FCC Deadline for VOIP Provider E-911 Compliance Passes

March 9, 2006 by Garrett Smith

Providers of Voice over IP (VOIP) services may be restricted from enrolling new customers after the FCC specified date for E-911 compliance is past.

The FCC had previously imposed a 120 day window for VOIP providers to deploy reliable, enhanced 911 services in all their service areas by Monday, November 28, 2005.

Since the deadline passed, little public disclosure of E-911 compliance has been offered by the FCC or the affected VOIP providers. It had been previously estimated by the VON Coalition, a VOIP industry consortium, that about two-thirds of VOIP users would have access to E-911 services by the deadline.

Linksys Announces WBP54G 802.11G WIFI “Dongle”

Linksys has a cool new product on the market that should appeal to owners of the popular Linksys/Sipura line of Analog Adapters and SIP phones. The WBP54G adds 802.11G Wireless capabilities to any compatible Linksys/Sipura device, eliminating the need to run RJ45 network cable to network such devices.

Now you can locate a Linksys Phone Adapter or SIP Phone almost anywhere, without the cost and hassle of running network cables. The WBP54G was specially designed to convert a SIP phone or adapter into a wireless device, so it can connect to your home network without an Ethernet cable. This lets you put your phone where it’s most convenient for you, and not be constrained to the area around your Internet connection.

“To make installation even more convenient, the Wireless-G Bridge shares electrical power with the Phone Adapter, so only one AC Adapter (and wall plug) are needed. To get connected, just plug your existing Phone Adapter’s power jack to the Wireless-G Bridge, and the Bridge’s attached power and data cables to the Phone adapter. The included Setup Wizard makes it easy to configure the Bridge to your wireless network’s settings. To protect your data and privacy, all wireless transmissions can be encrypted with WEP or industrial-strength Wi-Fi Protected Access (WPA/WPA2) security.

New SIP PBX Appliance from D-Link – DVX-1000

D-Link has developed a SIP based PBX appliance called the DVX-1000

The DVX-1000 fulfills many of the phone system requirements of the SMB market, including popular features such as call forwarding, call hold, find me-follow me, and voicemail. Inbound calls are routed through the integrated auto-attendant and hunt groups to assist callers to their destinations. The DVX-1000 utilizes standard phone lines via an external phone line gateway or cost effective Internet Telephony

One DVX-1000 VoIP PBX appliance can support up to 25 extensions, which can be located in any physical/geographic location provided that location has Internet access. Multiple units can be peered to increase the number of extensions or unite a company that has many locations under a single PBX system. Additional extensions are added via license codes that are obtained via your reseller or directly from D-Link.

The DVX-1000 IP PBX is fully user configurable via a user friendly web configuration tool. The administrator assigns each extension a profile of telephony features, which allows the best match for a users job function. Each user can fine-tune their assigned profile via the web to match their daily business schedule.

In the world of traditional PSTN telecom, conferencing is typically an expensive external hardware or service. The DVX-1000 includes a phone conferencing bridge, which eliminates the need for an external conferencing server or service provider, and adds distinct value to the D-Link offering. Users are able to schedule and invite parties to conferences via a simple web configuration tool. Conference Notifications are sent out by e-mail, which includes the conference phone number and access codes.

The DVX-1000 uses advanced security features to protect your voice network from unauthorized access. To prevent hackers from breaching the system, the DVX-1000 uses MD5 SIP authentication encryption encoder software. The DVX-1000 also includes an integrated firewall for intrusion detection and protection against denial of service attacks.

The DVX-1000 features a fanless solid-state design offering years of non-stop operation. The compact housing can be easily fastened to the wall of your distribution closet or stacked with your existing Ethernet switches or PSTN Gateways. The DVX-1000 is designed with dual processors for supporting up to 25 simultaneous calls. Its class leading performance allows a 1-to-1 extension to phone line mapping, allowing it to scale with your business.

Digium Ships New TDM2400 Series Full Length Analog PCI Cards

Digium has begun shipping their new TDM2400 Series Full Length Analog PCI Cards to VoIP Supply.

The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium’s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system.

Starting with a base PCB Board that is available with or without Echo Cancellation, the TDM2400 can be configured with FXS or FXO Resources, added in 4 port increments, up to a maximum of 24 total ports. This allows for a wide variety of FXS/FXO combinations, and in many cases eliminates the need for an external gateway or channel bank altogether.

The Digium TDM2400 Series boards are designed for use with Asterisk Open Source PBX

More from: Asterisk Garrett Smith

UTStarcom F3000 – First 2nd Generation WIFI Phone?

UTStarcom has been one of several successful entrants into the WIFI VoIP handset space in recent months. Their initial product, the F1000, set the preliminary standard in terms of usability, feature set and price point. Other vendors including Hitachi-Cable and Zyxel have also brought wireless SIP handsets to market based on the 802.11XX WIFI standard.

The initial crop of wireless VOIP phones all seemed to have their own individual strengths and weaknesses. With the upcoming release of the F3000, UTStarcom seems to have built upon the successful foundation established with the F1000, and have set about addressing some of the limitations and inadequacies of first generation WIFI VOIP handsets.

The UTStarcom F3000 has added support for the current encumbent 802.11G Wireless standard.

The UTstarcom F3000 promises increased security with the addition of WPA encryption, which utilizes the temporal key integrity protocol (TKIP). TKIP utilizes a hashing algorithm to scramble the keys and, by adding an integrity-checking feature, ensures that the keys have not been tampered with.

User authentication, which is generally missing in WEP, through the extensible authentication protocol (EAP). WEP regulates access to a wireless network based on a computers hardware-specific MAC address, which is relatively simple to be sniffed out and stolen. EAP is built on a more secure public-key encryption system to ensure that only authorized network users can access the network.

The preliminary spec we have seen for the UTStarcom F3000 promises to address one of the major factors which has limited the mass-adoption potential of first generation Wireless VoIP handsets, namely Handover/Roaming between different AP and SSID. The ability to “roam” between various access points and SSID within a WIFI network, without losing call connectivity of SIP registration in the process, will seriously bolster the case for WIFI VOIP adoption.

Finally, some aesthetic enhancements are evident in the new F3000, including a larger, color LCD screen and a clamshell form factor typical of current mobile/cellular/GSM phones.

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