NETGEAR’s NEW WIFI PHONE ALLOWS USERS TO MAKE SKYPE CALLS WITHOUT A PC

May 5, 2006 by Garrett Smith

The newly announced Netgear SPH101 Skype WiFi phone is the worlds first device that allows you to make calls on the Skype network anywhere there is a wireless internet connection. The Netgear SPH101 allows users to send and receive call in addition to access to their contact lists through the phone’s full color screen.

Make Skype calls without a PC

The phone’s only major set back is that it can not be used in WiFi hot spots as it does not have a web browser for for web-based login or authentication.

More from: VoIP Hardware Garrett Smith

Plantronics Utilizes DECT Technology for new CS55 Wireless Headset System

by Garrett Smith

Plantronics Utilizes DECT Technology for new CS55 Wireless Headset System

Plantronics has incorporated DECT 6.0 wireless spectrum technology in their new CS55 Wireless Office Headset System. This system combines the clarity of traditional corded technology with the mobility of wireless, so employees can move around the office to wherever business takes them—without interrupting or missing a call.

Features Include:

Roam hands-free up to 300 feet
Extended hands-free talk time—up to 10 full hours
One-touch control over calls and volume
Enjoy the flexibility of three convertible styles
Ensure call privacy with digitally encrypted security
Reduce the wait: battery recharges in three hours

Continuing their tradition as a technology leader, Plantronics presents CS55 as the first headset in the United States to implement 1.9GHz UPCS voice-dedicated wireless VoIP technology. With the included Plantronics HL10 handset lifter, remotely taking or ending a call is as simple as pressing a button.

More from: VoIP Hardware Garrett Smith

Quick Guide to PoE (Power Over Ethernet)

by Garrett Smith

The RJ45 (CAT5) cabling that connects your office or home networking can be utilized for more than transporting voice or data across your local area network (LAN). The IEEE 802.3af Power over Ethernet standard was created as the standard method of delivering power to hardware devices such as IP Phones, IP Video Cameras and other network equipment. With PoE, there is no need to power some devices locally with an external AC power supply, they can simply receive their required voltage over the network cabling.

A POE system is comprised of three elements, the power sourcing equipment (PSE), the device which needs to be powered (PD) and the cable.

The Power Sourcing Equipment (PSE) is connected to a device, and determines if the device is IEEE 802.3af compliant or non-compliant. If determined to be a non-compliant device, no electrical current is passed through the network cable to the device. If the device is determined by the PSE to be a PoE compliant device, the required voltage is supplied to the device to power it.

There are several varieties of Power Sourcing Equipment to fit your specific requirements.

Power Sourcing Equipment (PSE)
Single Port (RJ45) PoE Injector – designed to provide PoE to a single device, a single PoE Injector has (2) RJ45 network connections. The network cable is plugged into the PoE Injector, where the necessary voltage is added. A second ethernet cable runs out of the PoE Injector and is connected to the device to be powered. Some examples of Single Port PoE Injectors include thePW130, and the Linksys WAPPOE”6

Mid-Span PoE Hub
Midspans come is port spans typically ranging from 4-24 ports. A Midspan PoE Hub does not have ethernet switching capability, it is designed to be paired in tandem with an existing ethernet switch, adding voltage for IEEE 802.3af compliant devices as needed. Midspans are a good choice for businesses that have existing 10/100 Ethernet switches already installed, and are less expensive than replacing legacy switches with newer, PoE enabled switches. The Midspan hub is typically stacked on the ethernet switch and each network cable is fed into the hub, where power is added, and then cabling run out to each specific network device. Some examples of Midspan hubs include the SEI Juicebox and a variety of products from PowerDsine.

PoE Switch
Many manufacturers offer ethernet switches with IEEE 802.3af PoE Midspan functionality built in. PoE Switches typically come in port span increments from 12-48 ports, and offer both 10/100/1000 Ethernet managed switching capabilities and PoE injection. Some examples of PoE Switches include the Linksys SRW224P and the Edgewater 2402

Benefits of Power Over Ethernet?

