VoIP Gateways & Routers These VoIP gateway, VoIP router, & VoIP telephony solutions really work. Over seven years running Patton engineering has been
developing and refining SmartNode VoIP technology. And Patton puts our advanced VoIP technology into every SmartNode VoIP router and VoIP gateway we make.
SmartNode offers VoIP gateway and VoIP router solutions with virtually every VoIP telephony interface, including ISDN PRI & BRI, T1, E1, FXS, FXO, and more. Plus SmartNode supports SIP, H.323, and MGCP VoIP technologies. That means SmartNode VoIP technology solutions will fit your network.
Among VoIP technologies, SmartNode is mature. Tested and deployed in hundreds of real-world networks, SmartNode enterprise VoIP telephone and VoIP provider solutions are fully proven. That's why enterprises, carriers, and VoIP providers all over the world choose Patton SmartNode.
H323 SIP MGCP VoIP Gateway SysMaster VoIP Gateway/Class 4 Softswitch offers universal solution that allows dynamic call management in real time. All calls are routed via the available PSTN or VoIP channels based on custom routes for individual resellers. The gateway allows proxy services so that origination and termination partners are securely separated. The gateway also supports all PC-to-Phone customers with custom IVR prompts. Multiple Gatekeeper registrations are supported to several large termination providers for problem-free call management.
There are some feature follow here:- * H.323 and SIP protocols support * Up to 480 PSTN/SS7 channels and 1920 VoIP channels
* Registration in Unlimited Gatekeepers * Two IP channels in one call that can be used for VoIP proxy services
* All codecs capable of operating concurrently * Unlimited IVR profiles including ANI, PIN, PIN/ANI, pass-through authentication mechanisms
* Unlimited dialing plans/routing tables to allow custom routing based on GW IP address/subnet or channel type.
VoIP Gateways: VoIP that Fits Anywhere TalkSwitch VoIP gateways extend the reach of your TalkSwitch, extending the benefits of the 48-CVA to even your smallest locations. Not all locations require full TalkSwitch 48-CVA models on site. TalkSwitch VoIP gateways integrate locations with as few as 1 or 2 users economically, enabling you to use the Internet to call your far-flung outposts without paying for long distance.
Each small location requires only a TalkSwitch VoIP gateway to integrate with the main office system. Locations retain their traditional phone connection as well. Additional locations can be added incrementally at any time, making growth both easy and affordable.
And TalkSwitch gateways work with all standard analog telephones or our proprietary TS 100 telephone sets, which means you can have IP telephony without buying new, expensive IP telephones.
Gateways The Norstar VoIP Gateway is a VoIP trunk-side networking solution that enables voice applications to be handled over the data
infrastructure. The VoIP Gateway allows existing small multi-site Norstar customers to cost effectively enter into the VoIP environment while providing an opportunity for large multi-site companies with Norstar systems to slowly migrate their voice network to IP.
The Norstar VoIP Gateway is an external box that interfaces to the Norstar system over up to four loop start analog lines, providing a 10/100 BaseT Ethernet connection LAN.
There are few Key Features:- * Private Network Dialing * Caller ID from originating Norstar to destination Norstar user
* Interoperability with other Nortel Networks systems
* Distributed and supported by Nortel Networks
* Browser-based OAM
is VoIP Gateways A network device that converts voice and fax calls, in real time, between the public switched telephone network (PSTN) and an IP network. The primary functions of a VoIP gateway include voice and fax compression/decompression, packetization, call routing, and control signaling. Additional features may include interfaces to external controllers, such as Gatekeepers or Softswitches, billing systems, and network management systems. Short for Public Switched Telephone Network, which refers to the international telephone system based on copper wires carrying analog voice data. This is in contrast to newer telephone networks base on digital technologies, such as ISDN and
FDDI. Telephone service carried by the PSTN is often called plain old telephone service (POTS).
Media Geteway Reference Platform
The VoIP Media Gateway Reference Platform is an integrated white box media gateway designed to introduce Voice over IP (VoIP) application developers and system integrators to the flexibility and media processing-specific features of a gateway solution built on Intel products and Paraxip* Gateway software.
this integrated platform provides a flexible and configurable SIP gateway with the ability to host or support a co-resident SIP-based media application, and particularly applications developed using Intel
Net Structure Host Media Processing Software Version 2.0 for Windows*. It was created to provide a turnkey experience for development, interoperability testing, and pilot deployments.
VoIP Gateway Market Growth According to Research And Markets, the global service provider voice over Internet protocol (VoIP) gateway market at $165.3 million in 2003 is expected to reach 985.7 million dollars in the year 2009. The market is anticipated to grow at a steady positive rate over the years. The growth is brought by replacement of digital proprietary voice switching systems with systems that do manage to put voice over the Internet reliably and
clearly. Voice over Internet protocol (VoIP) can be implemented in two ways: calls can originate from the traditional TDM circuit switched technology, or originate from an Internet protocol router. IP to TDM and TDM to IP VoIP are very different. These fundamental differences in technology referred to as the same term (VoIP) have served to confuse the market, with distinctly different products being positioned as VoIP solutions.
