Australian SkypeIn numbers now available.

For those skype users who have been asking for Australian SkypeIn numbers the good days are now here. Australian SkypeIn numbers are now available, this was announced yesterday on skype blog. Australian SkypeIn numbers cost thesame as other countries i.e 10 euros for 3 months or 30 euros for a whole year. Also you can choose from among the following area codes, New South Wales, Australian Capital Territory, Nothern Territory, South Australia, Western Australia, Queeensland, Victoria and Tasmania. So if you have a need for an Australian SkpeIn number, go to your account page and get yourself one while stock lasts

Voipstunt lowers rates for more countries

Good times are here for users of voipstunt. Rates are getting lower once again. Now users of voipstunt can now call landlines in Albania, Bermuda, Dominican Rep., Gabon, Iraq, Laos, Paraguay, Serbia & Montenegro, South-Africa and Uruguay for just 2 cents per min.

Call Costa Rica, Bolivia , Lithuania and Reunion for just 1 cent per min

Voipstunt has now included Bolivia, Costa Rica, Lithuania and Reunion in the list of super deal destinations. Voipstunt users can now call landlines in these countries for just 1 cent per min.

Clear-Com Launches SoftVoICE 1.0 to Deliver Superior Intercom Quality

Clear-Com, an intercom solutions provider, announced SoftVoICE 1.0, intercom software for use on standard personal computers running Windows XP SoftVoICE works in partnership with VoICE (Voice over Internet Communications Equipment), a 1-RU 4-way VoIP interface frame that connects remote users seamlessly and efficiently over low-cost house LANs, private WANs, and other communication links using Internet protocols. Each VoICE frame enables an Eclipse Matrix to convert four of its physical station ports into SoftVoICE connections. Connecting remote personnel with fixed sites can challenge studios and switching centers because links are tricky and costly to coordinate while telecom solutions suffer quality and reliability issues. SoftVoICE, on the other hand, is as simple to use as an instant messaging client while delivering broadcast-quality audio from around town or around the globe. SoftVoICE significantly reduces the cost and complexity of audio connections between studios and talent in remote locations or tight spaces. And with advances in image compression and video streaming, nothing prevents the talent from receiving moving or still pictures on a desktop or laptop to fully engage in the conversation. The same holds true for public safety employees, live event engineers, critical business conference attendees, or even military forces.

Phihong's One-Port Midspan Supports High-Speed Wireless Access Points

Phihong USA has developed a one-port midspan for high-power devices that are capable of powering 802.11N access points. Typical applications for the 30W midspan include wireless/WiMax network access points, security cameras and IP telephones with streaming video. "Along with the new standard, customers are looking for equipment that is interoperable across multiple vendor platforms, "said Keith Hopwood, vice president of marketing for Phihong USA. "Our 30W one-port midspan provides Power-over-Ethernet technology that does just that." Phihong is a member of the IEEE 802.3af task force, which was formed to increase the power levels distributed via Ethernet to at least 45W. When implemented as a standard, the PoEPlus initiative will more than double the wattage available to powered devices. The 30W midspan features diagnostic LEDs and is fully compliant with the IEEE 802.3af standard in detection, disconnect and voltage control. The device is also Gigabit-compatible.

VBrick Adds Presentation Materials to Streaming Video Capabilities

VBrick Systems has added presentation capabilities to its Webcasting repertoire. VBCorpCast is the firm’s latest release and is designed to facilitate simple and even more effective streaming than ever before via the public Internet. Key features of the two new releases include the incorporation of live, synchronized Microsoft PowerPoint slides and guided Web browsing, as well as the addition of Windows Media Format to its streaming video format library. Both new kits, the VBCorpCast and VBEduCast — targeting corporate and educators respectively — enable users to initiate all multimedia presentation capabilities using the product’s Microsoft PowerPoint interface, eliminating the need to learn new applications. The VBrick Webcast kit is a complete, out-of-thebox presentation streaming solution that combines video with multimedia slides and Web content, as well as interactive audience polling and Q&A capabilities. The package comes complete with everything users will need to set up and complete live Web broadcasts — the VBrick appliance, a camera microphone, wires to connect the appliance, and 50 GB reflecting service, in case you don’t have enough bandwidth available for your presentation. VBCorpCast also records the entire presentation, including slides, questions, and online polling, and automatically archives it in the Internet for later viewing. At the completion of the presentation, a simple click of the mouse button will begin the archiving process, which takes only as long as it takes to upload the file to the server.

Easy, Secure Access from Anywhere at Anytime made possible by Encentuate

Encentuate Inc.,a provider of enterprise access security solutions, announced the release of Encentuate Web Workplace, a solution that simplifies, strengthens, and tracks access to information systems via Web portals, extranets, and browsers. The solution also ensures that single sign-on, strong authentication, incremental security and compliance tracking are available consistently wherever a user gets on the network, providing secure and easy access anytime, anywhere to authorized users. Encentuate Web Workplace expands Encentuate’s vision to provide identity management at the enterprise endpoints and complements the existing support for personal and shared workstations, Citrix, and Terminal Services. Web Workplace ensures that users need just one password and no installation of desktop software to access multiple information systems remotely from home offices and Web cafes or through browsers on handheld devices like PDAs and other mobile devices. Web Workplace also allows IT administrators to leverage the Encentuate IMS Server to centrally manage identities and access policies. One of the key benefits of Web Workplace is the ability to track and centrally report all access events to the IMS Server. IT can then develop comprehensive user-centric reports to audit access to information systems across enterprise end-points.

iPass and Nokia Develop WiFi Connectivity Client for Nokia

iPass Inc. and Nokia Enterprise Solutions (news - alert) announced they are developing iPass wireless connectivity software for the Nokia 9500, the Nokia 9300i, and the Nokia Eseries devices. Based on the iPassConnect universal client, this software will extend availability of the iPass secure remote access service to users of Nokia Business devices. In addition, Nokia began a user pilot of the iPass Corporate Access service, ahead of a planned company-wide roll-out around the later this quarter. Through the iPass service, Nokia plans to provide its remote and mobile employees with secure and reliable connectivity to company resources via the Internet. Users of the WiFi-enabled Nokia mobile business devices will be able to join the hundreds of thousands of active quarterly iPass users who can securely connect to the Internet and key corporate applications from approximately 50,000 WiFi venues worldwide. iPass Corporate Access gives remote and mobile workers a safe and simple way to connect to the Internet from over 160 countries around the world with dial, ISDN, wired, and wireless broadband connectivity. It offers a full range of services, including centralized policy-based management and end-to-end security enforcement that allow IT managers to maintain control over how users connect to the Internet without jeopardizing key corporate assets.

Citel Enables Gradual Enterprise VoIP Migration

Citel introduced the EXTender IP6000, a system that enables enterprises with multiple locations to gradually migrate their telecommunications to an IP telephony-based network. With the EXTender products in place, businesses will be able to connect remote call centers, home workers, and branch offices to a central digital PBX over an IP network. The system offers reduced telecom operating costs, single voicemail and call center applications, central reception, and four-digit dialing throughout the enterprise. As the business prepares to complete the migration to SIP, the EXTender IP6000 can be software-upgraded to accommodate the premises or service provider hosted IP PBX, integrating the existing handset and wiring infrastructure at each location. This process allows businesses to experience a smoother migration to SIP telephony in the future, without having to replace existing infrastructure.

Pandora and ESPRE Deliver Video Conferencing Solution

Media technology company ESPRE Solutions, Inc. (news alert) announced a licensing and integration agreement with IP communications service provider Pandora Networks for the delivery of a video conferencing solution. The agreement integrates video products from both companies: ESPRE’s eViewChat and Pandora’s Worksmart On-Demand IP Communications solution. eViewChat is an 8-way video conferencing solution. Worksmart is a software suite that gives small and medium-sized businesses control over communications services including voice, video, messaging, and collaboration. Integrating the two products means that Worksmart users will be able to “take advantage of ESPRE’s proprietary video compression to deliver businessquality video communications at far less bandwidth than that of competing video solutions,” the companies said. The integration includes ESPRE’s virtual private network solution, eViewNet, allowing users “to operate through corporate firewalls in order to make seamless video conferencing connections without the need to involve IT or other technical resources,” the release stated.

Sona Mobile Rolls Out BlackBerry Media Player Application

Sona Mobile Holdings is launching a BlackBerry Media Player software application, designed to offer multimedia applications on the latest generation of RIM devices. The new application will offer near TV quality layback of synchronized video and audio files. Sona Mobile provides secure mobile solutions for access to financial and enterprise information as well as gaming and entertainment applications. The company’s technology enables delivery of a rich client experience with top performance and security. The Sona Wireless Development Platform architecture and 3D mobile methodology work across a wide range of mobile devices and operating systems. These include Research In Motion’s BlackBerry and Microsoft Corp’s Windows Mobile lines. “For the very first time, BlackBerry users can receive either BerryCast (PodCasts wirelessly updated) or streaming video on their mobile devices,” said John Bush, CEO and president of Sona Mobile. “RIM customers take advantage of a downloadand-play method of delivering multimedia files to BlackBerry devices. We believe that this application will be well-received in the marketplace.”

Auto-LAN Configuration Device Simplifies Network Optimization

Global IP Telecommunications has successfully completed the proof of concept version of its software that will automatically configure LAN devices. LANbot resolves frequent discrepancies and redundancies that inevitably are found in network installations with multiple devices sharing limited resources. LANbot is a versatile LAN setup and installation utility designed to allow users of all levels to automatically configure their LAN devices for optimal performance. LANbot automatically resolves conflicts (e.g. port conflicts), configures both hardware and software, and provides a common user interface in at least three different languages. Global IP Telecom says LANbot will be the tool of choice for bringing new and existing customers online for ISPs. The patent pending LANbot can automate the analysis and setup of LANs. It can detect routers and all other network devices and, rather than require a separate user interface for each device, it provides a single uniform user interface capable of showing all devices and conflicts in the same window. The goal is to have just one button to configure the whole LAN automatically. When this button is pressed, the software analyzes all network and devices and automatically configures them with the most optimal settings.

XO Holdings Launches Nextlink

XO Holdings announced that it launched Nextlink, (news - alert) a new broadband wireless service provider that will offer services to mobile and wireline operators, businesses, and government agencies. Using licensed wireless spectrum covering 75 metropolitan markets across the United States, Nextlink will offer customers broadband wireless services as an alternative to conventional broadband services delivered over copper. Nextlink’s services will first be available in Dallas, Los Angeles, Miami, San Diego, Tampa, and Washington, DC, with additional market launches over the next two years. Nextlink is currently providing broadband wireless services to a national wireless company, delivering wireless backhaul and network redundancy and diversity services across markets in south Florida. In conjunction with the launch of its services, Nextlink also announced today the selection of Hughes Network Systems, LLC as its strategic wireless equipment supplier. Nextlink’s services are “fixed wireless” broadband offerings that rely on licensed local multipoint distribution system (LMDS) wireless spectrum in the 28GHz - 31GHz range. For locations up to seven miles and in line-of-sight of a Nextlink wireless hub, Nextlink provides wireless broadband services with speeds from 1.544 Mbps (T-1) up to 622 Mbps (OC-12). Nextlink’s services include wireless T-1, wireless metro Ethernet, and wireless dedicated Internet access.

Send and Receive Fax on Skype using PLUSFAX

PLUSFAX lets you send and receive fax with your Skype account. You can fax paper documents to any skype user using plusfax. The process of faxing a paper document to skype users is similar to faxing paper document to an e-mail address which i described in my previous post on plusfax. The only difference is that you enter the skype id instead of an e-mail address. Faxing digital document to any skype user involves downloading a virtual printer from plusfax. Once this virtual printer has been downloaded and installed you can then send fax directly from any MS Windows Application with Printing capability. Visit plusfax website for more information and trial.

