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Link Primário E1 / T1 para VoIP SIP
Gateway VoIP Digital GXW4504
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 4 seção E1/T1/J1 e suporta 120 chamadas simultâneas para satisfazer as necessidades de VoIP de grandes e médias empresas.
Recursos em destaque
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.
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 15 simultaneous calls.
Features
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.
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:
Características destacadas
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
Gateway Khomp UMG 104 (BNC)
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:
Key features
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
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.
This UMG model is a compact system in which three basic parts can be defined:
Integration with IP PBX
Specifically, this UMG model is a compact system in which three basic parts can be defined:
Integración con PBX Tradicional
Gateway Khomp UMG 104 (RJ)
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
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.
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 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.
Gateway VoIP Digital E1/T1/J1
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.
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.
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
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 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.
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.
Applications
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.
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.
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.
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.
Features:
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.
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.
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
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
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
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.
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.
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.
Sangoma Netborder SS7 Media Gateway License up to 8E1/T1
Sangoma Netborder SS7 Media Gateway License up to 16 E1/T1
Sangoma Netborder SS7 Media Gateway License up to 32 E1/T1
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
- 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.
Use our comparator and choose the correct product
4PRI, 120 chamadas simultâneas | 2 X RJ45 | 2 x ...
1PRI (T1/E1) | Up to 15 simultaneous calls
1PRI (T1/E1) | Up to 30 simultaneous calls with HP-clock
1PRI
4 Ethernet network ports, 1 E1 / T1 link, echo cancellation, ...
2E1, BNC connector, 3 network ports 100 / 1000Mbps, expandable through ...
1 E1 / T1 with 30 channels, echo cancellation, 30 VoIP ...
Server + gateway, 2 USB, 1 VGA input, 1E1 / T, ...
1E1, RJ connector, 3 network ports 100 / 1000Mbps, expandable through ...
2E1, RJ connector, 3 network ports 100 / 1000Mbps, expandable through ...
1E1, BNC connector, 3 network ports 100 / 1000Mbps, expandable through ...
1PRI, 30 chamadas simultâneas | 2 X RJ45 | 2 x ...
2PRI, 60 chamadas simultâneas | 2 X RJ45 | 2 x ...
The beroNet 1 PRI Gateway (S2M, E1)
Vega400 E1/T1: E1/T1 Digital Gateway + 60 VoIP calls license
The SmartNode 10100 enables the delivery of VoIP services by bridging ...
Digium's G400 VoIP Gateway is a 4-port software-selectable T1/E1/PRI appliance that ...
Digium's G800 VoIP Gateway is a 8-ports software-selectable T1/E1/PRI appliance that ...
Vega 100G VoIP gateway connects digital telephony equipment to IP networks ...
Vega 200G VoIP gateway connects digital telephony equipment to IP networks ...
Vega400 E1/T1: E1/T1 Digital Gateway + 30 VoIP calls license
NetBorder SS7 VoIP gateway appliance for 4 T1/E1 ports with transcoding ...
NetBorder SS7 VoIP gateway appliance for 8 T1/E1 ports with transcoding ...
NetBorder SS7 VoIP gateway appliance for 16 T1/E1 ports with transcoding ...
NetBorder SS7 gateway VoIP para 32 T1/E1 con transcodificación en una ...
Gateway Digium G100 is a single span T1/E1/PRI gateway that provides ...
Gateway Digium G200 is a dual span T1/E1/PRI gateway that provides ...
Sangoma Netborder SS7 Media Gateway License up to 4E1/T1 (Software) to use ...
Sangoma Netborder SS7 Media Gateway License up to 8 E1/T1 (Software) ...
Sangoma Netborder SS7 Media Gateway License up to 16 E1/T1 (Software) ...
Sangoma Netborder SS7 Media Gateway License up to 32 E1/T1 (Software) ...
Bero*fix box up to 128 channels
Bero*fix box up to 64 channels