Gateway Products For over a decade now the prospect of using the internet to carry voice calls has been ?next years technology?. Although there has not yet been any revolution in the way we route our phone calls, a number of enabling technologies, services and providers are now in place which can finally deliver a reliable, high-quality solution at very low cost.
Most businesses and individuals who are serious internet users now have un-timed and effectively un-limited connection to the internet. Users can spend all day downloading data from the other side of the world at no added cost. And yet, when those same users make a phone call they are charged by the minute, whether the call is local, national or international. In practice the data may well travel over exactly the same route, on the same wires, owned by the same people. Only the billing mechanism and price is different.
Gateway Voice Over IP Gateway has become effective and flexible Solution and this Solution is applied for the various offices data and voice connectivity. In additional to the concern on connective performance, reliability is also needed under the extreme circumstances. Network elements also have key role to perform.
Gateway enhances carrier services and also supports the transparency of the phone calls for lower cost and easy access. Flexible call integration is been developed at lower cost and easy access, which also helps for programmable call progress tones and distinctive ringing. It also helps in providing flexible numbering plans which helps in select the low cost route automatically having transparency at both the ends.
Analog Media Gateway The Media Pack? Analog Media Gateway product family is based on Audio Codes' field-proven and best-of-breed VoIP technology. Featuring 2, 4, 8 or 24 analog ports, the gateways connect analog terminals, PBXs or key systems to the IP network using FXO or FXS connectivity. Compliant with multiple protocols including SIP, H.323, MGCP and MEGACO, the Analog Media Gateways enable flexible deployment and interoperability for the evolving next generation networks. Using
Audio Codes' Analog Media Gateways, Network Equipment Providers and System Integrators can effectively deliver carrier-hosted converged services as well as enterprise-based applications.
VoIP Gateway Internet PBX We are a Culver City, California company that specializes in Internet PBX telephone systems that allow you to access one of the best VoIP gateways. PBX phone systems available at Fonality are affordably priced for small and medium-sized businesses. This Web site has been created to provide you with all the information you need to make the right Internet PBX telephone system choice for your business. We offer Internet PBX service and equipment including IP phones, servers and analog phone ports at affordable prices. Our skilled sales
representatives are available to help you select the PBXtra? product that best fits your needs.
At Fonality we are very proud of the Internet PBX, VoIP gateway, PBX phone system features we offer with our PBXtra? product line. Combined, they allow us to offer our customers the opportunity to take advantage of low long distance calling rates. With our VoIP network our clients will pay no long distance call
charges for the calls they make inter or intra-office.
BV 1260 VoIP Gateway The next generation in a growing line of award winning and industry proven Internet voice gateways, Oki's BV1260 4-port (FXO) incorporates an extensive set of new and refined high-quality features into a new compact, sleek unit that is easily installed on a desktop or rack. The BV1260 was designed to integrate seamlessly and unobtrusively into an existing infrastructure, and with a robust set of QoS features the BV1260 produces exceptional, Oki quality voice while maintaining a low profile on the network.
Standards based the BV1260 is H.323 compliant, in addition to supporting a full range of voice codecs including G.711, G.729, G.723. The gateway is equipped T.38 Realtime data flow control for facsimile transmission.
VoIP Gateways Designed for the enterprise market and OEMs, BOS Claro VoIP solutions are the preferred choice for sites requiring from just a few connections to mid-market sites with hundreds of connections. BOS VoIP products are distinguished by their seamless integration. For end users, this means absolutely no change in their familiar work environment, eliminating a learning curve. For enterprises, it translates into a more affordable, attractive investment, as VoIP products fit in with existing equipment, and demand no changes or additions. Delivering significant, measurable economies of cost, BOS VoIP solutions are the easy, affordable way to earn all the benefits of VoIP without the pain.
BOS VoIP Gateway technology provides the ability to connect telephone calls between sites over both public and private data circuits, achieving high quality voice calls at zero cost. BOS gateways act as the central pipeline in the system, transparently routing calls along the path of least cost.
VoIP Gateway Trunk and Carrier Based Routing Voice wholesalers use multiple ingress and egress carriers to route traffic. A call coming in to a gateway on a particular ingress carrier must be routed to an appropriate egress carrier. As networks grow and become more complicated, the dial plans needed to route the carrier traffic efficiently become more complex and the need for carrier sensitive routing (CSR) increases.
The Gateway Trunk and Carrier Based Routing Enhancements feature implements CSR for Cisco voice gateways. The gateway feature described in this document adds the following routing features:
* Implementation of trunk groups and enhanced key matches on several platforms and interfaces
* Reduction of the number of dial peers in a dial plan by using profile aggregation and multiple trunk group supports
* Enhanced hunting schemes * Call capacity updates on carriers and trunk groups
* Carrier ID support * Trunk group label support
VoIP Gateway The client is in the business of building a multi-service GSM/IP-based wireless access technology, using GSM, H.323, and core IP network technologies. The project implements a wireless access system, using the GSM air interface, that interfaces via an IP-based network infrastructure components to corporate and public voice / data networks, and provides voice services to the corporate / campus user.