Send Fax to any e-mail address using PLUSFAX

PLUSFAX let you send fax to any e-mail address easily without the need to download and install any software. You will send the fax directly from your fax machine. Its so easy to use, you go to the plusfax index page and click "fax paper document to any e-mail address". Select which of the plusfax routers you want to use, routers are available in USA, UK, France, Hong Kong and Italy. Enter the e-mail address you want to send the fax to and your own e-mail address. Print the plusfax cover page and place it on your document. Load your fax machine with your document and the plusfax cover page, dial the number of the router you selected and press send. Your paper document will be delivered to email recipients as a digital image that can be viewed using a web browser or displayed using standard imaging software.

SOHOware Announces Outdoor WiFi Solutions

SOHOware, an integrated wireless systems pioneer, announced new outdoor wireless solutions that help WISPs and other Service Providers quickly deploy public Internet access at multi-dwelling (MDU), hotel, and resort properties. The AeroExtend family further enhances SOHOware’s integrated MDU solutions by adding dual-band outdoor wireless capability, which combines robust 5.8GHz backhaul trunks with simultaneous 2.4GHz WiFi access. There are two radios in the AeroExtend device, providing dedicated bandwidth to the access point and wireless backhaul. The dual-radio, dual-band platform offers multiple benefits; AeroExtend installations require less number of devices overall per deployment, providing a more visually discrete solution and reducing initial capital investment for cost sensitive property owners. The AeroExtend family includes the WLG2502 dual radio AP-Bridge and a range of complimentary high gain antennas for 2.4 and 5GHz bands. Wireless network modes include point-to-point and point-to-multipoint bridging, plus simultaneous access point. The WLG2502 includes Power over Ethernet support and complete mounting accessories. Performance is 54Mbps for 11a and 11g, with turbo mode for 108Mbps.

Broadcom Ships Legacy-Friendly 802.11n WiFi Solutions

Broadcom Corporation, a producer of semiconductors for wired and wireless communications, has begun shipping its Intensi-fi wireless LAN technology to several top manufacturers of wireless networking gear. The Broadcom Intensi-fi draft-802.11n solutions provide increased performance wireless connectivity. More importantly, they ensure compatibility among next generation products from various manufacturers. Broadcom’s Intensi-fi chipsets combine high quality radio and digital technologies to deliver increased levels of wireless performance and reliability in the home or in the office. Intensi-fi’s throughput of more than 180 Mbps enable consumers and businesses to take advantage of emerging voice, video, and music applications on their wireless networks. Because of its performance and compatibility, Intensi-fi technology is expected to extend beyond PCs and networking equipment into consumer devices to enable an entirely connected digital home. Look for Intensifi technology to make its way into WiFi phones, storage drives, set-top boxes, broadband modems, TV and stereo equipment, gaming systems, and digital cameras. Not only is its Intensi-fi technology interoperable with a wide variety of 802.11a/b/g and draft802.11n devices, but it can even improve the performance of legacy WiFi products. This is music to the consumers’ ears, for it adds value to their previous investments and protects them from immediate obsolescence. Intensi-fi routers employ a “good neighbor” mode that ensures optimal performance in mixed networks when using the optional 40 MHz transmission mode, dynamically managing the use of 40 MHz channels and reverting back to 20 MHz channels when traffic is high or legacy clients need to communicate.

WildPackets Intros WiFi RF Environments Analyzer

VoIP analysis technology provider WildPackets announced the OmniSpectrum, a portable RF spectrum analyzer that runs in a standard Windows laptop PC and identifies the devices causing interference on a WiFi network. OmniSpectrum extends the existing capabilities of the OmniAnalysis Platform by making the 802.11 physical layer visible and intelligible, enabling network engineers to see which devices, WiFi and non-WiFi, are causing interference. OmniSpectrum identifies 802.11 devices, others that are generating signals in the 802.11 frequency band, and those that are unauthorized or transient. The system also identifies the class of the interfering device, like a cordless phone, and the manufacturers and model numbers of the offending equipment. This facilitates problem identification and resolution. WildPackets’ OmniAnalysis Platform is a distributed network analysis system for optimizing network services and maximizing uptime on enterprise networks. The platform uses analytical techniques, including network forensics and application performance indexing, to troubleshoot network problems.

3G Wireless Connectivity from Digi International and Sierra Wireless

Sierra Wireless and Digi International (news - alert) together will provide 3G wireless con nectivity for customers using Digi’s ConnectPort WAV VPN routers. Specifically, Digi, a networking solutions developer, will be offer compatibility with Sierra’s AirCard 860 and AirCard 850 HSDPA PC Cards, AirCard 775 EDGE PC Card, and MC5720 EV-DO PCI Express Mini Card embedded module. The Sierra Wireless products will provide 3G wireless connectivity Digi’s router product’s. Digi’s ConnectPort WAN VPN product is a commercial grade, upgradeable 3G router that provides high speed wireless connectivity to remote sites and devices. It can be used for primary wireless broadband network connectivity to equipment at remote locations, as well as for a backup to existing landline communications. It is a multi-functional product, able to perform multiple network tasks, like cellular router, firewall, switch, VPN appliance, and terminal server — all in one device. The ConnectPort router enables connection to a central site using a wireless wide area connection — via Sierra Wireless PC Cards and embedded modules — offering maximum flexibility to users when a wireline connection is not available. ConnectPort WAN supports both CDMA, EV-DO, and GSM HSDPA/UMTS 3G networks. Currently, the Digi ConnectPort WAN VPN is certified by Cingular and Sprint, with certifications with additional service providers expected soon. The AirCard 860 and AirCard 850 wireless WAN cards are Type II PC Cards that can be conveniently stored inside the laptop. The AirCard 860 utilizes the 850 and 1900 MHz UMTS frequency bands while the AirCard 850 utilizes the 2100 MHz UMTS frequency band and is targeted primarily for use in Europe. Both offer average data rates of 500 to 800 kbps, with bursts over 1 Mbps on HSDPA networks. In areas where HSDPA or UMTS network coverage is not available, the AirCard 860 and AirCard 850 will connect to EDGE and GSM/GPRS networks. The AirCard 775 fits into a laptop’s PCMCIA slot and provides data rates averaging 100 to 130 kbps, with peak rates up to 216 kbps. The AirCard 775 card also supports global roaming, functioning on EDGE and GSM/GPRS networks on all four GSM frequency bands used worldwide. The dual-band MC5720 PCI Express Mini Card, built using the MSM6500 Mobile Station Modem chipset from QUALCOMM, offers typical download data rates of 400-700 kbps, with peak speeds up to 2.4 Mbps. To provide coverage in areas where an EV-DO network connection is not available, the MC5720 module is also compatible with widely available 1X networks and will seamlessly switch without interrupting the user.

Venturi Wireless Upgrades PetroCom’s Gulf of Mexico Network

Venturi Wireless, which is focused on providing solutions that optimize the performance of wireless data networks for mobile operators, announced that mobile carrier PetroCom (news - alert) has deployed Venturi’s optimization solution within its Gulf of Mexico network. Venturi Wireless’ Adaptive Airlink Optimization is a cross-layer technology goes beyond simple application layer compression to virtually eliminate the inefficiencies of TCP at the transport layer. Thus, it can provide mobile data users faster access and a generally better experience — it can deliver access speeds up seven times higher with improved reliability and consistency. The patented Venturi Transport Protocol (VTP) lies at the heart of the solution. It was originally developed to cope with satellite data transmission issues, and has since been modified and transformed into a wireless solution. It overcomes the high latency and packet loss issues that result from poor RF conditions caused by weakened signal strength, interference, fading or high load. In addition to a better end user experience, Venturi Wireless optimization also maximizes spectrum and network resource utilization.

CTI Selects Netrake To Launch Caribbean VoIP Services

Netrake announced that VoIP carrier CTI (news - alert) has selected Netrake to support its launch of VoIP services throughout the Caribbean region. Netrake’s nCite session border controller provides CTI with secure traversal of firewall and network address translation (FW/NAT) systems, as well as denial of service (DOS) attack prevention through deep packet inspection at both the SIP signaling and VoIP media layers. Netrake’s session controllers also provide end-to-end quality of service (QoS) assurance and SLA enforcement for each call across network borders. CTI employs Netrake’s nCite session border controllers to provide enterprise and wholesale VoIP services across the United States, the Caribbean and Latin America. Based in Miami, Fla., CTI is offering lower cost yet feature rich residential & enterprise VoIP services. Additionally, CTI will leverage its extensive IP network to offer termination services to local carriers around the region, thereby providing those carriers with competitive pricing. CTI will also be able to offer users their own Virtual Private Network (VPN) for delivering calls on and off of the public switched telephone network (PSTN), such as ‘landing’ calls from one nation’s PSTN to another nation’s PSTN, as well as utilizing a unique four (4) digit dialing plan world wide.

Gizmo Project Now Includes VoIP Conferencing

Gizmo Project uses your Internet connection to make calls to other computers. With the click of a mouse, you’re connected to friends, family, and colleagues anywhere on earth. You talk clearly for as long as you want — for free. To address growing demands for conferencing capabilities, Gizmo Project has enhanced its VoIP services by tapping Vapps, a provider of VoIP conference calling solutions, for its conferencing platform in order to offer free worldwide conference calling services to Gizmo users. Vapps provides a carrier-class system for service providers and large enterprises to deliver high-quality reservationless, attended and operator assisted conferencing solutions. The solution is based on the company’s Conference Bridge 1000 (CB1000), a SIP-based conferencing platform that offers customized conferencing on a customer by customer basis both on legacy TDM and newer IP-based telecom systems. With the Vapps VoIP conference calling solution, Gizmo subscribers can connect multiple users over the Internet for cost-effective and reliable conference calls. All the features of a traditional conference calling service are available through Gizmo’s Web site.

IPTV Goes Straight to the Screen

NeuLion is expanding video possibilities with its IPTV, going beyond the desktop, straight from the Internet to the TV. The NeuLion iPTV platform is an end-to-end solution that uses the public Internet to stream multimedia content to any TV or PC from a common library. The result is DVD-quality streaming video — in real-time — to any device. The NeuLion iPTV platform encodes, delivers, stores, and manages an unlimited range of multimedia content, and the Operational Support System (OSS) maintains all billing and customer support services. The service delivers high quality video to the home or business over an existing high speed Internet connection, allowing viewing of high quality content, just as if it were being delivered by a local cable provider. But, without traditional geographical limitations, NeuLion’s platform can span the globe, connecting niche audiences and creating communities connected by common interests. What’s more NeuLion’s set-top box also supports interactive features, from VoIP connections to gaming and more. The key is that the NeuLion platform is both platform- and vendor-neutral — it can deliver video to any platform or any media device, using surprisingly low bandwidth rates. NeuLion’s content partners supply media and identify the marketplace. The NeuLion iPTV Platform encodes, delivers, stores, and manages an unlimited amount of multimedia content, and the Operational Support System (OSS) maintains all billing and customer support services.

AT&T Delivers Content to Broadband Customers

AT&T and Starz Entertainment Group have joined forces to offer SEG’s Vongo Internet movie delivery service to AT&T High Speed Internet customers. The service delivers movies and other video content over the Internet for playback on Windows-based PCs, laptops and select portable media devices as well as on TVs. The new agreement will feature a co-branded AT&T and Vongo Web site with a special 14-day free trial offer to AT&T High Speed Internet subscribers. The companies will also market the Vongo service on the AT&T Worldnet portal. “Vongo’s compelling content increases the value proposition for AT&T High Speed Internet customers,” said Scott Helbing, chief marketing officer-AT&T Consumer in a statement. “With Vongo, we’re positioned to deliver quality content, as we build a digital lifestyle platform for our customers.”

