Grandstream BudgeTone 101

The single-line Grandstream BudgeTone 101 is one of the most economical choices for use with Asterisk and VoIP chores, with a street price considerably lower than most of the competition. I purchased mine on the Internet for $50 plus shipping. In spite of its low price, the BT-101 is a full-featured phone with all the features I need for an Asterisk PBX in a SOHO environment.

The BT-101 has an LCD display that shows the date, time, volume setting, and connectivity of the unit while it's not in use. When an incoming call arrives, it displays the caller ID. When you pick up the handset to dial, the LCD changes to a light blue background and the number you enter appears where the date had been displayed. Beneath the LCD, a pair of up and down arrow buttons allow you to raise or lower the volume, or cycle through menu options, depending on context. Next to them are buttons to display incoming and outgoing call logs, a menu button, a message waiting light, and a large button you can program to check your voicemail. Hold, Transfer, Conference, and Flash buttons are arranged in a column to the right of the BT-101's keypad. Beneath the keypad there are Speakerphone, Send/Redial, and Mute/Delete buttons.

Getting the BT-101 working with Asterisk requires configuration on both sides of the equation. You can do some configuration of the BT-101 from the phone itself, by pressing the Menu button and negotiating the items in the menu with the up/down arrows, but you can't configure it all from there. For complete access to configuration, you'll need to use the admin pages on the BT-101's built-in Web server. I used both: first configuring the BT-101 with its and the router's IP addresses from the phone, then using the Web interface to complete the configuration.

Click Here for the Full Review 

Posted by Dal at 12:08 AM | Direct Link Posted to Asterisk Hardware

 This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks.

For this revision, we have fixed many bugs and added several new features like Agent-Specific Scheduled Callbacks, Lead Filters and more User Permissions for VICIDIAL. We have also tested the suite on Asterisk versions through 1.2.7.1

All client web-apps and administration pages are available in English, Spanish and Greek, with rough translations of French, German, Italian and Portuguese for the client web-apps only.

Check out the project blog for more information:
http://astguiclient.blogspot.com

Let me know what you think.

Thanks,
MATT---

Posted by Dal at 10:16 AM | Direct Link Posted to Asterisk Software

Here’s how the decision to go with VoIP for a small family-owned business looks from the ground up, concentrating on what’s important to that crucial SMB market:

A family owned business launched in 1938, Pacific Lumber is one of the four companies owned by the Morse family. The family has eight locations including lumber yards, truss plants, door and mill work manufacturing and sales offices. Their 300 employees cater to large and small home builders in Oregon and Washington.

About three years ago, after opening a new lumber yard in Bend, Oregon, the costs to operate and maintain their legacy phone system soon escalated beyond what they could tolerate. Though the telephone equipment was all paid for and worked as advertised, even small changes were complex, expensive and time-consuming.

Lesson: Without an extensive in-house IT department, SMBs love products that are anything but complex, expensive or time-consuming. And yes, they're out there.

Alan Churchill, Director of MIS at Pacific Lumber, began looking at possible system replacements. He hoped that moving to an IP (Internet Protocol) telephony product could save money. Key to the project was the need to connect all locations on a single IP network. “We wanted one person answering the phone for all yards at Pacific,” said Churchill. “Plus we needed a system that was cost-effective and easy to manage. We were looking for a phone system that we could simply plug into our existing WAN.”

Click Here for the Full Article 

Posted by Dal at 10:13 AM | Direct Link Posted to VoIP News

"Skype has grown in leaps and bounds by making it simple for anyone across the world with an Internet connection to do something they could not do before - talk for as long as they like, to whoever they like for no cost. Passing 100 million registered users within such a short time reinforces how much people love how easy Skype makes it to call friends, family and colleagues all over the world for free," said Niklas Zennstrom, CEO and co-founder of Skype. "We owe the Skype community a debt of gratitude for helping us realize this exciting milestone and look forward to keep growing together."

Posted by Dal at 10:10 AM | Direct Link Posted to Skype

This is a good thing for privacy advocates, but may not sit as well with government and law enforcement agents, who see it as an opportunity for terrorists and other criminals to go undetected. Read more here.

Skype was one of the first popular computer-based VoIP services. It's now owned by eBay, and it allows you to make free voice calls and send Instant Messages from your computer to another computer. You can also pay a per-minute fee to make calls to regular landline phone numbers and cell phones through a service called SkypeOut. And there's also a service called SkypeIn, where you're assigned a regular phone number for your Skype account so people can call you from landlines and cell phones. You have to download and install the Skype program, which is available for Windows, Macintosh OS X, Linux and even Pocket PC.

