Wednesday, April 26, 2006

First Look @ the Polycom IP430

Tuesday, April 25, 2006

Aspect Software


Aspect Software (News - Alert), Inc., the world’s largest company solely focused on the contact center, today announced it will provide and support the Digium open source internet protocol (IP) PBX, the Asterisk Business Edition – a professional-grade version of the industry’s first open source IP –PBX – for customers of its Unified and Signature product lines. The Aspect Software packaged offering includes:

· The Asterisk Business Edition license,

· SIP phones (optional),

· Application servers and IP gateways (optional),

· Interoperability with Aspect contact center products,

· Installation and deployment of the solution, and

· Post-deployment support.

The increased adoption of session initiation protocol (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 open source telephony offers to companies.

“We recognized that organizations desire greater transport choices and our new open source IP PBX offering is one example of Aspect Software developing viable solutions to meet customer demands,” said Gary Barnett, chief technology officer and executive vice president of technical services at Aspect Software. “Asterisk Business Edition provides the capabilities and scalability required to address the needs of the dynamic contact center and when packaged with Aspect Software products, companies can now invest their limited resources in application innovation.”

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 data and voice mail.

“Asterisk is enabling a new wave of businesses and developers to create solutions and applications for the phone system, in ways that are surprisingly similar to the early days of the personal computer,” said Mark Spencer, creator of Asterisk and president of Digium. “When proven leaders like Aspect Software, a dominating player in the contact center industry, join the revolution, that means a great deal for Asterisk and open source in general.”

Aspect Software has demonstrated interoperability between the Asterisk open source IP PBX and Aspect Uniphi Suite from the company’s Unified product line. Aspect has since expanded the number of Unified and Signature line of products that can interoperate with Asterisk, as well as established readiness for installation and ongoing support of the packaged offering through Aspect Global Services.

“Industry experts have acknowledged that the biggest obstacle to wide-spread adoption of open source applications has been installation and ongoing support. With Aspect Software deploying Asterisk Business Edition package, ensuring interoperation with our contact center solutions and offering post-implementation maintenance and support, companies are able to take advantage of the benefits of open source without the worry of disrupting service to customers,” said Barnett. “Now, there is nothing to stand in the way of companies being able to leverage the benefits that open source provides, including inexpensive voice transport.”

“In our view, the underlying PBX infrastructure is a commodity. It is only when our solutions are integrated with that infrastructure that a company can see the real value in their contact center,” said Barnett. “Think of a house. The plumbing is very important to the function of the house – but the type of plumbing installed isn’t a huge differentiator. It is the house and all it has to offer that really sells it to the buyer. The same is true with an IP PBX. Essentially, they all provide the same call handling capabilities and organizations are asking themselves why they should pay for a solution that they can get for little or no cost? It’s when packaged with unified, multi-channel contact center solutions that companies can see the real value and benefits.”

Free VoIP for a year

To kick off the end of the ViaTalk beta period, it is launching a three day promotional offer. If customers sign a one year contract they can get the second year free.

For just under USD $200 customers get one year of service plus one free year (with no activation or equipment fee), unlimited calling to the U.S. and Canada, for new customers.Features include E911, caller ID, operator services, enhanced voice mail and 3 way calling. Optional features include a soft phone, toll free service and priority support. The service is available in the majority of States in the US.This offer ends 4/26, however. ViaTalk is a division of HostRocket, which offers web hosting at discount rates.

Monday, April 24, 2006

Linux Hosting for a Penny!

Linux Hosting Only 1p With Zen Internet

18 April 2006
Zen Internet, a leading UK business ISP (Internet Service Provider) has announced it will be running a promotion offering all new customers the opportunity to trial any Linux Web hosting package for only a penny.

The offer runs until Tuesday 16th May 2006 and applies across the entire range of Linux shared hosting packages. With only a one month minimum contract and free setup, the offer aims to promote Zen’s user experience without any obligation to commit to a longer term contract.

Customers can choose from five packages, each with their own unique proposition. These range from the “Bronze” package aimed at home users to the “Reseller”, aimed at Valued Added Resellers (VARs) looking to establish themselves as independent hosting providers. The Reseller option provides everything needed to allow organisations to add Web hosting to their existing portfolio, therefore increasing their business revenue.

All of the packages provide a complete range of features designed to guide users through every stage of their Web site development. From managing e-mail accounts to adding photo galleries and setting up shopping cart facilities, all can be achieved at the touch of a button.