Seamless IP Networking Across the Great Lakes from Rogers, Sprint

Rogers, Canada’s largest wireless voice and data communications services provider, has signed an agreement with Sprint (news - alert) allowing busi ness customers with locations in both countries to operate on a seamless IP-based network through a new MPLS (multi-protocol label switching) network-to-network interface (NNI). The newly founded relationship allows Rogers’ customers to seamlessly connect to the Sprint network and enjoy a consistent communications experience north or south of Niagra Falls. Technology, SLAs, and support will all be seamlessly exchanged between the two providers. North American MPLS delivers both the flexibility and reach of the Internet and the security and performance of a private network. Business customers have secure connections offices, customers, partners, and suppliers, but do not have to purchase, operate, and maintain additional hardware — which makes for an efficient, cost-effective solution. The advantage of MPLS solutions is that companies will be able to avoid many capital expenditures and additional hires to maintain networks on both sides of the border. Cross-border offices will be able to operate seamlessly on an IP-based network, experiencing seamless connectivity with security, redundancy, and quality of service.

Globecomm Unveils SkyBorne IPTV Regional Headend

Globecomm Systems recently introduced its SkyBorne IPTV Regional Headend (RHE) for telephone companies and broadband carriers looking to add video services. The latest in the company’s line of pre-engineered systems, SkyBorne RHE provides content acquisition, content management, subscriber management, content packaging, and delivery to the carrier’s distribution network — all in a single, cost-effective, customizable package. Optional revenue-generating features include systems for advertising insertion and video-on-demand. According to Globecomm, SkyBorne RHE, which is based on field-proven modular building blocks, provides the performance, reliability and operational integrity of a custom broadcast system at a competitive price. It can adapt quickly and easily to changing customer needs. In addition to the Regional Headend, SkyBorne also includes lifecycle support services such as network monitoring, installation, field service and repair to preserve the value of the original investment and provide maximum uptime and throughput.

Boise State Adds SIP Trunking to VoIP Network

Time Warner Telecom has announced the successful installation of its SIP IP connections for VoIP services to Boise State University — what TWT says is the largest VoIP deployment in an American educational environment. The installation will entail running VoIP technology over Boise State University’s existing campus-wide metro Ethernet service — a Cisco Powered Network that encompasses more than 14,000 telephone numbers and 4,000 handsets. Time Warner Telecom’s 20 Mbps-capable SIP trunk service replaces the University’s existing T1s to boost bandwidth by nearly 20%. The scalable SIP installation will allow IT managers to connect directly to a VoIP PBX, thereby allowing the school to reduce the number of gateways it needs by six — all of which were required previously to convert digital voice signals into IP. TWT’s SIP trunking solution overlays the existing Ethernet LAN platform and is designed to seamlessly connect with existing Cisco VoIP applications. It is also highly scalable, which means that, as needs change, the school can expand the system again with relative ease and no disruption. SIP trunking entails using Session Initiation Protocol (SIP) for call control and routing, which enables a single IP-only connection to the carrier. Voice simply becomes another application across the IP network. In theory, this results in a pure IP to IP call if both end users are using SIP trunking.

Minacom and Electroline offer VoIP, Video, & Internet Testing

Minacom, a VoIP test solution provider for Cable MSOs, announced that its DirectQuality R7 Service Level Test Automation platform now supports remote testing to Electroline cable transponders, providing advanced VoIP, IPTV, Internet, and Fax over IP service quality and reliability assurance throughout the hybrid fiber-coax (HFC) network. Electroline statusmonitoring transponders support non-intrusive IP-loopback, in addition to packet, RF, and power supply performance and reliability monitoring. With this announced interoperability, operators can now use Minacom’s PowerProbe service level test probes running the RTP Streamer test agent to perform loopback tests to Electroline transponders supporting the SMRP protocol. Loopback testing provides a complete, concise service and network performance assessment by measuring bi-directional VoIP speech quality (MOS, R-factor), packet jitter, delay, loss, and other IP impairments, without interrupting transponder service transmission functions. Supporting Minacom’s awardwinning E-Model implementation, the RTP Streamer test agent allows operators to simulate triple-play traffic patterns with configurable codec, IP precedence, VLAN, jitter buffer and other key parameters to accurately replicate services and provide user-perceived quality of service (QoS) metrics.

Trenton Introduces System Host Boards With Dual-Core Processors

Trenton Technology has recently released several new PICMG 1.3 system host boards (SHBs) featuring dual-core processors from Intel. Trenton’s SLT is a server-class SHB that has two, Dual-Core Intel Xeon Processor LV 2.0GHz CPUs resulting four independent processing cores on a single system host board. The advanced power management capabilities of this new processor platform maximizes power efficiency, lowers system thermals and enables efficient high-density clustering of SHBs in a wide variety of telecom applications. Trenton’s TML is a graphics-class SHB featuring a single, long-life Intel Core Duo Processor T2500. This dual-core SHB supports a x16 PCI Express electrical link to a PICMG 1.3 backplane that enables full link support for the latest x16 PCI Express graphics and video cards. Another key feature of the TML is the ability to support RAID 0, 1, 5 or 10 implementations of the four, onboard SATA/300 interface ports. Both class of system host boards support multiple PCI Express links from the SHB to a PICMG 1.3 backplane. These high speed, high bandwidth PCI Express links provide multiple communication pathways to and from the SHB that are capable of supporting a vide variety off the shelf PCI, PCI-X and PCI Express option cards.

Elma Announces New 1U EasyPlug Compact PCI Chassis

Elma Electronic Inc., a global manufacturer of electronic packaging products, announced a new Type 39 1U EasyPlug CompactPCI enclosure. The Type 39c HA line features 9U cPCI backplanes from Elma Bustronic, which include pluggable 47-pin connectors for hot swapping power supplies. There are also pluggable fan tray headers and optional shelf management modules. Redundancy options can be built-in with all of these components. The backplanes are available in standard cPCI, H.110, or PICMG 2.16 options. With a sheet metal design and full pluggability, the Type 39c HA family offers ease of manufacturing and saves costs. The chassis are available in 1U-4U heights in horizontal-mounting orientations. Compliant to the latest PICMG specifications and IEEE 1101.10/.11, the enclosures feature side-to-side 200CFM (300LFM) cooling, 300mm depths, and rear I/O options.

Gig-E and PoE Stackables from Alcatel

Alcatel has introduced its OmniSwitch 6850 family of 10 Gigabitcapable stackable 10/100/1000 and 10/100/1000PoE workgroup switches. The OmniSwitch 6850s are the first Alcatel switches to provide Gigabit power over Ethernet (PoE) capabilities with extensive port diversity, full Alcatel Operating System (AOS) support, Gigabit to the desktop, 10 Gigabit uplinks, and enhanced performance for a flexible, intelligent, and highly available network. The OmniSwitch 6850L models provide Fast Ethernet to the desktop and are software upgradeable to Gigabit. This latest release continues Alcatel’s strategy to provide enterprises a cost-effective means for migrating to Gigabit Ethernet desktop connections through a software key. The OmniSwitch 6850s and 6850Ls protect enterprises’ investments by providing for future needs, such as advanced edge security, support for user mobility and dual IPv4/IPv6 capability. The switches’ versatility afford customers the flexibility to buy only what they need today without sacrificing performance they will require in the future. They also will benefit from the wirerate layer 2 switching and layer 3 routing supporting optimal QoS for voice and video.

Acme Packet Intros SBC Features

Session border control provider Acme Packet introduced additional features and a partnership to offer a suite of session border controller capabilities to enable cable voice, video, and multimedia services. The company now supports an ENUM server interface and has interoperability in field deployments with Nominum’s Navitas, an ENUM-based IP-application routing directory server. Acme Packet is also expanding its hosted NAT traversal mechanisms with support for the IETF standard STUN (Simple Traversal of UDP through NATs), STUN-Relay/TURN (Traversal Using Relay NAT) and ICE (Interactive Connectivity Establishment). Acme Packet is currently in commercial production at 4 cable operators in the U.S. and at some cable companies in Europe. For these operators, Acme Packet’s NetNet session border controllers provide access to the network edge and interconnect borders to protect and secure the MSO service infrastructure. Acme Packet’s ENUM server interface for the Net-Net SBC platforms supports private and public ENUM directory services and enables MSOs to use IP transport endto-end for VoIP calls between subscribers on different networks. This eliminates any PSTN transport.

JDSU Tackles High Capacity Networks

JDSU has announced that its IP network troubleshooting and data analysis platforms, DA-3400 and DA-3600A, can now perform VoIP call quality monitoring on high capacity networks. The DA-3400 is able to support 8,000 simultaneous calls and the DA3600A can support 64,000 calls to provide accurate and high-quality measurements when network load or utilization is at its highest. The enhanced DA platforms also generate important industry metrics used to help ensure VoIP voice quality, such as MOS and R Factor. These can be measured for all calls on a gigabit Ethernet circuit that is being utilized 100% by VoIP traffic. Other features include display filtering, display segmentation, and users configuration of the DA line for simple network management protocol or e-mail notification so they are alerted to any network quality issues quickly. Part of JDSU’s Service Assurance Solutions portfolio, the DA-3400/3600A’s new VoIP features also include the ability to show signaling details of VoIP calls using a graphic display that captures signal/packet set-up and tear-down during a VoIP call exchange. This feature also details the timing and response code, delivering prompts that warn the user the VoIP signal did not go through.

TANDBERG 3G Mobile Video Applications Drive Revenues

Assisting service providers in delivering video content and interactive videoconferencing to mobile consumers, TANDBERG introduced 3G video capabilities that enable service providers, including mobile operators and content providers, to differentiate their offerings and maximize their revenues. With the TANDBERG Video Portal, service providers manage and stream live or archived video content such as news reports, sports highlights, and movie previews to 3G mobile users. The intuitive drag-and-drop interface makes it easy for a service provider to build a customizable user experience that 3G mobile users navigate to quickly access content over their mobile phones. Service providers seamlessly manage content from multiple content providers with the Video Portal, and content usage statistics are easily downloadable for invoicing and statistic purposes. In addition, anyone can record content to the Video Portal for applications such as real estate services and video dating from any 3G mobile phone, or a SIP or H.323 device. Furthermore, 3G content can be live streamed to the internet or television studios. Videoconferencing is another innovative mobile video application for 3G mobile users. Whether responding to emergency situations, remote troubleshooting or participating in multiple-site business meetings, 3G mobile users have the ability to interact face-to-face instantly from anywhere. With the TANDBERG 3G Gateway, service providers can offer mobile videoconferencing to their customers using a scalable and redundant solutions built to integrate with carrier infrastructures using E1/T1 or SS7. Diverse billing rates based on premium numbers or video short codes give service providers a flexible solution for maximizing call revenues.

ARRIS and UTStarcom Team on FMC Solution

Converged network solutions provider ARRIS announced today a three-part, wireless-related agreement with end-to-end networking solutions provider UTStarcom. (news - alert) The agreement — which covers development, licensing, and supply — will enable “the fourth leg of the quadruple play for cable Multiple System Operators (MSOs) worldwide,” according to ARRIS. ARRIS and UTStarcom’s joint solution will enable seamless roaming between cellular and WiFi connections for end users with dual-mode handsets. With the agreement, ARRIS’ FMC solution—which enables MSOs to add wireless telephony to their service offerings—teams up with UTStarcom’s Continuity FMC solution, designed “to increase network efficiency and coverage through either wireline or wireless networks,” ARRIS said in the press release. ARRIS will license UTStarcom’s FMC software for an initial term of three years.Both companies have specified rights to create new features for emerging FMC solutions. Finally, UTStarcom will provide ARRIS with hardware and software to sell to cable MSOs. ARRIS is granted “exclusive marketing rights to the global MSO community” and UTStarcom retains “exclusive marketing rights to the telco community.”