Easy, fast and convenient to install
Reduces overall power consumption
Reduces installation costs
Reduces cable runs
IEEE 802.3af is a unified, worldwide standard
It will save you money

Polycom Communicator C100S – Another Cool New Product for Skype

by Garrett Smith

Recent news from Ebay executives peg the current number of Skype users in the US at 6 million, and growing fast. The company claims more than 100 million users worldwide, but until recently, the majority of Skype users were overseas in Asia and Europe. With rapidly growing usership in the US market, many major electronics vendors are rushing to bring “Skype Certified” products to market.

Polycom, well known for audio and video conferencing, is targeting business customers who utilize Skype service, with its latest product, the Polycom Communicator C100S.

The Communicator is a lightweight, portable speakerphone offering a standard USB connection to a PC or laptop. The Polycom Communicator is a “Skype Certified” product, meaning it has undergone standard interoperability testing in cooperation with Skype, and is officially sanctioned for use with Skype service.

With a retail price point under $150, the Communicator promises a superior user experience versus a standard microphone or headset. In addition to two microphones, the Communicator incorporates an echo cancellation function in the USB driver; in use, it disables Skype’s own echo cancellation. In addition to its primary function, Polycom says it can be used to play music and audio from the connected PC.

Polycom Communicator promises the ultimate hands-free Skype experience. Based on the same technology used in Polycom’s legendary line of triangular SoundStation conference phones, the Skype-certified Polycom Communicator enables crystal-clear, natural conversations when using Skype. Enjoy the freedom of not wearing your headset for hands-free Skype calls, or plug into the built-in stereo headphone port for private conversations.

The Polycom Communicator delivers high-fidelity wideband voice quality that sounds like you are in the same room with the people you’re calling. Two high-quality microphones provide excellent range for group conversations with up to four participants. Polycom’s Acoustic Clarity Technology eliminates echoes and feedback, maximizing your Skype experience.

More from: VoIP Hardware Garrett Smith

RedFone Communications Introduces Dual T1 Version of their popular foneBridge T1 Interface for Asterisk

by Garrett Smith

RedFone Communications has released a new, dual T1 version of their popular foneBridge solid-state T1 interface for use with Asterisk Open Source PBX.

foneBRIDGE is a T1 or E1/PRI-to-Ethernet Bridge. It is an integrated black-box “appliance” designed to streamline installation and enable redundant design of Asterisk based VoIP systems. Designing, building, and installing an Asterisk system has never been this easy!

foneBRIDGE eliminates the need to install proprietary TDM (PRI/T1) hardware cards in approved/compatible server configurations. Instead, foneBRIDGE terminates T1/E1 and/or PRI lines on the trunk side and provides direct Ethernet communication to a network of Asterisk servers using native Asterisk TDMoE formats and utilities.

foneBRIDGE provides an economical means of buliding and maintaining a redundant Asterisk solution. Instead of purchasing a TDM board for each redundant server, multiple redundant servers can share a single foneBRIDGE. If the foneBRIDGE itself fails, replacing it is far simpler that replacing a TDM card in an active server.

Linksys SPA-3102 Analog VoIP Adaptor Now Available at VoIPSupply.com

May 2, 2006 by Garrett Smith

VoIPSupply.com is now carrying the Linksys SPA-3102, the next generation of Linksys 3000 series telephone adaptors.

The SPA-3102 NA from Linksys is slated as the replacement for the popular SPA-3000, and offers both 1 FXS station side port and 1 FXO PSTN jack, and features integrated call routing for local and emergency calls, excellent for service providers and remote offices!

Phone Adapter + PSTN Gateway

The SPA-3102 NA features VoIP adapter functionality found in the SPA-2002 and SPA-1001 with the additional benefit of an integral connection for legacy telephone network “hop-on, hop-off” applications. SPA-3102 NA users will be able to leverage their broadband phone service connections more than ever by automatically routing local calls from cell phones and land lines to a VoIP service provider and vice versa.