James Steel, Hosting Product Manager, explained: “This promotion provides the ideal opportunity for us to demonstrate the confidence we have in our product set. Our shared Linux packages provide complete value for money and are affordable to both businesses and home users alike. In a marketplace where barriers to entry are low, even for the more established providers it is important to constantly exceed customer expectations and provide continued value for money.”

This promotion follows the official launch of the shared Linux range back in November 2005, all of which include the cPanel control panel that allows users to manage their Web sites from an easy to use and intuitive front-end application. Key features of the Linux hosting packages include a complete e-mail solution with anti-virus protection, detailed Web site statistics, site design wizards, FTP and database management alongside Zen’s first class customer support.

Zen’s success as a business ISP and UK hosting provider has been acknowledged by many awards and accolades over recent years. Most notably, Zen picked up 3 ISPA (Internet Service Providers’ Association) awards in February 2006, including the highly prestigious award for ‘Best Business ISP’.

For more information or to sign up for the 1p hosting offer visit

Saturday, April 22, 2006

Asterisk Recording Interface Security Bypass and Information

Two vulnerabilities have been identified in ARI (Asterisk Recording Interface), which could be exploited by attackers to gain knowledge of sensitive information.The first issue is due to an error where the "includes/main.conf" file is accessible without authentication, which could be exploited by attackers to obtain sensitive information.The second flaw is due to an input validation error in the "misc/audio.php" file that does not validate the "recording" parameter, which could be exploited by attackers to gain access to arbitrary files (e.g mp3, wav or gsm).

Friday, April 21, 2006

IAX versus SIP

another peice of useful information!

Wednesday, April 19, 2006

Digium Forms the Asterisk Advisory Council; Council to Participate in Managing the Asterisk Open Source Telephony Project

HUNTSVILLE, Ala.--(BUSINESS WIRE)--April 19, 2006--Digium Inc., the creator of Asterisk(TM) and pioneer of open source telephony, today announced the formation of the Asterisk Advisory Council. The Council was developed to respond to the increased interest and participation in the Asterisk open source telephony project.

Composed of five experienced Asterisk community contributors, the Council will assist in the management of the Asterisk open source telephony project. Responsibilities of the council include the selection and supervision of community developers, management of release cycles, and maintenance of Asterisk contributions, among other duties.

"As the Asterisk market continues to grow rapidly on a daily basis, we saw the need to expand the team managing the open source project," said Kevin Fleming, co-maintainer of Asterisk and senior software engineer at Digium. "By identifying these key community members to participate in our council, we can ensure that the project continues to add innovations and improve without any delays."

The following members have been appointed to the council:

-- Brian Capouch, Assistant Professor and Chair of the Computer Science Department at Saint Joseph's College: Capouch has integrated Asterisk with a number of other processors including home automation, network monitoring, camera-based security, and the openWRT distribution of Linux. He teaches a college course on VoIP, has presented at a number of industry conferences, and is working on a forthcoming book on Asterisk to be published by Addison-Wesley.

-- Olle E. Johansson, Asterisk Developer, consultant and Evangelist, founder of Edvina AB, Sweden: Johansson has contributed to the SIP channel among other parts of Asterisk, worked as a bug marshal and has written documentation on the software and the Asterisk wiki. He is also one of the founders of Astricon - the Asterisk conference, and regularly performs Asterisk training sessions.

-- Tilghman Lesher, Developer for VCCH, Inc., a leading provider of innovative solutions based on open source software: Lesher has contributed a large amount of code to the core of Asterisk and is the author of a number of applications and dialplan functions. He has been programming for over twenty years, with eight years of professional experience.

-- Jeremy McNamara, Founder of The NuFone Network, the first Asterisk-based Inter-Asterisk eXchange (IAX) provider: McNamara has been working in all aspects of the telecommunications industry for more than nine years and has extensive experience with the development, testing and deployment of Asterisk-based solutions.

-- John Todd, Tello Corporation: Todd comes from an IP networking background, having worked in several large ISPs, ITSPs, and application service providers. He is currently developing next-generation network elements and systems, some of which involve integrating Asterisk with proprietary systems for customers and providers. Todd is also an active participant and speaker at various VoIP forums and conferences.

Details of the Council's organization, membership, management policies, decisions and current projects will be available on

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.

