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Primary Link E1/T1 for VoIP SIP
Gateway VoIP Digital GXW4504
The GXW4500 series are E1 / T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1 / T1 provider, businesses can drastically increase the amount of PSTN / ISDN trunks integrated with their VoIP network.
This particular model offers 4 E1 / T1 / J1 section and supports 120 simultaneous calls to satisfy the VoIP needs of large and medium companies.
Características destacadas
Founced in 2002, Grandstream Networks is the leading manufacturer of voice/video telephony and IP video surveillance systems. Grandstream addresses the needs of small and medium businesses through products that reduce the costs of communications, increasing security in order to improve productivity. Their products based on the standard SIP protocol offer wide interoperability in the industry, unrivaled features and flexibility.
Grandstream's IP Executive terminals are the next generation of IP Phones. They offer the highest telephony functions for businesses that need specific features like multiple lines and SIP identities, high-quality audio, integrated PoE and broad interoperability. They represent the perfect choice for businesses that need a high quality multi-line executive IP Phone at an affordable cost.
Grandstream's IP Standard Desktops are a new generation of ideal IP Phones as a working tool for small and medium businesses.
High-quality audio, customizable application service, soft keys, call waiting, call forwarding and transfer, tripartite conference and acoustic echo cancellation are some of the functions that have these great Grandstream handsets.
The Grandstream DECT series is presented as the new generation of high-quality, easy-to-use wireless IP phone. It offers a great variety of functionalities and a wide range of radio coverage.
Its compact size, excellent voice quality and excellent features make it a tandem with excellent value for money.
Grandstream offers several videoconferencing solutions at the user's disposal. We find, on the one hand, the series of Videophones IP of Grandstream, that combines in one device the benefits of an IP telephone with videoconference and the functionality of an Android tablet.
Incorporating a tablet's solutions gives you access to a multitude of Android apps, including business productivity such as Skype, Microsoft Lync, GoToMeeting, etc. A device that offers a multiplatform solution for all business communication and productivity needs (voice, video, data and mobility).
On the other hand, Grandstream also offers GVC3200, an Android based revolutionary videoconferencing solution, which supports multiple conferencing protocols from which we also find an adapted version for needs of small and medium businesses, the GVC3202.
Grandstream Analog Telephone Adapters (ATA) offer the ability to deploy commercial voice over IP services over an analog phone. Grandstream offers the following models to the user:
Grandstream video suveillance cameras are ready for any outdoor or indoor environment, where weather and light conditions are constantly changing. It lenses allow an adaptation according to the monitoring needs of the users allowing to monitor nearby areas, entrances to buildings, etc.
Grandstream has a wide range of Gateways ideal for any business that wants to incorporate the new IP technology into their daily activity. In addition, it is presented as the ideal solution to monetize the investment made in an analog telephone equipment, fax and tradicional PBX systems.
- GXW4004/4008 series represents the ideal solution for companies that wants to connect one or more lines from a traditional PBX to a VoIP telephone system with 4 and 8 FXS ports respectively.
- The GXW4104/4108 models convert SIP/RTP IP calls into traditional PSTN calls. Witn 4 and 8 FXO ports respectively, the installation is identical on both models and offers complete interoperability with IP PBX systems, softswhitches and SIP servers.
- The GXW4216/4224/4232/4248 has 16/24/32 and 48 analog FXS telephony ports respectively. In addition, they incorporate advanced security protection, excellent voice quality, easy provisioning and excellent performance in handling high volumes of voice calls.
In the portfolio of Grandstream, we also find this type of solutions, ideal for meeting rooms or meetings of any office thanks to its excellent mobility and audio quality. It has bluetooth, WiFi, 7-way conference and connection of a second GAC2500 unit in "Daisy-chain" mode with a secondary speaker and another microphone of expansion. That is why its flexibility and mobility are the main differentiating qualities of this audio conference equipment.
In addition, being based on Android 4.4 supports Google App Store applications such as Google Hangouts, Skype, Skype for Business (Lync), Internet browser, Adoble Flash, Twitter, Facebook, Youtube, Google Calendar, import / export data from Mobile phone via Bluetooth, etc. API/ SDK available for advanced custom application development.
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
The Mediatrix G7 is a reliable and secure VoIP analog adapter and a Gateway for SMBs.
Specifically, this Mediatrix G7 of 1 PRI offers the best solution for connecting equipment for cloud telephony services and IP PBX systems to PSTN terrestrial lines.