The use of standard GSM terminals allows mobility, both within the corporate domain, as well as between the corporate and PLMN domains. An end goal for this product is to provide office-subs with one-number access to, as well as seamless roaming and hand over between, the corporate and PLMN domains. Later releases shall include packet data services via GPRS.
Quintum Announces New Tenor AF VoIP Gateway Quintum Technologies, a leading innovator in VoIP technologies, is pleased to announce their new Tenor AF switch, designed for 8 analog lines in both station side and trunk side configurations. The new Tenor AF supports up to 8 simultaneous VoIP calls.
The new Tenor AF has a smaller footprint than the current Tenor AS, and offers up to twice as much capacity. More importantly, it has a significantly smaller footprint than the Tenor AX model that currently supports 8 VoIP calls.
The Tenor AF, designed for small to medium sized enterprises, can intelligently route calls between the VoIP, a traditional PBX/analog phone or the PSTN. With its universal dial plan, the Tenor AF can support both public and unique dialing requirements.
The Tenor AF includes Quintum?s newest feature set of easy to use configuration and management tools, including the Tenor Configuration Manager (GUI) and Wizard, and the Tenor Monitor that offers
real time monitoring of alarms, call status and Call Detail Records. Tenor can easily be remotely managed behind NAT firewalls utilizing the new Tenor Remote Management Session Server.
How to Configure a IP/VoIP Gateway When you configure your telephony and data networks for Microsoft Exchange Server 2007 Unified Messaging, you must correctly configure the IP/VoIP gateways so that they communicate with the Exchange Server 2007 computers on your network that have the Unified Messaging server role installed. You must also correctly configure the IP/VoIP gateways to communicate with Private Branch eXchanges (PBXs) in your organization. This topic discusses how to configure an Intel
Net Structure PBX-IP Media Gateway, an Audio Codes Mediant 2000, and an Audio Codes
Media Pack Analog Media Gateway to communicate with a PBX. When you configure an IP/VoIP gateway, you must consider whether the IP/VoIP gateway device is analog, digital, or analog and digital. If the IP/VoIP gateway interface that connects to a PBX is analog, you must correctly configure the appropriate settings to enable the IP/VoIP gateway to communicate with a PBX. If the IP/VoIP gateway interface that connects to a PBX is digital, there may be no additional configuration necessary to enable the digital interface to communicate with a PBX.
TeleVoIPGate 120 TDM/VoIP Gateway Teleprime introduces the economical TeleVoIPGate 120, an advanced programmable & scalable gateway for connecting legacy telephone networks to IP. This full featured carrier grade gateway access system provides connectivity to your H.323 or SIP networks from a wide range of legacy circuit switched & PSTN type applications. Supporting FXS POTS, BRI-ISDN or T1/E1 PRI type ports, the TeleVoIPGate not only gives you IP access, you can support virtually any type of call mapping configuration for custom service offerings.
TeleVoIPGate is ideally suited for extending access to SS7 via the companion Signaling Gateway SP201 series products that map directly for flexible call routing and service offerings. Users now have an economical SS7 STP A link or switch access F link connection access to SIP or H.323 network solution.
Mediatrix 1402 ISDN BRI VoIP Gateway Allowing enterprises to lower communications costs over any IP link, the Mediatrix® 1402 BRI VoIP gateway provides four VoIP channels. The Mediatrix 1402 units are integrated VoIP gateways with two ISDN basic rate interface ports. It constitutes an ideal solution for LAN-based voice applications or for connecting to a service provider?s broadband access over DSL, WLL or cable. Mediatrix 1402 ISDN BRI VoIP Gateway
The Mediatrix 1402 BRI VoIP gateway provides private voice PBX networking and remote PBX extensions by connecting ISDN phones to the IP, and PBXs with each other through VoIP. Supporting four simultaneous calls from the IP network or the PSTN, the Mediatrix 1402 allows any office to use an existing IP network for lower-cost voice communications.
How to Configure an IP/VoIP Gateway
Unified Messaging Server You must configure the IP/VoIP gateway devices correctly when you are deploying Microsoft Exchange Server 2007 Unified Messaging for your organization. This topic discusses how to configure the interfaces of the Intel and AudioCodes IP/VoIP gateways to communicate with the computer running Exchange Server 2007 that has the Unified Messaging server role installed.
Exchange Server 2007 Unified Messaging supports the Intel Net Structure PBX-IP media gateway (PIMG) and AudioCodes Mediant 1000 and 2000 IP/VoIP media gateways, all of which, when configured correctly, can connect to a variety of third-party PBX systems.
The HTTP requests that are sent across the network when you are configuring an IP/VoIP gateway device are transmitted as unencrypted text. To increase the level of security for the IP/VoIP gateways on your network, use IP security (IPsec) or Secure Sockets Layer (SSL) to help protect the administrative credentials and data that is being transmitted over the network.