More on the Linksys SPA941, 942, 922 and Beyond

The SPA-941 is a great example where a business class product can work equally as well for a home user that is looking for a more advanced set of features and customizability with their home VoIP service. Granted, you need to have a whole home network and run a dedicated VoIP PBX to truly take advantage of the full feature set. But isn’t that the goal of any home networking fanatic? The review points out a few interesting facts and flaws of the phone that make it good read. One gripe with the phone is that it only has a single Ethernet jack. This means that you either need a small hub or multiple ports wherever you install the phone if you want to keep the port available to other devices.

Linksys is most commonly known for its consumer oriented home networking gear but has started to move decidedly up-market with the Cisco acquisition. Recent introductions of GigE routers with advanced networking features and IP PBX solutions for small and home offices are clear indications that Linksys is hoping to drive additional product sales in these markets. The good news is that these products and services also drive advances into the consumer end as well. Hopefully we’ll see more advances out of this type of gear so that you don’t have to be a certified PBX installer to configure something like this for your home.

Source - ehomeupgrade

Great How to article on Asterisk@home

Tuesday, April 18, 2006

Unveiling the New snom 300

Available @
Release of the New snom300

The new snom 300 is designed for different environments: for small offices, call centers, lobbies, recreation rooms, or in the home. The demands of everyday office communication can be easily managed through numerous telephone functions. The snom 300 has a headset connection and can be used as a freestanding or wall-mounted model.

A two-line graphical LCD display enables the display of call information, and the menu-driven user interface provides the simplest of feature management. Via the navigation key, the user is guided intuitively through the telephone menu. More complex telephone functions, call details and configuration possibilities are accessible via the browser over the connected PC. Six free user or administrator-configurable (or carrier-pre-configurable) function keys can be easily allocated to security-related menu functions, or assigned to multiple lines.

The snom 300 comes factory-equipped to enable two of its six programmable keys to be configured as line appearances, and snom provides upgrades that let you configure (up to) all six function keys in this way - flexible enough to suit the needs of every user. This option enables an individual adaptation of the device to specific areas of application and the personal user behavior - a functionality that is becoming increasingly popular, particularly in call centers and for sales agents.

As the snom 300 supports all of the common compression codecs such as G.729a and G.723.1, it is compatible with numerous components of other manufacturers and can be used in low-bandwidth environments. An integrated 2 Ethernet port switch enables connection to the network over an RJ-45 interface simultaneously with the PC connection!

The New Linksys Voice System 9000 LVS9000 rocks the IP Telephony - VoIP World!

IP PBX, Phones, Analog Gateway and VoIP Service Enable Small Businesses with Big Business Phone Features and Functionality

IRVINE, Calif., March 13 /PRNewswire/ -- Linksys®, a Division of Cisco Systems, Inc., the recognized leading provider of voice, wireless and networking hardware for the consumer, Small Office/Home Office (SOHO) and small business markets, today announced a new line of SIP-based telephony products targeting SOHO and very small business customers. The new line of IP telephony solutions called the Linksys Voice System 9000 (LVS 9000) includes an IP PBX/Key system, a wide range of IP desktop phones, an Analog Gateway for connection to the Public Switched Telephone Network and a selection of authorized Internet Telephone Service Provider (ITSP) offerings for the Value Added Reseller (VAR) or small business to choose from. Used together with an ITSP voice service, the LVS 9000 provides a complete IP telephony system for small businesses with up to 16 users.
The Linksys Voice System 9000, a fully-featured multi-line Internet phone system for small businesses, includes the following components:

* SPA9000 IP PBX
* Range of IP Desktop Telephones
* SPA3000 PSTN Gateway
* WBP54G Wi-Fi(TM) Dongle to enable IP Telephones to be wireless
* POES5 Power-over-Ethernet (PoE) Dongle
* Select ITSP support

North America ITSPs supporting the LVS9000 include:
* Cbeyond®
* InPhonex®
* Primus Telecommunications®
* Race Technologies®
* VoicePulse®
* Zingotel®

The above ITSPs carry the Linksys Participating Small Business Service Provider designation. This designation means each ITSP has met specific Linksys criteria including specialized training on the LVS 9000, certification of the solution on their network, creation of a service offering to support the LVS 9000 and creation of a partner program which rewards Linksys Approved -- LVS Authorized Partners for selling their services.

Participating Small Business Service Providers may offer their LVS 9000 service to small business through Linksys Approved -- LVS authorized partners or they may offer the LVS 9000 solution bundled with their service directly to small businesses via their website.