Click Here for the Full Article 

Posted by Dal at 10:07 AM | Direct Link Posted to Skype

Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more.

Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it started as a product purely for ITSPs, it now has much wider scope.

 Details, including links to the feature list and a demo of the web interface, are at:
http://integrics.com/products/enswitch/

In addition to the re-branding, Enswitch 2.0 includes the following new  features:

- Can act as a gateway between legacy PBXs and the PSTN. Plug the E1/T1s from the PSTN into Enswitch, and from Enswitch into the PBX, and Enswitch will route calls between them. Extra features such as SIP handsets, IVR menus, hunt groups, call recording, voicemail, and much more, can then be gradually introduced, allowing a smooth transition from a PBX to a VoIP environment. Routing of numbers away from PBX phones to SIP phones or other features can be done in seconds on the Enswitch web interface.

- Time of day routing can be overridden by calling in from a telephone, entering a password, and choosing which destination to send calls to.

- Improved call shop interface based on customer feedback.

- Announcements can be played before a number rings. Files can be recorded via telephone or uploaded on the web interface from a .wav file.

- Emails sent on self signup and with monthly fees can be customised by each reseller on the web interface.

- Each reseller can set which credit card types they accept.

- Web interface performance improvements for large systems.

- First partners and resellers are active:

http://integrics.com/partners/

We will be formally launching the partner and reseller program soon, and will make an announcement when we do.

--
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/

Posted by Dal at 09:56 AM | Direct Link Posted to Asterisk Software

VoIP Magazine reported on the then-upcoming Yahoo Messenger with Voice enhancements late last year. (See Yahoo's New VoIP Service Could Face Foreign Hurdles.)

Recently added features include Phone Out, Phone In, a Contact Search Bar, and free voicemail. Phone Out enables users to make VoIP calls from a PC to landline or mobile phones at low rates using prepaid credit. Phone In allows subscribers to receive calls on a PC from traditional or mobile phones for $2.99 per month or $29.90 per year, phone numbers can be chosen from a variety of local prefixes. The Contact Search Bar makes finding stored contact information faster by looking up contacts automatically as users type. Yahoo's free voicemail service records messages received via Phone In or PC-to-PC calling.

Click Here for the Full Article 

Posted by Dal at 12:45 PM | Direct Link Posted to VoIP News

In a panel debate, John Meakin, group head of information security at the Standard Chartered Bank, argued that simple IP telephony clients such as Skype would be attractive to corporates if vulnerabilities were removed and performance improved.

"When Wi-Fi came along, people told vendors what a pile of crap the Wireless Equivalent Privacy [WEP] protocol was, and vendors responded; we need that now in VoIP," said Meakin. "Data networks are not perfect. If we are contemplating pushing voice into that same can of worms whilst glibly saying that security is OK, we are up the creek without a paddle."

Click Here for the Full Article 

Posted by Dal at 12:35 PM | Direct Link Posted to VoIP News

Time Warner Telecom Inc., a leading provider of managed voice and data networking solutions for businesses, today announced the successful installation of its SIP IP connections for VoIP services to Boise State University.

"Migrating to Time Warner Telecom's SIP trunk connectivity represents the next evolution of our VoIP communications solution," said Brian McDevitt, manager of telephone network services for Boise State University. "We'll be able to save about half the cost of what we paid previously through significantly reduced network administration, retirement of gateway devices and essentially free long-distance calling between Time Warner Telecom nodes."

Boise State University's campus-wide deployment of VoIP technology over its existing Time Warner Telecom metro Ethernet service encompasses over 14,000 telephone numbers and 4,000 handsets making it the largest university nationwide to do so.

Time Warner Telecom's 20 Mbps capable SIP trunk service replaces existing T1s to cost-effectively boost bandwidth by nearly 20 percent. The simplified, converged network will further assist Boise State managers by allowing them to better manage IT staff headcounts and meet the doubling of voice communications needs. The SIP installation, which is highly scalable, will allow IT managers to connect directly to a VoIP PBX, thus enabling the retirement of six previously required gateways necessary to convert digital voice signals to IP protocol.

"We believe that SIP trunking will change the focus of communications capabilities in an incredible way," explained McDevitt. "It's going to provide a migration path to a whole new generation of applications as well as to significantly change our long-distance cost structure. Domestically, it will certainly have the same impact on business long-distance rates as consumer VoIP already has on residential costs. Eventually, we envision that taking shape internationally as well."