Widely interoperable with SIP softswitch and IMS providers, the Mediatrix G7 series offers seamless integration of legacy PBX systems for SIP trunking and PSTN applications. Among its most outstanding features we point out:
Key features
The Mediatrix 4400 series is Mediatrix's most economical solution that includes ISDN Gateways with BRI, ISDN and Ethernet ports making it the most robust and versatile option for businesses. This series of Digital Gateways for VoIP allows you to connect ISDN equipment, such as PBXs, via BRI interface to an IP network or as a gateway to the PSTN network.
The Mediatrix C7 Series offers a series of gateways that combine FXS and FXO interfaces to integrate multiple applications into a single platform. These VoIP Gateways offer excellent value for money and are ideal for connecting small and medium-sized networks to an IP network.
Models C710, C711, C730, C731 and C733 are currently available. Depending on the model it can present from 4 to 8 telephone ports and connect up to 8 analog phones, modems, fax or up to 8 RTC lines or trunks from a PBX to the IP network.
One of Mediatrix's flagship products is the Sentinel, an integrative product that has the capability of a customizable Session Border Controller (SBC) and Gateway. It is ideal for medium and large enterprises that need VoIP deployments with flexible architecture, including fault tolerance, SIP normalization and demarcation point.
The Base Station of our Sentinel initially presents:
Thanks to these 8 slots we will be able to customize our Sentinel as we want, combining the cards and licenses that best suit our needs.
MODULES
SBC Licenses
Mediatrix G7 - 2PRI
Specifically, this Mediatrix G7 of 2 PRI offers the best solution for connecting equipment for cloud telephony services and IP PBX systems to PSTN terrestrial lines.
Mediatrix G7 - 4PRI
Specifically, this Mediatrix G7 of 4 PRI offers the best solution for connecting equipment for cloud telephony services and IP PBX systems to PSTN terrestrial lines.
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:
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
This particular model offers 1 E1 / T1 / J1 section and supports 30 simultaneous calls to satisfy the VoIP needs of large and medium companies.
Gateway VoIP Digital GXW4502
This particular model offers 2 E1 / T1 / J1 section and supports 60 simultaneous calls to satisfy the VoIP needs of large and medium companies.
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 400G 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.
Patton Smarnode 10100
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.
Digium®, Inc., the Asterisk® Company, created and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999 by Mark Spencer, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Switchvox, Digium's Unified Communications solution to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of telephony hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom communications solutions. At Digium, we're changing the way businesses communicate.
Asterisk is the world's most popular open source telephony project. Under development since 1999, Asterisk is free, open source software that turns an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services.
Code for Asterisk, originally written by founder and CTO, Mark Spencer of Digium, Inc., has been contributed from open source software engineers around the world. Currently boasting over two million users, Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.
The Digium cards in the TE series are high performance, cost effective, with digital telephone interfaces that support the T1 and E1 environments. The environments are selectable on a base ny card or by port. This characteristic allows the translation of the signaling between the equipment T1 and E1, and allows to connect banks of economic channels T1 with circuits E1.
Sometimes it may happen that your IP communications systemn is echoed, this can happen as a result of the waiting times that a VoIP system often has, unlike an analog system. As a result, your conversation may suffer an echo.
Although many devices already incorporate the echo canceller, Digium offers the Digium High Performance Echo Cancellation (HPEC) hardware. Echo cancellation hardware is also advantageous when handling large volumes of calls or high numbers of channels that would otherwise strain the CPU which would result in poor audio potential quality. Here are some of its features:
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.
Vega100G: 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
Vega200G: 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.
Each E1/T1 interface can be independently configured as network side or terminal side. The Vega 400G gateway can therefore be connected to a PBX & the PSTN simultaneously. This configuration provides:
Integrated Bypass Relays For Resiliency The Vega 400G 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 400G & thereby providing dual redundancy.
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
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 1U box.
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
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 on one card with full below described features which makes the berofix cards very unique. The berofix Boxes supports the following hardware DSP based Voice-Processing features:
The berofix Boxes are available in 3 different channel densities: the berofix 400 (4-16 Channels), berofix 1600 (16-64 Channels) and berofix 6400 (64-128 Channels). See table below to check max. Channel densities depending of the Media Functions you are using. In "berofix Channel Option" below select your baseboard which fits to your demand.
Use our comparator and choose the correct product
4PRI, 120 simultaneous calls | 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, 5 Gigabit ports
2PRI, 5 Gigabit ports
4PRI, 5 Gigabit ports
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 simultaneous calls | 2 X RJ45 | 2 x ...
2PRI, 60 simultaneous calls | 2 X RJ45 | 2 x ...
The beroNet 1 PRI Gateway (S2M, E1)
Vega400G E1/T1 Digital Gateway + 90 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 ...
Vega 400G E1/T1 Digital Gateway + 30 VoIP calls license
Vega 400G E1/T1 Digital Gateway + 60 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 for 32 T1/E1 with transcoding in ...
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