In addition to the margins VARs can make selling LVS 9000 hardware and their associated services, Linksys Approved -- LVS authorized partners are eligible to earn bounty and/or recurring revenue for selling the service offerings of Participating Small Business Service Providers. More information is available by contacting any of the above ITSPs.

"There are more than 35 million small businesses world wide looking for custom IP communication solutions that are custom built for their needs," said Tarun Loomba, Linksys Sr. Director of product marketing. "To date most small businesses are forced to use enterprise or consumer voice solutions that are either too costly or difficult to use or not robust enough to support their needs. The new Linksys IP PBX and IP phones bundled with our authorized service provider partner's offering were developed for the small business in mind. The LVS9000 will make the deployment of full featured voice networks easy to install and simple to use at a price small businesses can afford."

The Linksys LVS 9000 series include:

Linksys IP PBX: SPA9000

The Linksys IP PBX lies at the heart of the telephony solution, delivering advanced, multi-line features commonly found in enterprise telephone networks. With the Auto Attendant feature, small businesses with up to 16 users can customize their voice network to direct calls to the right person or department and program the system to allow on- and off-business hour greetings. A promotional message can be created to greet or provide information to callers placed on hold or users can "pipe in" their music of choice. The system also allows users to park calls, fax over the Internet and use the intercom for paging. There are over one hundred telephony features built-in. Out of the box, the SPA9000 supports up to four (4) users. Via a software license key upgrade, up to sixteen (16) users can be supported. Details can be found at

Software Functionality
- Configuration Server
- SIP Proxy and Registrar - Up to 16 Users
- Application Server
- Music on Hold Server
- Media Proxy
- Web Server - For Local Management
- ATA via 2 Onboard RJ11 FXS Ports
Hardware Interface Functionality
- Two Phone Ports:
* Analog Telephone
* Fax Machine
- Two Ethernet Ports
* LAN Connectivity
* Call Traffic and Signaling Routing
* PC Management Interface
IP PBX Features
* Automated Attendant
- Reduce Call Load On Receptionist (No Receptionist)
- Professional Welcome for All Callers
- Single Number Access to All Employees
- Guaranteed Call Completion
- Fast, Easy Set-Up and Maintenance
* Automatic Call Distribution / Routing
* Bridged / Shared Line Call Appearance
* Call Transfer
* Call Forwarding
* Local and Corporate Directory
* Call Pickup
* Group Paging
* Intercom
* Call Hunt Groups
* Direct Inward Dialing (DID) and Voice Mail Integration with ITSP
* Music on Hold
* Do Not Disturb
* Three Party Conference Calling
* More features can be found at

Linksys IP Desktop Phones: SPA901, SPA921, SPA922, SPA941, SPA942 This new line of affordable IP desktop telephones provides a wide variety of phone options for small businesses. Choose phones with one or two Ethernet ports, 1, 2, or 4 extensions, optional Power over Ethernet functionality, or a durable no-display model. These telephones work with all SIP-based IP Telephony solutions, but when used with the SPA9000, the phones install in minutes, automatically configuring themselves with business features set by the Service Provider. With an optional Wireless-G Phone Bridge (WPB54G) users can install phones in hard-to-reach places without running cables. The line of Linksys IP Desktop phones can also be used in conjunction with leading SIP application server platforms to provide an IP Centrex service model for service providers looking to provide PBX/Key system features to small businesses.

Model Ethernet Ports # of Extensions Display
SPA901 1 1 None
SPA921 1 1 Graphical
SPA922 2 (1 with PoE) 1 Graphical
SPA941 1 2, upgradeable to 4 Graphical
SPA942 2 (1 with PoE) 2, upgradeable to 4 Graphical

Linksys Analog PSTN Gateway + Phone Adapter: SPA3000
The SPA3000 provides a means to integrate a legacy Telephone service with the LVS 9000 solution. Also, if Internet service is down, calls can be redirected to a traditional PSTN carrier using the SPA9000 Analog Gateway so service continues without interruption.