"We're delivering on the promise of convergence today," said Tab Roper, vice president and general manager for Time Warner Telecom in Boise. "The advanced benefits of business-class VoIP are fueling the spread of this technology across the public and private sectors and our SIP trunk solution is ideally suited to support this objective.

It operates over our leading metro Ethernet native LAN platform and connects seamlessly with all Cisco VoIP PBX applications. Our service is scalable from two to hundreds of megabits per second, is highly reliable, and meets the need of every business to reduce costs." Time Warner Telecom's metro Ethernet Native LAN uses Cisco equipment and has been designated a Cisco Powered Network service, signifying that this service is built around Cisco's industry leading technology.

Posted by Dal at 12:26 PM | Direct Link Posted to VoIP News

 
"Spectrum used for WiMAX deployments may prove to be less harmonized. Without a globally recognized frequency band, the economies of scale will be reduced."  Three spectrum areas, 2.5GHz, 3.5GHz and 5GHz, have initially been selected by the WiMAX Forum for certified equipment.

However, these have already been allocated for various other uses in some countries and some have already chosen just one of these for potential WiMAX use.

Click Here for the Full Article 

Posted by Dal at 12:21 PM | Direct Link Posted to WiMax

For example, business owners who rely on Skype for Internet calling can now add speech-enabled automation services to manage tasks such as customer service inquiries and frequently asked questions or add communications capabilities to their existing commerce websites.

Angel.com's Site Builder toolkit also allows Skype users to create new, revenue-generating services, such as help lines for professionals with expertise in a specific topic or industry. Angel.com has become one of Skype's official platform partners, and Skype will promote Angel.com to users as a trusted provider of voice services and development resources.

"Skype's recent launch of Skype for Business confirmed the growing demand for a new breed of business communications solutions," said Michael Zirngibl, President and CEO of Angel.com. "By combining Angel.com's on-demand IVR platform with Skype's Internet communications software, businesses of any size will now be able to set up high-quality IVR and call center applications, without ever having to deal with a traditional telco again."

To encourage Skype users to get started with Angel.com, the companies are currently offering a program that provides guidance and assistance throughout the application-building process. Business owners and individuals who would like to roll out a new application will receive, at no charge, access to Site Builder, dedicated support from an Angel.com representative, a new toll-free number, and promotion of approved voice services to the Skype and Angel.com user communities. To learn more about the program, visit: http://www.angel.com/skype.

"With Angel.com as a new Skype platform partner, our users gain access to a service that makes high-end voice technology available and affordable to businesses and single users alike," said Lenn Pryor, Developer Platform and Relations Team Manager for Skype. "Angel.com is a great fit for our rapidly growing base of business customers."

Posted by Dal at 11:58 AM | Direct Link Posted to Skype

The Wi-Fi service would be free to those who already pay Vonage a $14.25 per month fee to place calls to U.K and Ireland phone numbers. The Cloud has nine hotspots around the UK, with satellite service points in airports, transit stations, coffee shops, hotels, and college campuses.

Click Here for the Full Article 

Posted by Dal at 11:39 AM | Direct Link Posted to Vonage

The beta version of Aspect Unified IP 6.2, expected to be available at the beginning of the third quarter 2006, is the next iteration of the company's Unified IP Contact Center product line. Aspect Unified IP incorporates an automatic call distributor (ACD), voice portal, quality management and recording, a predictive dialer and Internet contact. It also provides unified reporting, routing and administration functionality and hosted capabilities with multi tenancy, while running on the customer's transport of choice. These transport options include:

--  Open source IP PBX solution, such as the Asterisk Business Edition package offered by Aspect Software at significantly less than the cost of traditional IP PBX
--  Closed source IP PBX, offered by companies like Cisco, Avaya and Nortel
--  Any session initiation protocol (SIP) 2.0 compliant PBX
--  Traditional voice telephony
    

The Aspect Software IP strategy takes advantage of the openness and ubiquity of Voice over Internet Protocol (VoIP) notably via SIP to deliver a new and better contact center. It gives customers the choice between embracing IP to address increasingly dynamic processes and practices with its open Unified IP Contact Center product line or to migrate to IP over time with its Signature product line. It provides greater choice, flexibility, productivity and control without compromising the contact center functionality that customers expect from Aspect Software offerings.

"Our Unified IP strategy recognizes the evolution, standardization and market acceptance of IP, notably via SIP, as well as the maturation and market acceptance of a unified contact center offering. And we are seeing increased market opportunities as a result of this natural convergence," said Gary Barnett, chief technology officer and executive vice president of technical services at Aspect Software. "Additionally, as this convergence occurs, we uniquely understand that businesses are demanding a choice of IP plumbing: open source or closed source IP and Aspect Unified IP clearly meets this need."