BroadVoice Intros WiFi/Cellular Phone

VoIP service provider BroadVoice announced that it will begin selling a wireless VoIP/GSM phone, code-named “Falcon,” this summer. The BroadVoice Falcon will enable users to use the company’s unlimited VoIP calling plans utilizing WiFi networks. When the BroadVoice user is outside WiFi coverage, the Falcon switches to a cellular network and works as a mobile phone using a separate account with a GSM service provider. “Whether you’re at home, at work, or at a coffee shop, airport, hotel, or other location with a public hot spot, you can use your BroadVoice Falcon phone to make unlimited calls anywhere in the United States, and up to 35 countries, without paying for minutes,” said Gene Cornfield of BroadVoice. “When you’re on the move and WiFi isn’t available, the Falcon automatically uses the GSM cellular network, just like a mobile phone.” Once a customer buys a Falcon phone with BroadVoice service, they can select and activate a telephone number online and begin making and receiving VoIP calls over WiFi immediately. As soon as the customer plugs in a standard SIMM card from any GSM900, GSM1800, or GSM1900-based cellular carrier with whom he has an account, he can place cellular calls without having to contact the cellular service provider. Calls are received on either the user’s BroadVoice number, or the number assigned by his cellular carrier.

3 Rivers Chooses Pannaway’s Convergence Network Solution

Pannaway Technologies has won a major deal with 3 Rivers Communications, which will be using Pannaway’s broadband access system for the delivery of new next-generation services — including IPTV — to its rural subscribers. The cooperative will use Pannaway’s ADSL2+ and Active Ethernet FTTH solution to deliver a robust set of triple play features, along with the necessary bandwidth capacity to add future serives. Pannaway’s copper and fiber-based access solution will deliver SIPenabled triple play, including Primary Line VoIP for guaranteed Lifeline calling and E 911. Broadband Aggregation Routers (BARTM) will be deployed in the cooperative’s central office (CO) for scaleable 1 to 10 Gigabit Ethernet transport, while Broadband Access Switches (BASTM) will reside in remote terminals (RT) for high-performance last mile voice, video and data delivery. The ILEC will also use Pannaway’s Broadband Access Manager (BAM) to simplify the deployment and provisioning of its new copper and fiber-based network. The use of IP-Ethernet and SIP allows Pannaway to deliver improved video quality with enhanced rate/reach benefits. Enhanced network performance coupled with 10Gbps transport scalability ensures that telcos’ networks are future-proofed, that they will have the performance and capacity to support emerging bandwidth-intensive services.

Fusion Files Patent Application for its VoIP

Verso Technologies, a global provider of next generation network solutions, announced the introduction and The I-Master application adds value to both vertical and horizontal markets, offering a standards-based interface using SIP to third-party softswitches and RADIUS interface upstream for AAA (Authorization, Authentication and Administration) functions. The offering is a real time call control and service logic solution with embedded IVR functionality supporting a wide array of call flows. A user-friendly Graphical User Interface (GUI) enables operators to easily create and manage customized call flows The combined deployment of Verso’s I-Master Revenue Assurance application and the application server enables realtime authentication, rating and call control for calling card, fixed line ADSL, dial-up, WiFi, and enhanced IVR services, resulting in increased customer loyalty and revenue generating ability for any next generation network (NGN) environment. Additionally, the I-Master Application server supports account balance preauthorization to enable simultaneous concurrent service usage on Fusion Telecommunications announced that it filed a patent application with the United States Patent Office for its Directed SIP Peer-to-Peer (“DSP”) technology, acquired by Fusion in February 2006. The patent application describes a system that Fusion plans to utilize to provide its free service between SIP devices. Fusion is incorporating its DSP technology into the Company’s international network for Internet voice calls between any combination of computers, Internet connected telephones, wireless devices, and other SIP-enabled hardware. Fusion believes its new “efonica” branded softphone and uniquely configured VoIP network will provide significant advantages over most VoIP peer-to-peer networks. Fusion’s technology eliminates the method of routing utilized by many VoIP peer-to-peer networks, in which many users’ Internet bandwidth and/or PCs are utilized as part of the carrier’s larger network to set up calls for thousands of other users. “Our DSP technology should be of particular interest to security conscious individuals and businesses, a fast-growing segment as SIP is quickly becoming the de facto VoIP standard for communicating between VoIP hardware devices. We believe our entire communications package should be particularly advantageous for Fusion’s primary target of the emerging markets of the Middle East, Asia, Latin America, Africa, and the Caribbean, and their related communities of interests,” said Matthew Rosen, Fusion’s President and CEO.

Siemens HiPath ProCenter Helps Improve First Contact Resolution

Siemens Communications announced HiPath ProCenter Enterprise Version 7.0, a new Internet protocol-ready contact center solution designed to help enterprises improve first contact resolution, drive up productivity and increase customer satisfaction. The solution includes pre-built integrations with front-office customer relationship management (CRM) applications from Microsoft Corp., SAP AG and Siebel Systems. The Siemens HiPath ProCenter Enterprise solution helps enterprises improve the efficiencies of multiple customer interaction channels — including voice, e-mail, and live Web interactions — with Siemens’ award-winning presence and collaboration tools. With presence-driven applications, front-line agents can get real-time information about the availability of subject matter experts and connect to them across various media types throughout the enterprise. “Contact center industry research continues to demonstrate that first contact resolution is a key driver of customer loyalty, revenue growth and operating cost effectiveness,” said Al Baker, vice president, Enterprise Division, Siemens Communications Inc. “This presence-driven solution makes first-contact resolution possible even for highly complex or sensitive customer interactions.” The solution’s presence and collaboration tools help drive first contact resolution via Team List and Team Bar features that enable agents to view real-time presence and availability states of peers, managers, and experts outside the contact center. Available users can be included in call transfers, conferences, or consultations with just a mouse click.

InterTel’s New 7000 Communications System

The new Inter-Tel 7000 system can scale up to 2,500 users per site and is designed as a pure, standards-based communications platform. Using SIP technology at its core, the system is redundant and secure and offers an easy-to-use interface for remote management and configuration. The new platform will also offer full PBX-style features as well as a number of enhanced features like embedded presence management and mobility. It is designed to support Inter-Tel’s advanced IP-powered applications like contact center and messaging solutions, collaboration and the company’s existing lines of multi-protocol IP and SIP endpoints. “Inter-Tel has done a masterful job in developing a platform that is being designed to successfully leverage advanced IP technology to provide a rich, intuitive feature set that can be an asset to any business,” said Mark Ricca, a partner with Intellicom Analytics. “Standards-based software solutions, with SIP at the core, will enable organizations to avoid obsolescence by leveraging the communication options that Internet standards make possible,” noted Allan Sulkin, president of TEQConsult Group. “The fact that the Inter-Tel 7000 delivers presence capabilities as part of its core feature set differentiates it from other solutions in the market,” said Rob Arnold, an industry analyst with Current Analysis. “This is a real incentive for businesses looking to leverage these tools without having to add servers or additional software.”

Nortel Communication Server 1000

Communication Server 1000 is a server-based, full-featured IP PBX, providing the benefits of a converged network plus advanced applications and over 450 world class telephony features. Fully distributed over IP LAN and WAN infrastructure with built-in reliability and survivability, Communication Server 1000 supports business-critical applications, including unified messaging, customer contact center, IVR, wireless VoIP and IP phones. CS 1000 is designed to scale to meet growing enterprise requirements up to 15,000 IP clients per call server, multiple Call Servers networked with transparent IP networking. Built-in reliability based on VXWorks operating system and proven feature set with multiple resiliency mechanisms, including survivable Call Servers, Signaling Server redundancy configurations and sur vivable WAN gateways. Extensive desktop portfolio includes IP phones, software phones, 802.11 Wireless VoIP phones, as well as digital and analog phones to meet diverse end-user requirements supports business-critical applications, including IP Contact Center, CallPilot unified messaging, and integrated services such as conferencing, one-number-follow-me Personal Call Director, recorded announcement, network-wide attendant and messaging.

NEC’s UNIVERGE NEAX family of voice systems

NEC’s UNIVERGE NEAX family of voice systems has been designed for a lifeline rather than a life cycle. Relatively simple upgrades have kept its systems current in technology and functionality, while preserving 80% or more of the customer’s existing investment. The UNIVERGE NEAX 2400 Internet Protocol eXchange (IPX) fuses existing NEC technologies with dynamic advancements in hardware and software to satisfy the most stringent system requirements. All of NEC’s net working services and Dterm Series E digital telephone features are provided when deployed over an IP network. Peer-to-peer switching is also introduced in the NEAX 2400, directly connecting all stations participating in a call to each other. The SV7000 provides over 780 serv ice features that enhance productivity, reduce operating costs, and improve communications efficiently. Innovative hardware and software design allows it to serve efficiently and grow incrementally over its entire size spectrum, ranging from 50 ports to 16,000 ports. Expanding from its minimum configuration to its maximum capacity with virtually no loss of existing hardware, The SV 7000 can grow in a cost-effective manner along with the user’s requirements.

Nero SIPPSTAR IP PBX System

The SIPPSTAR IP PBX is an innovative, cost-effective, modern, software-based alternative toexpensive conventional telecommunications systems that makes it possible for you to use innovative VoIP technology in your business. You have the option of operating the SIPPSTAR IP PBX as an extension of your conventional system. To do this, you simply have to insert an ISDN card into the server on which your SIPP STAR IP PBX runs, and then connect it via the S0-bus using a conventional ISDN cable. This operating mode provides you with significantly higher user scaling. You can also convert to the new technology gradually, without doing away with your old system. Another option is to operate SIPPSTAR as an independent telephone system. If you wish to make your telephone calls via a regular ISDN connection, you need an ISDN card in the server on which SIPPSTAR is operated. In this way, the SIPPSTAR IP PBX functions as a gateway server and feeds calls via an ISDN card directly into your ISDN multiple device, system or primary multiplexer (E1) connection on the local telephone network. Your company can then also be contacted from the public telephone network at the previous number.

Pandora Network’s IP PBX Service

Pandora Network’s provides Worksmart “on demand” PBX services, uniquely targeted at small to mid-sized businesses(SMBs). For the same price as a PBXonly solution, Worksmart provides an enterprise-class IP PBX, integrated with a comprehensive set of IP communications services.Pandora’s features include:• Integration — Offers PBX integrated with VoIP, a desktop client, call routing, private IM network, public IM access, video services, web cobrowsing, group collaboration, web contact center (integrated with Salesforce.com), ACD, ACD recording, ACD management and flat rate calling;• On-Demand — Offered as a man aged service vs. forcing SMBs to buy, integrate and manage expensive hardware/software;• Low Risk — There are no changes to the SMB’s existing systems, since Pandora can remain on top of the customer’s legacy system. SMBs can simply unsubscribe to the service and return to their legacy solutions if they are unsatisfied.

Motorola Centellis 1000 Series

The Centellis 1000 Series is a MicroTCA Open Application Enabling Platform from Motorola. The Centellis 1000 series platform provides highly integrated and verified hardware and software components, reducing development costs and accelerating time-to-market. This allows telecommunications equipment manufacturers (TEMs), defense primes, and original equipment makers (OEMs) in a broad range of market segments and applications, to focus their development efforts on critical, differentiating features that provide a competitive advantage. The Centellis 1000 series is designed to the draft specification of the MicroTCA open standard, making it physically smaller, with finer-grained scalability than Motorola’s initial communications servers that are based on the AdvancedTCA industry standard. This fine-grained scalability enables MicroTCA platforms to support a pay as-you-grow business model that allows customers to realize solutions with less capital expenditure and expand the computing platform capabilities in small, low-cost increments as demand for the new service increases. This advantage is particularly relevant to some of the new point-of-access applications such as WiMAX and IP PBX. The Centellis 1000 family will be used in a wide range of applications, such as WiMAX access points, VoIP access gateways, and cellular base stations where reducing the capital cost of installing or extending next-generation network elements is very important. Small physical size, low power consumption, and enhanced serviceability also make these new communication servers ideal for a variety of applications in defense/aerospace, federal, medical, and industrial market segments.