* One Line FXO Port Provides Call Routing To and From PSTN
- Back Up in Case of Internet Access Problems
- Back up for Emergency E911 Service
- Transparent Migration from Legacy PSTN Service
- Optional Routing of Local, Toll Free Calls to PSTN
* One FXS Port Provides Connection for Analog Phone or Fax Machine

Pricing and Availability
The Linksys IP PBX, IP Desktop Phones and Analog Gateway were developed for large residential or small business customers. Evaluation units are available immediately through Linksys U.S. Distribution partners, including Ingram Micro, Tech Data, and D&H Distributing. Authorization from Linksys, and a Linksys Voice Service Provider Agreement is required to purchase these products.

VARs interested in selling the LVS 9000 solution and the associated service offerings of Participating Small Business Service providers should first apply for Linksys Approved Partner status at Upon acceptance, a Linksys Approved Partner will need to successfully complete a training module on the LVS 9000 to obtain LVS 9000 Authorization.

Estimated Street Pricing for the new LVS 9000 is as follows:

* SPA9000: $399.99
* SPA9000UPG: $300.00 (Upgrade license for SPA9000 to support 16 phones).

IP Desktop Phones
* SPA901: $89.99
* SPA921: $119.99
* SPA922: $159.99
* SPA941: $149.99
* SPA942: $179.99
* SPA941UPG: $30.00 (Upgrade license for the SPA941 to support 4
* SPA942UPG: $30.00 (Upgrade license for the SPA942 to support 4

IP Analog PSTN Gateway
* SPA3000: $89.99

* WBP54G: Wireless-G Phone Bridge for Linksys VoIP endpoints: $ 39.99
* PS100: IP Phone Power Supply: $ 14.99. Optional power supply for the
SPA922 and SPA942. It must be ordered separately.
* POE5S: PoE Dongle for VoIP: $ 29.99. Enables PoE on Linksys VoIP
endpoints: SPA901, SPA921, SPA941, SPA3000, SPA9000

These new solutions will complement the upcoming deployment of Linksys One solutions. Linksys One is an ideal solution for small business with 5-100 users needing a complete communications solution that addresses voice, video, and applications. The LVS series was developed to address the residential and very small business market.

About Linksys

Founded in 1988, Linksys, a division of Cisco Systems, Inc., (Nasdaq: CSCO - News) is the recognized global leader in Wireless, VoIP and Ethernet networking for consumer, SOHO and small business users. Linksys is dedicated to making networking easy and affordable for its customers, offering innovative, award-winning products that seamlessly integrate with a variety of devices and applications. Linksys provides award-winning product support to its customers. For more information, visit

Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and products are trademarks or registered trademarks of their respective holders. Copyright © 2006 Cisco Systems, Inc. All rights reserved.

The Linksys SPA941 and SPA942

ITSP PROvisioning VoIP Fulfillment - SPA VARS

Alternate Route:

Advanced, Affordable, Feature Rich IP Telephone for SOHO and Small Business!

The Linksys SPA941 and SPA942 are functionally powerful yet easy to use business phone featuring comprehensive interoperability and secure mass deployment capability. Stylish and functional in design, the SPA-941 ideal for residential, SOHO, enterprise and small to medium business service offerings including IP PBX, hosted IP telephony and IP Centrex.

Basic but fully featured business phone with large, hi-res pixel display, full duplex speakerphone and headset port. The Sipura SPA-941 IP telephone can be configured as a two (2) line or, via simple software upgrade, a four (4) lines full featured business phone with pixel based graphical display, speakerphone and headset port. Each line can be independently configured with its own SIP account or shared across a single account. SPA-941 functionally powerful, easy to use business phone featuring comprehensive interoperability and secure mass deployment capability.

The SPA942 ise equivalent to the SPA941 but offers a second ethernet port and power over ethernet. (POE) IEEE 802.3af .

Interoperability and SIP Based Feature Set

Experienced telephony service network operators recognize that technical acumen coupled with responsive pre and post sales support are critical for a successful deployment. Linksys-Sipura's extensive interoperability track record with VoIP industry infrastructure leaders via standards based and platform specific SIP signaling enable network providers to quickly roll-out competitive, feature rich service offerings.

Armed with a mature feature set with hundreds of programmable parameters, the SPA-941 utilizes the call processing functionality found in existing Linksys-Sipura products. Linksys-Sipura VoIP endpoint solutions solve many time-to-market requirements of enterprise users and leverage the advantages of an IP network like easy acceptance of station moves, presence and shared line appearances across geographically separate locations.

Carrier-Grade Security, Provisioning and Management

Sipura's leadership in architecting secure provisioning solutions with detailed performance and troubleshooting features enable network providers to deliver high quality support to their subscribers. The SPA941 offers all the key features and capabilities with which service providers can provide customized services to their subscribers.