A typical user calling from a land line or mobile phone will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA-3000 via a local phone number or by using a telephone connected directly to the unit. The advanced authentication and call routing intelligence programmed into the SPA-3102 NA will connect the caller via the Internet to the far end destination with security and ease. Using the SPA-3102 NA at the far end, calls can be answered immediately or further processed as a local call to any legacy land line or mobile phone allowed by the SPA-3102 NA dial plan.

If power is lost to the unit or the VoIP service is down, calls can be sent to a traditional carrier via the FXO interface.

VoIP Supply, LLC Launches Online Rebate Center

May 1, 2006 by Garrett Smith

VoIP Supply, LLC Launches Online Rebate Center
Rebate Center Features Exclusive Offers on Products from Industry Leading Vendors

BUFFALO, NY (May 1st, 2006) VoIPSupply.com a leading provider of Voice over IP hardware, software and services, today announced the launch of a rebate center for www.voipsupply.com. The rebate center features exclusive instant and mail-in-rebates on VoIP products from industry leading vendors. The rebate center is the result of increased vendor participation in VoIP Supply promotional initiatives.

Benjamin P. Sayers, President and CEO of VoIP Supply, LLC stated “The rebate center offers both consumers and vendors the opportunity to benefit from VoIP Supply’s market presence. Consumers realize greater cost savings, while vendors realize increase sales revenues through VoIP Supply rebate center promotion initiatives.”

The rebate center, which is located at www.voipsupply.com/rebate_center , will be updated weekly with both instant and mail-in-rebate offers on a large variety of products. The rebate center currently features offerings from industry leading vendors such as Polycom, Vegastream, and Zoom Technologies, with savings of up $500 per product.

New MP-11X Analog VoIP Gateways from AudioCodes Shipping

April 19, 2006 by Garrett Smith

AudioCodes has released a new series of flexible analog gateways, the MediaPack MP-11X series, in FXS/FXO spans ranging from 2-8 ports.

The MediaPack ™ Series Analog VoIP Gateways off affordable, feature-rich solutions for connecting legacy telephones, fax machines and PBX systems with IP-based telephony networks, and integrate seamlessly with new IP-based PBX architecture. MediaPack MP11X Series gateways are designed for interoperability with leading Softswitches, H.323 Gatekeepers and SIP servers.

New Models Include:

MP-112 – 2 Port FXS gateway
MP-114-2FXS2FXO – 2 Port FXS / 2 Port FXO gateway
MP-114-FXO – 4 Port FXO gateway
MP-114-FXS – 4 Port FXS gateway
MP-118-4FXS4FXO – 4 Port FXS / 4 Port FXO gateway
MP-118-FXO – 8 Port FXO gateway
MP-118-FXS – 8 Port FXS gateway

MediaPack Series (11x) Features Include:

Spans ranging from 2 to 8 analog ports
Selectable, multiple LBR coders per channel
T.38 compliant
Echo canceller, Jitter Buffer, VAD and CNG
Complies with MGCP, H.323 (v4) and SIP control protocols
Comprehensive support for supplementary services
Enhanced capabilities including MWI, long-haul, metering, CID and outdoor protection
Web management for easy configuration and installation
EMS for comprehensive management operations (FCAPS)

Linksys Ships PAP2T-NA Dual FXS Analog VoIP Adapter

April 13, 2006 by Garrett Smith

The Linksys PAP2T Internet Phone Adapter enables high-quality feature-rich VoIP (voice over IP) service through your broadband Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone ports to connect analogue phones or use one of the ports for a fax machine.

Each phone port operates independently, with separate phone service and phone numbers — like having two telephone lines. You’ll get clear reception and a reliable fax connection, even while using the Internet at the same time.

With Internet telephony, along with low domestic and international phone rates, an impressive array of special telephone features are available. Choose your preferred free local dialing US area code, regardless of where you live. Or add a virtual telephone number in any area code, forwarded to your Internet phone. You can even add a toll-free number.