Mitel 3300 IP Communications Platform

The Mitel 3300 IP Communications Platform (ICP) provides enterprises with a highly scalable, feature-rich communications system designed to support businesses from 10-65,000 users. The 3300 ICP provides enterprise IP PBX capability plus a range of embedded applications including standard unified messaging, auto-attendant, ACD and wireless. Operating across virtually any LAN/WAN infrastructure, the 3300 ICP provides seamless IP networking allowing for full feature transparency within distributed environments by supporting networking standards such as Q.SIG, DPNSS, and MSDN. The 3300 ICP provides organizations with the opportunity to IP-enable their legacy PBXs, protecting existing investments while delivering all the advantages of a converged infrastructure. The 3300 ICP supports the industry’s largest range of desktop devices including entry-level IP phones, Web enabled IP devices, wireless handsets, and full duplex IP audio conference units. Mitel’s Navigator integrates two of the most used business tools, the PC and the phone, to deliver real benefits to the user. The 3300 ICP also supports a powerful suite of applications including multimedia collaboration,customer relationship management and unified messaging. Industry standard Application Programming Interfaces (APIs) are supported for extensive third-party applications through Mitel Solutions Network (MiSN).

Iwatsu Enterprise 2.0 Communications Server

The Enterprise 2.0 Communications Server utilizes QuadFusion Technology to marry the four dominant communication protocols onto one platform. SIP, VoIP TDM, and H.323 can be used alone or in tandem, making the Enterprise 2.0 a truly versatile system. It converges voice and data traffic for higher cost savings, fewer hardware requirements, and more flexible bandwidth usage. Its reliable modular design allows small companies to grow up to 1024 ports with add-on features and pplications. It can integrates various applications, including transparent networking, unified communications, contact center solutions, in-building wireless and more. Web integration provides convenient browser-based system administration and reduced maintenance costs and flash-based software allows system updates from a remote maintenance console, eliminating the need to modify or replace hardware to support new software revisions. The systems also peer-to-peer communication enabling IP phones to “talk” to each other directly and rely less on system resources.

IPBX Systems

The LanPBX has all the features you would expect from a modern PBX, with a simple to use Web interface for both the users and management. Built on industry standards, it offers flexibility, ease of use, and all the features you expect. A new phone sys tem needs to take advantage of modern communications methods in order to help ncrease productivity, save money, and communications. The LanPBX offers unrivaled flexibility in the way it can be deployed, which allows for the ideal integration into your office environment, supporting multiple offices and remote workers with ease. The connection to the phone network and the physical configuration can be specified to integrate into your current IT practices. LanPBX conforms to SIP standards, ensuring that new SIP-compliant products can be added to the LanPBX through its Web-based management interface. The VoiceXML scriptable LanMS can be used to introduce new services from traditional phone-based features through to new features based on the IP architecture.

FacetPhone IP PBX telephone system

FacetPhone is an IP PBX telephone system that transforms the business phone system into software running on industry standard servers operating industry standard VoIP gateways. The system will never become obsolete, since individual com ponents can be replaced or added as needed with the latest components. FacetPhone comes with all the features you expect in a phone system, such as voice mail, auto-attendant, conferencing, callerID support, call detail recording, and call center features like automatic call distribution,call monitoring, call recording and call barge-in. In addition, with its graphical computer user interface, FacetPhone provides visual voice mail management, enterprise instant messaging, computer telephony integration with UNIX or Linux or Windows applications, presence and availability management, roaming extensions, and branch office and telecommuter support for toll bypass and true remote employee integration. The FacetPhone architecture is based on a Linux or UNIX server, external media gateways, and standard analog telephones which make the phone system inherently flexible, reliable and cost effective.

Fonality’s IP PBX Phone System

Fonality’s PBXtra product line provides small businesses with enterprise-class phone systems for 40 to 80% less than the cost of traditional PBX systems. PBXtra is a complete PBX application for small business customers who want a phone system with enterprise-class capabilities. PBXtra’s enterprise-class features include telecommuting, branch office support, voicemail-to-email, click-to-call, VoIP softphones, support for IP and analog phones, and advanced call center functionality. Fonality has streamlined and simplified the complex tasks of PBX setup, administration and management to make PBXtra the world’s first enterprise-class phone system that can be installed and administered remotely using a Web browser, without specialized training. The PBXtra software runs on standard PC hardware, Digium hardware cards,and uses layers of Open Source Linux and Asterisk software to provide the least expensive option for small businesses deploying IP PBX phone systems.

Ericsson’s convergence communication system

Ericsson’s (news - alert) multi convergence communication system is the MD110 Convergence Communication System, a system that well and truly integrates fixed and mobile telephony, IP phones, PC softphones, cordless phones, mobile/cellular phones, digital phones and IP Gateways. MD110 Convergence Communication System supports cost-effective, seamless communication across corporate voice networks, intranet, LANs, WANs, and public networks. Through convergence, enterprises will have the power to be productive by capitalizing on realtime, mission-critical business and communication applications. Now full IP Networking between MD110 and BusinessPhone enable convergence on existing IP connections, making for highly flexible work methods and better cost efficiency. Dynamic Network Administration, D.N.A., gives IT managers the choice of managing the network from a single point or from multiple points. You can manage a network of any size simply by adding D.N.A. servers and linking these together over a WAN. And you can distribute management responsibility to designated individuals anywhere within the network, each with different levels of access.

Dialexia’s IP-based PBX solution

Dialexia’s line of enterprise solutions is a suite of products that address small business communications needs. For companies that have an established Local Area Network (LAN) and want to converge data and voice over that network, Dial-Office is the ideal IP-based PBX solution. Dial-Office is an Internet alternative to the traditional circuit-switched enter prise phone system. It lets service providers deliver all of the features that companies are accustomed to in full featured PBX or Centrex systems with out the associated high costs. It handles internal calls, connects users to the worldwide phone network and sets up and manages their users and resources while offering a seamless integration of voice, data and video using a simple configuration and management tool. Dial-Office is also suitable for multi office connection, connecting branches which are geographically distant from each other. Built on Multi-domains Architecture, Dial-Office offers the flexibility of distributed location connec tions; unlike legacy PBXs, it offers employees a wide availability IP-Phone use in any location served by the company’s Wide Area Network

Colt Telecom’s IP PBX Service

Colt Telecom’s (news - alert) IP PBX service can either be hosted in Colt’s data centers or at the customer premises. It includes several hundred traditional PBX telephony features, in addition to new IP-enabled functionality such as softphone support and number portability. Additional services in development include video telephony, hosted voice recording and fixed/mobile convergence services. The advantage to customers is they can realize the benefits of an IP PBX,while not having to deal with system management. Colt’s pricing model is a flat-rate one, which includes an installation fee and a monthly charge for service. Colt’s service brings together the reliability, security, and features needed for business. The enhanced COLT IP Voice is now available as a stand-alone service or as part of the COLT Total service, which extends the power of IP telephony to a wider group of businesses, from small businesses with as few as 20 users to large organizations with as many as 1000 users. This launch marks the second phase of Colt’s “Total” converged voice and data service for SMBs. Additional rollouts will include video telephony, voice recording, and mobile services.

Cisco IP PBX review

Cisco Unified CallManager is a scalable, distributable, and highly available enterprise IP telephony call processing solution. It extends enterprise telephony features and capabilities to packet telephony network devices, such as IP phones, media processing devices, VoIP gateways, VoIP Phone Systems multimedia applications. Additional services, such as unified messaging, multimedia conferencing, collaborative contact centers, and interactive multimedia response systems are made possible through Cisco Unified CallManager open telephony APIs. Cisco Unified CallManager is installed on the Cisco Media Convergence Server 7800 Series of server platforms and selected thirdparty servers. Cisco Unified CallManager Version 5.0 enhances application delivery through the support of line side SIP and SIP trunk-side enhancements. These enhancements facilitate increased interoperability with third-party applications and devices, and provide the foundation for supporting innovative new presence-based applications. In addition, Cisco Unified CallManager 5.0 simplifies deployment and management by supporting a Linux-based appliance model implementation. This version also features networking and administration enhancements, including support for Cisco RSVP Agent which enables more efficient use of networks.

Citel IP PBX review

Rather than supporting multiple PBX systems and remote connections, Citel EXTender products allow enterprises to connect remote call centers, home workers, and branch offices to a central digital PBX over an IP network, significantly reducing telecom operating costs and simultaneously improving business operations by providing single voice mail and call center applications, central reception, and four-digit dialing throughout the enterprise. As enterprise customers become ready to complete the migration to SIP, the EXTender IP6000 can be software upgraded to accommodate an on premise or service provider hosted IP PBX, leveraging the existing handset and wiring infrastructure at each location. This phased migration path allows enterprises to immediately realize the advantages of a central PBX platform, then complete the full migration to SIP telephony in the future. The EXTender IP6000 expands the EXTender product line with a new lower price point and an assured upgrade path to SIP-based hosted or premise IP telephony in the future. The EXTender IP6000 is available in a 12-Port configuration, which can be scaled to accommodate the number of stations at each remote site. The EXTender IP6000 is compatible with many leading PBX platforms, including Avaya/Lucent Definity and Magix, Nortel Meridian and Norstar, Alcatel, Ericsson, Iwatsu, Toshiba, and Panasonic

Avaya IP PBX Review

Avaya IP Office is an all-in-one solution specially designed to meet the communications challenges facing SMBs. Due to its modular design, the solution can scale from 2 to 360 extensions to meet the needs of home offices, stand-alone businesses, and networked branch and head offices. IP Office supports a wide range of telephones, but the 5400 series Digital phones and the 5600 series IP phones have been specifically designed to work with IP Office and provide SMBs with a choice of solutions to meet business efficiency and customer service requirements. IP Office is a highly integrated voice and data communication solution that aims to deliver within a single product solution the complete communication requirements of the SME customer — including many of the benefits enjoyed by the larger enterprises. Avaya IP Office can easily scale up to 360 endpoints with more than 200 analog and digital trunks (up to 96 trunks; 192 analog trunks), giving small and medium size businesses room to grow. Multiple Avaya IP Office systems can be linked together using a standard data network, providing limited rich transparency and advanced applications, such as centralized voice mail and call center. Avaya IP Office allows users access from desktop computers, laptop PCs, with individual firewall security and access control.

Brekeke IP PBX review

OnDO PBX is a software-based telephony system. It is SIP-compliant and Web-based for easy installation and management. The PBX includes all the features of traditional PBX systems, such as call transfer, call conference, call forwarding, voicemail,and much more. Our OnDO PBX, SmallOffice and Standard Editions, which have multiplatform support, provide fully functional telephone systems often referred to as PBX systems. They are easily managed through a web-based administrative tool, and scalable. Both editions feature voicemail, call forwarding, call conference, call monitoring, call recording, and much more. You can install and use a fully functional IP-PBX, with OnDO PBX, quickly and easily. OnDO PBX Version 1.5, Standard Edition, now supports Automatic Route Selection (ARS) failover. With ARS failover, OnDO PBX seeks an alternate route if the specified route is unavailable, and makes outgoing calls on the best route depending on the situation.