The SPA941 can be remotely provisioned and supports dynamic, in-service software upgrades. Secure, encryption-based methods for communication, provisioning and servicing saves providers the time, expense and hassle of managing and pre-configuring or re-configuring customer premise equipment (CPE) for deployment.

Contact ABP to learn more about the Linksys phones and interoperability with softswitches and IPBXs.

Technical Specifications
Product Applications Scenarios
Feature List
Up to Four (4) Lines with Independent Configuration/Registration. The SPA941 ships with two (2) line appearances enabled. A two (2) line upgrade is available via a software license key installed locally using the SPA941 web interface or remotely via a secure profile update.

Pixel Based Display: 128x64 Monochrome LCD Graphical Display

Line Status - Active Line Indication, Name/Number

Menu Driven User Interface

Three Party Call Conference Support

Digits Dialed with Number Auto-Completion. Speed Dial Support

Shared Line Appearance Support

Hands Free Operation

call hold, call waiting

Music on Hold Support

Caller ID Name and Number & Outbound Caller ID Blocking

Call Transfer - Attended and Blind

Automatic Redial

Call Forwarding - Unconditional, No Answer, On Busy

Call park, call pick up, call swap

Hot Line and Warm Line Automatic Calling

URI (IP) Dialing Support (Vanity Numbers)

Personal Directory with Auto-dial (100 entries)

Multiple Ring Tones with Selectable Default Ring Tone per Line. User Downloadable Ring Tones and Ring Tone Generator (Free from

Call Log (60 entries each): Made, Answered, Missed Calls

Call Blocking - Anonymous and Selective

On Hook Default Audio Configuration (Hands Free/Headset)

Call back on busy. Do Not Disturb

Configurable Dial/Numbering Plan Support - per Line

Built-in Web Server for Admin and Config with Multiple Security Levels

Called Number with Directory Name Matching. Calling Number with Name - Directory Matching or via Caller ID

SecureCall Encrypted Voice Communication Support

DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy

Syslog, Debug, Report Generation and Event Logging

Automated Provisioning, Multiple Schemes-Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)

Require Admin Password to Reset Unit to factory Defaults Option


Technical Specifications
Power supply: DC Input Voltage: +5 VDC at 2.0 A Max. 100-240v - 50-60Hz (26-34VA) AC Input, 1.8m cord

Temp range: 41 to 113 º F (5 to 45 º C)

Humidity: 10 to 90% Non-condensing, operating and non-operating

Display: Pixel Based Display: 128x64 Monochrome LCD Graphical Display

Keyboard: Dedicated Illumninated Buttons for: Audio Mute On/Off, Headset On/Off, Speakerphone On/Off. Four (4) Soft Key Buttons. Four-Way Directional Button for GUI Navigation. Voice Mail Message Retrieval Button. Dedicated Hold Button. Settings Button – for Access to Feature, Set-up & Configuration Menus. Volume Control Button – Handset, Headset, Speaker, Ringer. Standard 12-Button Dialing Pad. Mute Button with LED.

Lights: Four (4) Illuminated Call Appearance/Line Buttons with Tricolor LED, Line LED State Indication – Active, Idle, On Hold, Unregistered. Line LED Configurable to 13 Different States (On/Off, Color, Flash). Message Waiting Indicator Light. LED Test Function.

Audio: Built-in Speakerphone and Microphone.

Network: (1) 10/100 BaseT E auto-detect RJ45 (Ethernet), Handset: RJ7 Connector Standard
Headset: 2.5 mm

Configuration: Web Browser Administration & Configuration via Integral Web Server. Telephone Key Pad Configuration via Display GUI Menu / Navigation. Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP. Asynchronous Notification of Upgrade Availability via NOTIFY.

Protocols Internet: IPv4, ARP, DNS, DHCP, ICMP, TCP, UDP, RTP, RTCP, Diffserv, VLAN, SNTP

VoIP: SIPv2, SIP Proxy Redundancy - Dynamic via DNS SRV, A Records, SIP Support in Network Address Translation Networks - NAT (incl. STUN), Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP. Jitter Buffer - Adaptive. Jitter Buffer - Adaptive. VAD - Voice Activity Detection w/ Silence Suppression .