Posted by Dal at 11:33 AM | Direct Link Posted to Asterisk News

By combining a global threat detection network leveraging real-time reporting by trust-rated users with a unique fingerprinting methodology, Cloudmark is able to identify and begin blocking new spam, phishing and virus attacks within moments, versus hours or days required with competing solutions. Noted for industry-leading speed in detecting and deterring new threats, Cloudmark is uniquely capable of accurately identifying and blocking these spoofed-number attacks. The company detected two new VoIP-specific attacks this week. As a precaution, Cloudmark advises against dialing phone numbers received in emails from institutions and to double-check and dial the numbers printed on ATM cards instead.

Adam J. O'Donnell, Ph.D., senior research scientist at Cloudmark, says, "We've seen two separate VoIP attacks hit our network this week, the first we've been able to analyze in detail. In these attacks, the target receives an email, ostensibly from their bank, telling them there is an issue with their account and to dial a number to resolve the problem." Callers are then connected over VoIP to a PBX (private branch exchange) running an IVR system that sounds exactly like their own bank's phone tree, directing them to specific extensions. In a VoIP phishing attack, the phone system identifies itself to the target as the financial institution and prompts them to enter account number and PIN. "The result," O'Donnell surmises, "can be personally financially devastating."

Traditional content and identity rules based on volume analysis for capturing spam do not work for phishing threats: phishers move quickly, using and breaking down multiple sites to launch the same attack. VoIP-based services allow phishers to cheaply add and cancel phone numbers that are harder to trace than conventional numbers. The Cloudmark Collaborative Security Network's use of unique fingerprinting algorithms is able to identify the phone numbers used in VoIP phishing attacks. The CCSN first spotted and began to block these threats last week. It is characteristic of the network to automatically stop threats without the research team having previously identified them, and thus likely that the CCSN has been stopping VoIP-based attacks for some time.

Dr. Jose Nazario, a senior security engineer within the Arbor Security Engineering & Response Team (ASERT) at Arbor Networks Inc., a network security leader for global business networks, notes, "Cloudmark's large customer base gives them a unique position to detect and prevent phishing attacks, which are highly sophisticated, targeted, transient and dynamic, thereby making it far more difficult to uncover and capture the perpetrators. Leveraging their unparalleled data helps Arbor by enabling its customers to track and stop phishers mid-attack."

Rapid, Intelligent Detection

Cloudmark offers two distinct services to thwart phishers, including an anti-phishing data service that provides confirmed phishing URLs to its customers. The Cloudmark anti-phishing engine fits within the service provider's infrastructure to provide filtering protection at the messaging gateway from fraudulent email. It scans each message and computes a set of fingerprints on the message, a process that is automatic, lightweight and highly scalable for large volumes of email. Cloudmark's approach consistently proves faster and more accurate than competitive methods of relying on fingerprinting algorithms to analyze the structure of messages sent by phishers and block new attacks in advance of receiving URL reports.

About Cloudmark

Founded in 2001, Cloudmark Inc. delivers the industry's fastest and most accurate spam, phishing and virus detection solutions. The Cloudmark methodology leverages an optimized combination of automation, human intervention and real-time reporting by millions of trusted and rated users in more than 160 countries. Used by service providers, enterprises and desktop users worldwide, Cloudmark's award-winning solutions are marketed direct and through partners worldwide. A privately held, San Francisco-based company, Cloudmark sits on the steering committee of the Anti-Phishing Working Group (www.apwg.com). More information about Cloudmark, is available at: http://www.cloudmark.com.

Posted by Dal at 11:25 AM | Direct Link Posted to VoIP News

Gizmo Project, developed by SIPphone, uses a broadband or dial up Internet connection to allow users to make peer-to-peer phone calls using a computer. Gizmo also offers inexpensive add-ons like Call In and Call Out that allows users to connect with the use of a landline phone.

"We chose Vapps' next generation conference calling platform because of the incredible simplicity the service brings to Gizmo users," said Michael Robertson, founder of Project Gizmo. "Traditional conference calling has never been as easy as simply typing in a Web address and connecting up to 500 users in minutes. We are very excited that we can offer this to consumers along with our other services."

This announcement follows closely on the heels of a number of other significant milestones for Vapps. Recently, Salesforce.com has deployed the Vapps VoIP conference calling platform for their small-business users. Vendors around the world are now recognizing that a full suite of Internet calling services is a valuable tool for businesses and Vapps is quickly becoming the provider of choice for servicing these business needs.