SessionSuite IP PBX Review

SessionSuite IP Telephony, modular standards-based IP communications software, delivers voice/video as distributed communication services that can be easily incorporated into enterprises’ business process applications. SessionSuite IP Telephony offers real-time voice/video services as standalone IP applications that can be accessed over any network by any application or user. It leverages enterprises’ existing data center infrastructure, integrating voice with the same back-office directory, authentication, authorization and accounting services employed for data applications. Optimizing the Internet, it economically extends the reach of services outside of enterprises’ traditional “four walls.” By combining industry Standard SIP with Web Services, enterprises can easily integrate voice/video services into their business processes and applications within a Service Oriented Architecture (SOA). Enterprises benefit from SessionSuite IP Telephony’s ability to provide a forward-looking solution where voice/video can be tightly integrated into business processes, while leveraging existing infrastructure investment to offer high-value, applications-based communication services to users, customers, partners and employees.

Asterisk IP PBX review

Asterisk is a complete PBX in software that runs on Linux, BSD, and MacOSX, providing all of the features you would expect from a PBX — and more. Asterisk does VoIP in any protocols and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware Asterisk needs no additional hardware for VoIP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk’s sponsors, Digium. Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next generation voice/data integrated networks and existing infrastructure. Asterisk provides a central switching core, with four APIs for modular loading of telephony applications,hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.

Aztech IP PBX Review

With Aztech IP PBX system IPX1050, calling an overseas office within the same organization is virtually free as it uses SIP to establish connections among users. A web browser-based tool will configure and assign telephone extension number to employees within an organization.Aztech IPX1050 is PoE enabled and does not equire any power adaptor, as the Ethernet cable will provide the necessary power The IPX1050 will induce significant cost savings by allowing customers to manage a single system for voice and data and the sharing of a single broad band connection. It scalable to 50 extensions and 8 PSTN lines, so customers can start with a system based on their current needs and then, as they grow, scale accordingly. It supports 50 voice mailboxes and 100 hours of recorded messages. The system is also softphone and WiFi compatible, leaving room for today’s convergent technology and comes with a convenient, easy to use Web-based installation tool. It also includes PoE support.

Alcatel IP PBX Reveiw

The Alcatel OmniPCX Enterprise is an integrated, interactive communications solution for medium sized businesses and large corporations. It combines the best of the old (legacy TDM phone connectivity) with the new (a native IP platform and sup port for Session Initiation Protocol, or SIP) to provide an effective and com plete communications solution for cost-conscious companies on the cutting edge. Alcatel’s solution is designed to improve productivity and enhance customer care, while reducing capital expenses and operations costs. It provides a high-availability platform (under UNIX and Linux), which delivers powerful communication tools, including a customer care center, business applications and tools that simplify daily administrative tasks.The Alcatel OmniPCX Enterprise also features an embedded mobility solution, Web-enabled soft phone, featurerich networking over all types of media, and resiliency mechanisms. The Alcatel OmniPCX Enterprise is built around six value propositions: • Architectural flexibility • Intelligent networking • Highest reliability • Simplified management • Agile workplace • Superior customer interaction

Allworx IP PBX Reveiw

Designed for companies up to 100 users, the Allworx 10x system is a state-of-the art communications system that integrates three essential business operations into one simple system. It is a feature-rich phone system, a robust data network system, and a message center that substantially improves productivity. The Allworx 10x provides enterpriseclass features, such as custom call routing, presence management, call center control, remote user access and site-to-site connectivity. It is a phone system that integrates VoIP with a fully featured PBX and Key system, providing amazing flexibility and power to the small business. The system also supports remote users and you can connect multiple Allworx systems to create one phone system across multiple sites.The Allworx 10x is flexible, scalable, and offers LAN and WAN access. It easily integrates into your existing net work, or it can function as your complete network server. You can even create a VPN that provides secure Internet access for remote employees or directly connects two office locations. The Allworx 6x delivers the same level of enterprise-class service and functionality for companies with up to 30 users.

AltiGen IP PBX Review

Out of the box, AltiGen provides a high level of capability for the money. Every AltiServ system is prepackaged with software and licensing for the operator, Extensions, and AltiView’s call control software. The system includes all standard PBX functionality, a comprehensive voice mail system, call detail reporting and an advanced auto attendant. The system can support all Voice over IP, analog, or a combination of both. Thesystem is easily scaled in both size and capability. This inherent flexibility will allow your company to easily adapt the phone system to the way you would like to do business. Based on your current needs and future plans, there are a selection of AltiServ base systems to fit every business. AltiServ phone systems are “All in One” products that can be easily administered on site or remotely.

Anta Systems IP PBX Review

Anta Systems has developed a turnkey VoIP solution that reduces the cost and complexity of VoIP deployment, enabling service providers and SMEs to reap the bene fits of VoIP without unnecessarily large capital investments. Anta’s Simplicity VoIP System is an end-to-end SIP based VoIP solution. The Simplicity SE-160 is a plug and play IP PBX designed for small businesses that has all of the traditional PBX features as well as a suite of enhanced applications such as auto attendant, visual voicemail, 9-way conferencing and remote access capability. It is designed to be deployed in multi-site SMEs or as a Managed IP PBX solution by service providers. The Simplicity ME-1000 brings to SMEs integrated functionalities of a media gateway, Softswitch, media server, application server, SBC. It is designed to enable VoIP service providers to quickly and cost-effectively launch basic residential and business VoIP services, as well as enhanced services such as IP Centrex,Visual Voicemail and VoIP Conferencing. The ME-2000 for SMEs integrates the functionalities of a media gateway,Softswitch, media server, application server, and SBC. It is designed to enable VoIP service providers to quickly and cost-effectively launch basic residential and business VoIP services as well as enhanced services such as IP Centrex, Visual Voicemail and VoIP Conferencing

Configuring PoivY on X-lite

This post is to give information about how to configure PoivY on X-Lite softphone. In X-Lite open ‘menu’ and select ’system settings’ then ‘sip proxy`, from the list of proxies deplayed select the an available proxy. You should then enter the settings below: Enabled YES Display Name (PoivY user name) Username (PoivY user name) Authorization User (PoivY user name) Password (PoivY password) Domain/Realm sip.poivy.com SIP Proxy connectionserver.poivy.com Out Bound Proxy sip.poivy.com Leave all other settings as default. This configuration is working fine for me and it also rings on incoming call.

Configuring PoivY on Express Talk

To configure PoivY on Express Talk softphone follow the steps below: 1. From the menu bar click VIEW and select SETTINGS 2. Select the "Line Tab" in settings 3. Select on which line you want to configure PoivY, there are lines 1 to 4 4. Enter the following cofiguration settings:

• Display Name: Your Name
• Sip Account Number(or user): your PoivY user-id
• Server(sip Proxy or Virtual PBX): connectionserver.poivy.com
• Password: Your poivy password.

5. Click OK These settings worked very fine for me. Feel free to comment or make suggestions regarding these settings. Related post : PoivY Express Talk - Free softphone with a difference

PoivY - The latest from Finarea with a difference

PoivY is the latest voip application from Finarea the company behind voipbuster, voipcheap, voipstunt etc. PoivY came with a major difference, no free destinations like in the other applications. This seems to be a major change in the Finarea bussines model which used to entice subscribers with unlimited free calls to lots of destinations. Even though there no free destinations there is two ways to make free calls using poivY. They offer super deal rate of 1.5 cents per min to about 24 destinations with include countries from Europe, North America and Asia. Also once you sign up you get 1 euro credit to call the super deal countries and you also get 66 free minutes to call super deal destinations for every person you refer to poivY. PoivY have all the regular features of all voip clients from finarea which include voip-in number, call waiting and call forwarding. PoivY is very easy to setup and use with no configuration required if you are using their client. The audio quality is really good and i hope that they will now improve their customer service now that there is no free destinations. Poivy can be downloaded from www.poivy.com.

VoIP Logic to use Highdeals Transactive Pricing and Rating Solution

VoIP Logic has selected Highdeal’s Transactive pricing and rating solution to improve the delivery and management of its next generation IP services. Highdeal’s Transactive modular software suite can price and rate thousands of transactions per second, giving carriers the functionality needed for fast and cost-effective delivery of today’s emerging services. “By integrating Highdeal’s billing and rating engine into our on demand delivery platform, VoIP Logic enables service providers to deploy a carrier-grade billing solution quickly and with few in-house resources,” said Kevin Burke, COO and CMO of VoIP Logic. Highdeal’s pricing and rating solutions solve the billing challenges brought about by the emergence of next generation services. By delivering unconstrained pricing and packaging flexibility, Highdeal enables the rapid implementation of convergent services with diverse payments models and multiple partners.

Earthlink, Linksys Team to Provide VoIP Solution

Internet service provider (ISP) EarthLink and Linksys, a division of Cisco Systems, have announced a co-marketed Voice over Internet Protocol (VoIP) hardware and service solution that provides everything customers need to make phone calls over their Internet connection. The new co-marketed VoIP solution features Earthlink’s trueVoice telephony service along with the customer’s choice of a phone adapter (SPA2002-ER) or wireless-G router (WRTP54G-ER) from Linksys. EarthLink’s trueVoice is compatible with any high speed connection and is a plug and play solution that can be installed in minutes to work on any touchtone phone. With the service, customers can achieve the cost savings and feature benefits of calling via the Internet, without having to invest in expensive new handsets. The Linksys phone adapter works with a standard wired or wireless router, while the wireless-G has a built-in phone adapter for home networking. Both products provide two phone ports for connection of two standard home phones or fax machines.

Nero Partners with Digital Rapids

Nero, a provider digital media technology, announced a partnership with Digital Rapids, (news - alert) a developer of professional hardware and software solutions for post production, broadcast, IPTV, Video on Demand (VOD) and other advanced media applications. The first collaboration between the two companies is the integration of Nero’s AVC/H.264 and High Efficiency AAC, which are core components of Nero’s ISO-standard Nero Digital(TM) technologies. The Nero Digital technology family includes MPEG-4 and AAC, AVC/H.264, and High Efficiency AAC for the ultimate viewing experience from mobile phones to High Definition video screens. Moving forward, Nero and Digital Rapids will continue working on a number of market-facing solutions, combining Nero’s technologies with Digital Rapids’ software and hardware solutions for professional video ingest, playout, encoding, transcoding, and delivery.

Primus and TalkSwitch Establish Interoperability in the U.S.

VoIP service provider Primus Telecommunications (news - alert) and voice systems designer TalkSwitch (news - alert) joined forces to announce full interoperability between Primus’ calling services and TalkSwitch’s 48-CVA PBX. The ability to bundle Primus’ VoIP calling service and TalkSwitch’s 48-CVA PBX will offer customers VoIP features, follow-me functionality, and conferencing, while maintaining a connection to the traditional telephone network. Joint deployments of Primus and the TalkSwitch PBX will provide call processing at the customer premises, while the user is free to select from a set of voice, data, and access options provided by Primus. Each TalkSwitch telephone system comes loaded with a host of features, including multi-level auto attendants, call cascade options, voicemail and will feature Primus’ VoIP calling plans.

Jacada Leads the Way to Next-Gen

Jacada Ltd., which develops contact center productivity solutions, has released version 3.0 of its unified desktop solution, Jacada WorkSpace, in a model created to meet the demands of the growing number of contact centers seeking a modernized agent desktop, while optimizing processes and maximizing CSR productivity. Jacada WorkSpace 3.0, which became popular under it former brand, Jacada Fusion Agent Portal, now represents the next generation of contact center desktops. It is a single agent console that unifies customer interaction tools with a single access point to all the mission-critical applications that enable the agent to effectively service customers. Enhancements in version 3.0 include universal agent capabilities, support for multiple, simultaneous call sessions, support for Linux servers, and Asynchronous JavaScript Technology and XML (AJAX) controls and features found in the new Web 2.0 rich client foundation. This release is only the beginning of greater changes within the industry, according to the firm. “Market experts and industry visionaries continue to place significant importance on the adoption of a unified CSR desktop,” said Jacada CEO Gideon Hollander. “But while desktop unification is a top priority in many contact centers, what this next generation desktop should actually be has, until now, remained undefined.”

