10/100/1000 Full Duplex, 1 x USB-B, 1.4Gbps bandwidth
Bracket de perfil bajo para tarjetas Digium A4
Bracket de perfil bajo para tarjetas Digium TE435
Bracket de perfil bajo para tarjetas Digium TE235
Bracket de perfil bajo para tarjetas Digium TE13x
Grandstream Camera: GXV3674_FHD Outdoor Day/Night Full HD IP Camera. GXV3674 is ...
The GXV3662 series are outdoor weatherproof IP dome cameras that are ...
Digium A4A: The Analog Telephony Cards 4-Port on PCIexpress in your ...
Digium A4A: The Analog Telephony Cards 4-Port on PCI in your ...
GXP2140 is a IP phone up 4 lines, 4.3 inch TFT ...
Up to 8 hours talk time, 80 hours standby time
Polycom® SpectraLink® 8400 Series Wireless Telephones (Blue). Transforming Workflows with Mobile ...
Controls the traffic in the air and provides the link between ...
2 Gigabit ports - 10/100 / 1000Mbps, link / status LEDs ...
Vega Enterprise SBC: Security and Interoperability for the Enterprise
Digium A8A : La tarjeta analógica PCI para su sistema Asterisk. ...
Analog Telephony Cards for Asterisk system. Up to 8 modular Port ...
GXV3610_HD Day/Night Fixed Dome: powerful weather-proof Infrared fixed dome HD IP ...
Integration with SIP PBX and Skype for Business, Bluetooth, scanners de ...
Bluetooth, scanners de código de barras integrados | Spectralink SAFE
Integration with SIP PBX and Skype for Business, Bluetooth, 2.5mm connector ...
Mitel RFP 34 IP (outdoor). Complete integration of DECT radio networks ...
Módulo de cancelación de eco hardware para 128 canales compatible con ...
32 Channel Hardware Echo Cancellation Module
32 Channel Hardware Echo Cancellation Module for A4x, A8x telephony cards ...
Tela colorida 4 linhas retroiluminadas, som HSP
Compatible com MiniUCS B601
Monaural, alcance até 120m, até 13h em conversação
Una poderosa solución para Call Centers de fácil uso.
Digium PCI board with 1 E1/T1 port.
Digium PCIe board with 1 E1/T1 port.
Seamlessly integrate fax onto your BRI phone system using only one ...
Integration with SIP PBX and Skype for Business, Bluetooth, 2.5mm connector
The berofix Gateways are a powerful and flexible Hardware Solution to ...
DECT repeater single-cell and multicell bases