Voice encoding: G.711 aLaw, uLaw, G.726 (16, 24, 32, 40Kbs), G.729A, G723.1 (6.3, 5.5 Kbps), DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO)

Security: Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP. Password Protected System Reset to Factory Default. Password Protected Admin and User Access Authority. HTTPS with Factory Installed Client Certificate. HTTP Digest - Encrypted Authentication via MD5 (RFC 1321). Up to 256-bit AES Encryption.


Product Application Scenarios
Enterprise Solutions

Branch Office Scenarios

Multi Tennant / Dwelling Scenarios

Government Scenarios

Educational Scenarios

Event Telephony Scenarios

Satellite Telephony Scenarios

IPBX OEM Partners

Saturday, April 15, 2006

Post-Disaster Communications Petition - READ - SIGN - SPREAD

The FCC has made the electronic comment filing procedure VERY simple. All you really need to do to weigh in is go to:, enter RM-11327in the first line where it requests the "Proceeding" (this is thePetitions "RM" number), fill in the other minimal contact inforequirements, and submit a brief statement in support.
If you are a blogger and if you would also like to see somethingpositive put in place before the next public disaster, so that refugeesand other victims who have lost access to their phone number andtraditional mode of communication are assured a communicationslife-line, PLEASE spread the word of this petition, and take a momentand submit your own comments.

VoIP Fulfillment - PROvisioning - ISP/VoIP Providers

.e4 is now offering complete ISP and VoIP service Provider fulfillment services. This offering extends our current reach to assist Hosted IP PBX and residential VoIP providers, by lessening the burden of inventory control and configuration.

With .e4 VoIP PROvisoning - ISPs and VoIP startups no longer bear the Up-Front cost associated with large inventories and staffing needs.

"As VoIP Start-ups grow, they currently forfeit profits for market share. Our service allows them to operate with low expense while maintaining very high margins in the early adopter VoIP Market.

This service affords these firms the ability to recieve lower cost ATAs ( Analog Telephone Adapters) and well as VoIP telephones and endpoints from leading edge manufaturers. It also creates the possibility for companies to utilize our VoIP private labeling program which creates brand recognition with the customers that they are working so hard to keep.

For more information or to Join the .e4 buying consortium please visit:

Or Contact:
.e4 Technologies
335W South Airport Road
Traverse City, MI 49686
P: 231.392.5968

Thursday, April 13, 2006

.e4 Launches Eicon Diva Server Line-up


Eicon Networks’ Diva Server BRI is a high-performance, active ISDN server adapter that provides both digital and analog connectivity for an ISDN Basic Rate Interface (BRI). The Diva Server BRI’s on-board Digital Signal Processors (DSPs) enable one server to support any mix of two concurrent communications with ISDN devices, V.90 analog modems, fax machines, and GSM compatible mobile phones. A mix of any two call types can be supported simultaneously making it an ideal open communications platform to handle your fax server, unified messaging, computer telephony (CTI) and remote access server needs. Additional Diva Server BRI cards can be added any time your need for capacity grows, and can be mixed and matched with other members of the Diva Server family.

Saturday, April 08, 2006

Price Drop on the Linksys SPA9000

The linksys SPA9000 SOHO PBX is now much cheaper....

for unbeatable prices visit:

Friday, April 07, 2006

Announcing MediaPack MP-11X Series

To order visit:

- Third Generation of Analog VoIP Gateways MP-112/4/8 - FXS+FXO

A History of Solid Design and Unsurpassed Reliability
MPs are deployed in North America, South America, Europe, Asia and Africa
Over 25 service providers have standardized on MediaPacks
Tier-1 OEM vendors have integrated the MediaPack Series into their iPBX or PBX migration solutions
The MediaPacks Interoperates with more than 30 soft-switches, gatekeepers and SIP proxies
Over 12 million ports based on AudioCodes’ technology are deployed worldwide

To order visit:

Thursday, April 06, 2006

Google Talk and Asterisk

A match made in Heaven!

Google Talk and Asterisk

to learn more visit:

Sunday, April 02, 2006

$100 End User Rebate on the Polycom SoundStation IP 4000

.e4 | VoIP Certified Polycom Reseller

.e4 is proud to announce that it is now a Certified Polycom Reseller:

Available @

Polycom SoundStation IP4000, IP 4000 with $100 End-User Rebate
Polycom SoundPoint IP301, IP 301
Polycom SoundPoint IP501, IP 501
Polycom SoundPoint IP601, IP 601