Aspect to support for Digium open source IP PBX

Aspect Software (news - alert) announced it will provide and support the Digium open source IP PBX, the Asterisk Business Edition — a professional-grade version of the industry’s first open source IP PBX — for customers of its Unified The increased adoption of SIP and standards-based technology points to open source as an increasingly viable option. The early adopters of this technology have been drawn by the low cost, as well as the greater control and flexibility that The Asterisk Business Edition IP PBX provides tested reliability of critical functions and features and includes support and full documentation. Based on the Asterisk open source PBX, the product offers companies the same call handling capabilities expected of closed PBX systems, at a substantially reduced cost, including features such as switched or packet “Industry experts have acknowledged that the biggest obstacle to widespread adoption of open source applications has been installation and ongoing support,” said Gary Barnett, chief technology officer and executive vice president of technical services at Aspect Software. “Now, there is nothing to stand in the way of companies being able to leverage the benefits that open source provides, including inexpensive voice

Salesnet’s On-Demand CRM Solution Takes Flight

During its six-year existence, JetBlue Airways has endeavored to be more than the typical airline — but for less. But as more and more airlines find it difficult to keep profits soaring, JetBlue also had to formulate a flight plan to keep its customers satisfied without cutting into margins or eliminating amenities. It began by selecting Salesnet’s (news - alert) patent pending on-demand CRM software to increase the effectiveness of its sales force. Salesnet’s is a cost-effective alternative to expensive, complex, packaged, or premise-based CRM software. Salesnet’s Guided Performance Selling (GPS) strategy essentially turns software, configuration, integration, and administration into service offerings designed to drive increased ROI — higher than purchased in-house solutions, anyway. The solution increases performance by defining best practices, guiding salespeople to use those best practices, and tracking ongoing success using those best practices. It also assists in defining the path between business goals and realistic solutions. On demand software is nothing new, but the ease of administration, adaptability, and use and the single, low monthly subscription cost make Salesnet’s an intriguing proposition. Ultimately, Salesnet was the most intuitive solution, including the sophisticated Salesnet Dashboard, which was able to display information significantly more clearly than other vendors’ solutions.

FrontRange Launches Service Management Software Upgrade

In an effort to support to its growing customer base, FrontRange Solutions (news - alert) introduced the latest installment to its IT Service Management (ITSM), the IT Service Management 5.0.4 Service Pack 1. The software is designed to assist IT service managers and support staff with new features, such as Inventory Management enhancements, including CI Comparison Utility, Inventory Identity, Discovery Enhancements and Scheduled Jobs. ITSM 5.0.4 Service Pack 1 can be installed on top of ITSM 5.0.4. “These innovations increase ITSM’s enterprise class functionality,” said Lori Samolyk, FrontRange Senior Product Marketing Manager in a statement. “The FrontRange ITSM suite is helping small and large enterprises manage their IT systems and processes in accordance with ITIL. The ground-breaking ITSM suite is a mirror image of the ITIL best practices, which are quickly becoming accepted as the business model for IT.” FrontRange’s products are designed specifically for small-to-medium-enterprise (SME) and distributed enterprise organizations.

CRM Solution Helps Target Interact Increase Customer Satisfaction

Management consulting firm Target Interact recently chose UCN Inc.’s solution for enhancing the reliability and efficiency of its call center opera tions. UCN provides on-demand contact handling software and telecommunication services over its national VoIP network. InContact is a hosted solution set designed to significantly enhance caller satisfaction, boost agent productivity, and improve overall profitability. Features include interactive voice response (IVR), skills-based ACD routing, computer/telephony integration (CTI), inbound/ outbound call blending, remote agent or multi-site support, and much more. InControl, part of the inContact suite, is an application development tool that simplifies customizing solutions to a drag and drop level. The ability to quickly create a call script allows contact management models to be tailored to each client’s individual needs. InContact identifies and routes a repeat customer to an appropriate agent using Caller ID, where caller information pop-up screens appear simultaneously with the call for the agent, whether they are working at home or in the office.

VoSKYTM Exchange - The Skype Solution for Businesses

Actiontec's VoSKY^TM Exchange enables small and medium size bussinesses to take advantage of skype's low cost rates without making any changes to their existing system. With VoSKY Exchange bussinesses can add up to four outgoing skype lines to their existing PBX without making any changes whatsoever to the PBX. This system is intended for companies with 10 to 300 employees. VoSKY Exchange plugs into the PBX on one end and to a windows xp computer on the other end. With this every phone on the exchange can make and receive skype calls. Outbound skype calls are made by dialing a single digit code that is configured for when VoSKY. Existing phones in the enterprise are used for the calls therefore no need for Headsets. VoSKY Exchange all supports skype group so all four skype lines share thesame account making administration easy. It also supports skye speed dial. The system has a rapid return on investment through savings on calls made with skype low rates. The only major requirement to use VoSKY Enterprise is a PBX with up to four free FXO ports and auto attendant.

PcPhoneline VTA 1000 – The dual mode Skype/SIP Gateway

If you are looking for an adapter that works with skype and SIP based voip services then the PcPhoneline VTA 1000 dual mode gateway is what you need. It connects to your computer through a USB port and have a phone port for your regular phone. You can make skype/skypeout and sip calls directly from your regular phone. I will be placing an order for one soon so that I can test it to see how best it works with sip services such as voipbuster, voipcheap, skypho etc that I use. I will write a more detailed review of it after the test. Meanwhile, I have read some very positive reviews about this product on the internet.George Pickles of voipuser .org wrote a nice review about PcPhoneline VTA 1000.

SPA-3000 Analog Telephone Adapter from Sipura

This highly efficient telephone adapter combines the functionality found in the SPA-2000 and SPA-1000/1 and have the additional advantage of possible connection to legacy telephone network. The SPA-3000 enable users to make the best use of their broadband phone service. Calls from mobile or regular phone can be routed through a voip service provider and vice versa eliminating long distance call rates. Users can call the SPA-3000 through a local number or a phone directly attached to it and the call will then be routed through a voip service provider to their destination. The SPA-3000 have advanced authentication, call routing intelligenece and security feature. With this device you use several voip service providers and always let it route calls through the best provider for your intended destination. You can also setup your own voip network using two SPA-3000 connected to both the internet and regular phone network. One SPA-3000 route your long distance or international call through the internet to the second SPA-3000 where the call can be answered at no cost or futher processed as a local call to any regular or mobile phone allowed by the SPA-3000 dial plan. In case of power failure or the Voip service is dawn calls are sent to a traditional phone carrier via the FXO interface. The SPA-3000 is in compliance with Part 68 United States FCC Regulations, ETSI TS 103 021 – 1, 2, 3 and Australian ACIF S002: 2001.

Gizmo project announce new Call In numbers in Spain, France and UK

Gizmo project has announced that they now have new call in numbers in Spain, France and UK. In Spain gizmo project users can buy local madrid numbers while Paris numbers are available in France. Numbers from inner London, Bristol, Manchester and Glasgow. Gizmo project Call In service let you get a local number of your choice on which people can call you directly on your PC from any telephone. Gizmo Call In service cost $3 per month. Related post : Gizmo project getting better

EQO - Mobile Skype Plugin

EQO is a software device that connects your mobile phone to your skype account so that you can make and receive skype calls anywhere you are. EQO allows skype users to make and receive skype calls directly from there mobile phone without the need for expensive 3G or Wi-Fi enabled phones and wireless broadband internet connection. Its very easy to setup and use, you download EQO from their website, install it on your system on which skype is running and enter some information about your mobile phone. The mobile phone application will then be configured and sent to your mobile phone directly and you can then install it with a click. Once EQO and skype are running on your system and EQO on your mobile phone, you can receive and make skype calls directly from your mobile phone. EQO works by using GPRS for the signalling and routing the calls to your mobile phone using skypeOut. Not all mobile phones are supported at the moment but a list of supported phones can be found on their website at www.eqo.com.

Vbuzzer Internet phone review

I have been testing Vbuzzer internet phone for some time now and its been working really fine. The PC-to-Phone rates are low and the audio quality is very good. Vbuzzer is the product of Canada based Softroute Corporation. Its a combined voip application and instant messenger. Important features include, call forwarding to any landline or mobile phones, free voicemail to e-mail service and recording of personal voicemail greeting. The PC-to-phone service is nicknamed BuzzOut while Phone-to-PC service nicknamed BuzzMe enable subscribers to receive unlimited calls on Vbuzzer. You get a free area 416 BuzzMe number that can be reached from any phone. They have at the moment an introductory offer which include the following:

1. Free Area 416 BuzzMe number
2. BuzzMe number with area code of your choice
3. BuzzOut Toronto - Unlimited free calls to any Toronto number
4. BuzzOut Canada - Unlimited calls across Canada for just $5.49 per month
5. BuzzOut Canada and US - Unlimited calls across Canada and US for just $7.99 per month

This offers are valid till june 30, 2006. Vbuzzer is SIP based and can be configured on phone adapters or ip phones. Click Here for more information and download

More Skype Dating sites !

More dating sites based on the skype voip application has emerged on the internet. Some days ago i wrote about verbdate and yesterday i discovered someonenew and single skypers. All this dating sites enable their members to meet other skype users. You register and create a profile with your skype name and if somebody likes your profile he or she can call you directly on skype. The days of sending e-mail to breate the ice is fast becoming outdated now you can just skype instead. So VoIP is not only changing the way we communicate but also the way we meet and date. So if you are single and looking and use skype you, can register with any of these service, you never know maybe the next time your skype will ring it might be " I just call to say I love you". Good luck Singles!.

Yahoo and MSN to interoperate

According to report on the Yahoo messenger website, Yahoo and MSN are working on a way to connect Yahoo messenger and MSN messenger. This they said will happen within the first half of 2006. This has been long awaited and i strongly believe interoperation is the right way to go. Most analysts believe that Yahoo and MSN are connecting in order to compete with Skype in the voip business. With interoperation users of Yahoo will be able to see friends who are online on MSN messenger and be able to share IMs and emotions, easily add new contacts from either application. It is also expected that users will be able to make voip calls from Yahoo to MSN and vice versa. More Yahoo Links : ============== Yahoo! Personals with 7 Day FREE Trial offer Get quotes from Yahoo! Autos Yahoo!Music Unlimited Free Trial Yahoo! Top Games

New Internet Phone Service launched in Vietnam

Vietnamese One Connection Internet Inc. (OCI) has launched a new broadband internet phone service in Vietnam, the first of its kind in this Asian Country. With this service subscribers can save up to 80 per cent on international calls. The service nicknamed B-phone has a monthly subscription charge of VND27,000 ($1.7). The user gets a broadband internet connection and a PAP2 Telephone Adapter. This is another example of how VoIP technology is helping developing countries to reduce the cost of international calls.The OCI Company, founded in 2001 is a subsidiary of the Ho Chi Minh City-based Electronic Information System Incorporation (EIS Inc.). OCI, known as a leading internet service provider in Vietnam, provides high-quality Internet and other online add-in services at cost-saving price and offers best information and communication technology for business community.

Jajah to integrate with Opera browser and Symbian OS

Jajah Inc. has signed an exclusive agreements to intergrate JAJAH within Opera browsers and the Symbian OS mobile operating system. Through these partnerships, JAJAH instantly reaches Opera’s 20 million users and Symbian’s 60 million users. JAJAH will also be integrated in a custom version of Opera Mini, the only full web browser for the world's 700 million Java-enabled mobile phones. Opera Software ASA is an industry leader in the development of web browser technology for the desktop and mobile Internet devices; Symbian Ltd. licenses Symbian OS, the market leading open operating system for advanced, data-enabled mobile phones

Jajah going Mobile

JaJah Inc. has announced that the popular Jajah web activated telephony will soon be available on mobile phones. This might happen as early as May 2006. Jajah has become so popular because of its ease of use and it has seen exponential growth in just a week since launching in the U.S.. No download, no complitated setup or hardware required and you dont even need broadband internet connection. You just use your web-browser to activate the call and make the call using your regular telephone. When launched jajah mobile will enable subscribers to initiate cheap voip calls directly from their mobile phones and this will considerably reduce the cost of calling from mobile phones which at the moment is still very high in most countries. Click Here for more information about Jajah web-activated telephony.