The Linksys Internet VoIP Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephony service provider, such as Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and much more.

Features on the new Linksys PAP2T-NA Includes:

Two voice ports (RJ11) for analog phones or Fax machines with two independent telephone numbers
One RJ-45 port for 10/100 Mbps Ethernet connection
Supports Dynamic Host Configuration Protocol (DHCP)
Supports Session Initiation Protocol (SIP)
Supports multiple voice compression standards: G.711, G.726, G.729, and G.723.1
Supports Simultaneous calls with G.729 codec
Web-based configuration through a built-in web server
Supports DTMF tone detection and generation
Supports FSK Caller ID, DTMF Caller ID and FSK VMWI
Supports Echo Cancellation and Voice Activity Detection (VAD)
Password protected access and configuration
Supports auto-provisioning with remote firmware upgrade

More from: VoIP Hardware Garrett Smith

Digium Announces New Hardware Products at VON

by Garrett Smith

Digium Announces New Hardware Products at VON
New Transcoder and Echo Cancellations Cards Improve Call Quality and Communication between PSTN and VoIP Gateways

HUNTSVILLE, AL and SAN JOSE, CA — (March 14, 2006) – Source: Digium Press Release — Digium Inc., the creator of Asterisk™, and pioneer of open source telephony, today announced the availability of new hardware solutions to enhance Asterisk transcoding and echo cancellation performance for VoIP and PSTN gateways. These new products include the TC400P VoIP transcoding card and the TE420P and TE415P four-port T1/E1/J1/PRI cards with onboard hardware echo cancellation.

“Our product team is always working to develop solutions like these that ultimately further the open source movement in VoIP,” said Mark Spencer, president of Digium. “Not only are we constantly striving to improve Asterisk’s performance, but we also want to contribute to the overall VoIP experience, while keeping costs low.”

The TC400P provides hardware transcoding of VoIP codecs; decreasing Asterisk’s work load and providing improved CPU efficiency and an increase in channel density over a software-only solution. The TC400P provides Asterisk with full transcoding support and hardware acceleration for the G.723.1 and G.729A codecs.

The TE420P and TE415P improve upon Digium’s existing TE411P and TE406P. These premium interface cards provide carrier-grade echo cancellation, Voice Quality Enhancement (VQE), DTMF decoding, and tone recognition. Both cards minimize loads on the processor and PCI bus, and are designed to perform in the most difficult environments. The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels.

Digium designed these solutions to be fully compatible with existing software applications and is completely integrated with the Asterisk platform. Additionally, the open source driver supports an API interface for custom application development. All new solutions support industry standard telephony and data protocols, including Primary Rate ISDN (both North American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC and Frame Relay data modes.

About Digium
Digium is the original creator and primary developer of Asterisk, the industry’s first open source PBX and Asterisk Business Edition, the professional-grade version of Asterisk. Used in combination with Digium’s PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures.

Digium provides quality hardware and software products that enable telephony applications including legacy PBX, IVR, auto attendant, next generation gateways, media servers and application servers. Digium also offers a full range of professional services including consulting, technical support and customer software development services.

About Asterisk
Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed to from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, and VoIP packet protocols such as IAX, SIP and H.323. It supports US and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respected owners.

NETGEAR’s NEW WIFI PHONE ALLOWS USERS TO MAKE SKYPE CALLS WITHOUT A PC

Plantronics Utilizes DECT Technology for new CS55 Wireless Headset System

Quick Guide to PoE (Power Over Ethernet)

Polycom Communicator C100S – Another Cool New Product for Skype

RedFone Communications Introduces Dual T1 Version of their popular foneBridge T1 Interface for Asterisk

Linksys SPA-3102 Analog VoIP Adaptor Now Available at VoIPSupply.com

VoIP Supply, LLC Launches Online Rebate Center

New MP-11X Analog VoIP Gateways from AudioCodes Shipping

Linksys Ships PAP2T-NA Dual FXS Analog VoIP Adapter

Digium Announces New Hardware Products at VON

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