Sipera Launches VoIP Security System

Richardson, Texas based Sipera Inc. has launched Sipera IPCS 310 System for comprehensive IP Communications Security. The Sipera IPCS 310 intelligently and transparently monitor VoIP traffic, detect anomalies in traffic and call patterns and identify threats, to protect end-user devices and network infrastructures against attacks. Enterprises are migrating their mission critical communications such as voice telephony to IP networks. This undoubtely has enormous advantages which include cost savings and more collaboration but this also expose them to potential treat of people who can explore the openess that comes with IP networks to launch malicious attacks. Which is why security products such as Sipera IPCS 310 are must have for an enterprise. The Sipera IPCS 310 utilizes sophisticated security functionality with patented protection techniques to intelligently analyze every call, and perform the correct action. Sipera IPCS products can be deployed in any existing VoIP infrastructure with no need for on-site interoperability testing, due to Sipera's close development and integration work with leading VoIP infrastructure manufacturers. More information about Sipera IPCS products can be found on http://www.sipera.com

Verbdate launches "Dating 2.0" service, a SkypeWeb enhanced dating service

Just as it has now become the trend for social networking websites to integrate voip applications most especially skype to enhance communication between their members, Canada based Verbdate have decided to go in this direction too. They have launched a online dating service taged Dating 2.0 which integrates SkypeWeb, Skype's integrated Internet presence feature, and location maps to help singles find and connect with each other in their own geographic locations, helping likeminded individuals speak anonymously with one another safely before meeting. The Verbdate service, which is 100% free, will serve the 80 million singles presently seeking a date online in North America today with a newly enhanced search and communication system. This new service allows members to see who is presently online and willing to accept a Skype call, an instant message, or an email to break-the-ice. Singles can search for compatible like-minded individuals with “tags”, keywords that describe an individual's likes, hobbies, and personality traits, resulting in a much more real world matching experience. SkypeWeb makes it easy for Web users to connect through Skype. Available in 27 languages, Skype is used by people in almost every country around the world, and communication over Skype has become a global phenomenon. The latest version of the software has further simplified the Skype interface, making it even easier for people to connect online. Additionally, members can add others to their buddies list and see where their friends and acquaintances last location was, and with Verbdate's soon to be released mobile client this buddy system will surely be a great tool for seeing the latest singles hotspot on a Friday night!. More information about subscription is available at www.verbdate.com.

TerraSip - The Mobile VoIP Community

TerraSip is a recent, globally aligned and world-wide operating enterprise. It offers free internet telephony based on the international SIP standard for TerraSip members. The membership is free. TerraSip's adjustment aims at mobile internet telephony, the voice over WLAN, briefly VoWLAN. Recently TerraSip started a first pilot project of VoWLAN with two WiFi network carriers in Germany and Austria, in order to test mobile internet telephony in the German-speaking countries. More details soon!

UTStarcom F3000 - Clamshell Type Wi-Fi Handset

This beautifully designed Wi-Fi enabled mobile handset was first introduced by the German VoIP provider Sipgate at cebit 2006 in Hanover. It has a clamshell type design and enable making voip calls wherever their is Wi-Fi broadband internet access. It has all the popular call features like three way calling/call waiting, call rejection, call transfer/call forwarding, caller ID blocking, repeat dial and call hold/resume.F3000 supports major VoIP/IP protocols such SIP, HTTP, RTP/RTCP, SDP, STUN and DHPC and also support G.711u, G.711a, G.729a, G.726 speech codecs. It has a nice 65K CSTN Color display with a resolution of 128x160. It supports the 802.11b/g with WEP and WAP encryption standards. Powered by a Li-ion battery it gives about 3 hours talking time and about 75 hours standby time. With a dimension of only 85x43x22 mm and weihgt of just 90g it fits nicely in your pocket like any mobile phone.

Configuring Voipdiscount on Express Talk

To configure Voipdiscount on Express Talk softphone follow the steps below: 1. From the menu bar click VIEW and select SETTINGS 2. Select the "Line Tab" in settings 3. Select on which line you want to configure voipbuster, there are lines 1 to 4 4. Enter the following cofiguration settings:

• Display Name: Your Name
• Sip Account Number(or user): your voipdiscount user-id
• Server(sip Proxy or Virtual PBX): sip.voipdiscount.com
• Password: Your voipdiscount password.

5. Click OK These settings worked very fine for me. Feel free to comment or make suggestions regarding these settings. Related post : Express Talk - Free softphone with a difference

Best voip calling rates for Ghana. Cheaper than calling cards

If you are looking for the cheapest calling rates to Ghana and with guarantied good audio quality, then here is what you are looking for. The table below gives the best calling rates for landlines and mobile phones in Ghana for several voip applications and service providers that i have used. Included in the table are the cheapest calling rates for Ghana and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio quality. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.153           0.166  

  skypho                   0.082           0.178

  voipcheap                -N.A-           -N.A-

  voipstunt                -N.A-           -N.A-

  Gizmoproject             -N.A-           -N.A-

  Internetcalls            -N.A-           -N.A-

  voipdiscount             -N.A-           -N.A-

  Jajah                    0.079           0.160

  poivy                    -N.A-           -N.A-

Best voip calling rates for Aruba. Cheaper than calling cards

If you are looking for the cheapest calling rates to Aruba and with guarantied good audio quality, then here is what you are looking for. The table below gives the best calling rates for landlines and mobile phones in Aruba for several voip applications and service providers that i have used. Included in the table are the cheapest calling rates for Aruba and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio quality. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.164           0.239  

  skypho                   0.113           0.183

  voipcheap                0.160           0.160

  voipstunt                0.144           0.144

  Gizmoproject             -N.A-           -N.A-

  Internetcalls            0.096           0.144

  voipdiscount             0.096           0.144

  Jajah                    0.098           0.185

  poivy                    0.120           0.204

Best voip calling rates for Philippines. Cheaper than Calling Card rates

If you are looking for the cheapest calling rates to Philippines and with guarantied good audio quality, then here is what you are looking for. The table below gives the best calling rates for landlines and mobile phones in Philippines for several voip applications and service providers that i have used. Included in the table are the cheapest calling rates for Philippines and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio quality. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.198           0.249  

  skypho                   0.147           0.170

  voipcheap                0.231           0.249

  voipstunt                0.204           0.228

  Gizmoproject             0.327           0.327

  Internetcalls            0.132           0.168

  voipdiscount             0.132           0.168

  Jajah                    0.137           0.160

  poivy                    0.168           0.168

Best voip calling rates for Bangladesh. Cheaper than Calling Cards

If you are looking for the cheapest calling rates to Bangladesh and with guarantied good audio quality, then here is what you are looking for. The table below gives the best calling rates for landlines and mobile phones in Bangladesh for several voip applications and service providers that i have used. Included in the table are the cheapest calling rates for Bangladesh and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio quality. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.136           0.136  

  skypho                   0.070           0.057

  voipcheap                -N.A-           -N.A-

  voipstunt                -N.A-           -N.A-

  Gizmoproject             0.304           0.271

  Internetcalls            -N.A-           -N.A-

  voipdiscount             -N.A-           -N.A-

  Jajah                    0.046           0.071

  poivy                    -N.A-           -N.A-

Best voip calling rates for Argentina. Cheaper than calling cards

If you are looking for the cheapest calling rates to Argentina and with guarantied good audio quality, then here is what you are looking for. The table below gives the best calling rates for landlines and mobile phones in Argentina for several voip applications and service providers that i have used. Included in the table are the cheapest calling rates for Argentina and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio quality. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.032           0.172  

  skypho                   0.025           0.163

  voipcheap                0.009           0.174

  voipstunt                free            0.156

  Gizmoproject             0.049           0.273

  Internetcalls            free            0.156

  voipdiscount             free            0.156

  Jajah                    0.016           0.187

  poivy                    0.018           0.192

Best voip calling rates for India. Cheaper than calling cards

If you are looking for the cheapest calling rates to India and with guarantied good audio quality, then here is what you are looking for. The table below gives the best calling rates for landlines and mobile phones in India for several voip applications and service providers that i have used. Included in the table are the cheapest calling rates for India and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio quality. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.154           0.170  

  skypho                   0.138           0.138

  voipcheap                0.160           0.160

  voipstunt                0.144           0.144

  Gizmoproject             0.263           0.263

  Internetcalls            0.120           0.120

  voipdiscount             0.120           0.120

  Jajah                    0.118           0.133

  poivy                    0.132           0.132

Best voip calling rates for Cuba. Cheaper than Calling Cards

If you are looking for the cheapest calling rates to Cuba and with guarantied good audio quality, then here is what you are looking for. The table below gives best calling rates for landlines and mobile phones in Cuba for several voip application and service providers that i have used. Included in the table are the cheapest calling rates for Cuba and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio qualities. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    1.025           1.025  

  skypho                   1.008           1.008

  voipcheap                1.001           1.001

  voipstunt                0.90            0.90

  Gizmoproject             N.A             N.A

  Internetcalls            0.90            0.90

  voipdiscount             0.90            0.90

  Jajah                    0.9836          0.9836

  poivy                    0.9720          0.9720

Best voip calling rates for Pakistan. Cheaper than calling cards

If you are looking for the cheapest calling rates to Pakistan and with guarantied good audio quality, then here is what you are looking for. The table below gives best calling rates for landlines and mobile phones in Pakistan for several voip application and service providers that i have used. Included in the table are the cheapest calling rates for Pakistan and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio qualities. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.143           0.164  

  skypho                   0.115           0.115

  voipcheap                -N.A-           -N.A-

  voipstunt                -N.A-           -N.A-

  Gizmoproject             0.408           0.364

  Internetcalls            -N.A-           -N.A-

  voipdiscount             -N.A-           -N.A-

  Jajah                    0.096           0.116

  poivy                    -N.A-           -N.A-

Best voip calling rates for Iran. Cheaper than Calling cards

If you are looking for the cheapest calling rates to Iran and with guarantied good audio quality, then here is what you are looking for. The table below gives best calling rates for landlines and mobile phones in Iran for several voip application and service providers that i have used. Included in the table are the cheapest calling rates for Iran and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio qualities. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.133           0.168  

  skypho                   0.101           0.147

  voipcheap                0.134           0.174

  voipstunt                0.120           0.156

  Gizmoproject             0.220           0.220

  Internetcalls            0.072           0.156

  voipdiscount             0.072           0.156

  Jajah                    0.046           0.181

  poivy                    0.084           0.180

Best voip calling rates for Iraq. Cheaper than Calling cards

If you are looking for the cheapest calling rates to Iraq and with guarantied good audio quality, then here is what you are looking for. The table below gives best calling rates for landlines and mobile phones in Iraq for several voip application and service providers that i have used. Included in the table are the cheapest calling rates for Iraq and with voip applications that gives very good audio quality. I have used all the internet phone applications included in this table and they all have remarkably good audio qualities. Some of the rates are even cheaper than calling card rates for thesame destination.

 Application/            landline          mobile
  provider                $/min            $/min

  skype                    0.372           0.372  

  skypho                   0.175           0.175

  voipcheap                0.120           0.214

  voipstunt                0.024           0.192

  Gizmoproject             1.035           1.035

  Internetcalls            0.024           0.192

  voipdiscount             0.084           0.192

  Jajah                    0.076           0.1327

  poivy                    0.084           0.216