"The addition of our advanced conference calling platform to Gizmo's package of service offerings gives consumers a one-stop-shop for Internet calling," said Ben Lilienthal, CEO and co-founder of Vapps. "Our VoIP platform offers an easy and cost-effective method of staying connected, no matter what location you may be in, which is why we are the premier partner for the world's largest Internet communications service providers."

Posted by Dal at 10:29 AM | Direct Link Posted to VoIP News

Time Warner Telecom extends its local networks into suburban office parks and downtown commercial areas to meet customer demand for an alternative fiber facilities-based choice for communications services. This SONET network is similar to the company's 43 other networks across the country that deliver national business-class voice and data solutions, locally and nationally.

"Customer demand typically includes the need for business continuity, diverse routing, data storage, metro Ethernet and a variety of next generation services that only fiber, facilities-based carriers, like Time Warner Telecom and larger incumbents, can offer," said John Schuchart, Time Warner Telecom's vice president and general manager in Dallas.

"This network extension also connects Frisco businesses to our 800-mile fiber ring that runs between Dallas, Austin, San Antonio and Houston, as well as to our national network of 20,000 route miles of fiber and 10 Gbps IP backbone," added Schuchart. "Time Warner Telecom offers communications solutions that enable businesses to converge their networks, reduce their total communications costs and improve their operating efficiencies."

Source: Yahoo Press 

Posted by Dal at 10:15 AM | Direct Link Posted to VoIP News

Vaz Hovanessian, chairman of BroadIP's parent company, ASX listed Broad Investments, said: "Some of the resellers we have secured claim they are currently writing in excess of 100 new VoIP customers a week...Demand has also been very strong from direct small to medium businesses and residential customers since launch.

BroadIP's claims come against the background of rapidly proliferating VoIP service providers. According to figures from telecoms research company, Market Clarity, there are now 128 VoIP providers in Australia, 26 of which offer wholesale services and 119 retail. However Market Clarity has not researched how many of the retail providers have their own infrastructure, as opposed to reselling services from the wholesalers. Eighty six Internet-based VoIP providers primarily focus on the residential market, and 59 on businesses.

Posted by Dal at 10:13 AM | Direct Link Posted to VoIP News

OmniAnalysis Workgroup products provide workgroups and small businesses with powerful but affordable solutions for troubleshooting and monitoring networks. OmniAnalysis Enterprise provides larger organizations with enterprise-class tools for troubleshooting and monitoring large, complex networks, including networks with full-duplex Gigabit or WAN segments. Both product families include a portable analyzer solution that network engineers can use in a laptop, as well as distributed analysis engines that can be installed in remote locations for 24/7 monitoring and analysis. Both product families also feature new enhancements that the company recently announced, including Application Analysis, Network Forensics Analysis and VoIP Expert Analysis.

"Ongoing business automation and Web-centric commerce have led SMBs to deploy increasingly complex, increasingly sophisticated networks," said Mahboud Zabetian, founder and CEO of WildPackets. "With the introduction of the OmniAnalysis Workgroup product family, we're making the Expert analysis and distributed architecture of our enterprise solutions available to SMBs. We're sure that the network engineers in these organizations will appreciate the advantages of remote, real-time analysis and award-winning Expert analysis."

Zabetian continued: "At the same time, we've enhanced the entire platform with new Application Analysis, Network Forensics and VoIP Expert Analysis features that every network administrator, whether working in a small business or an enterprise, will be sure to value."

OmniAnalysis Workgroup Products

The OmniAnalysis Workgroup product family features a new OmniPeek analyzer, called OmniPeek Workgroup, that supports Expert analysis on up to 500 conversations at a time from a single NIC. OmniPeek Workgroup is available for $995. For larger networks, OmniPeek Workgroup Pro, available at $2495, supports analysis on an unlimited number of conversations and from multiple NICs. Both OmniPeek Workgroup analyzers perform local analysis and can connect to up to two OmniEngines at a time. Both analyzers include support for media VoIP analysis. The Enhanced Voice Option, available for $2000, adds support for comprehensive VoIP call analysis and call playback to OmniPeek Workgroup Pro.

Rounding out the OmniAnalysis Workgroup product family, the OmniEngine Workgroup is a remote analysis engine for monitoring a variety of network topologies, including 10/100/1000 and wireless. OmniEngine Workgroup runs on Windows and is available for $1995.

All OmniAnalysis Workgroup products will be available for purchase online in the US and Canada, beginning on May 3, 2006.

OmniAnalysis Enterprise Products

The OmniAnalysis Enterprise product family includes WildPackets top-of-the-line OmniPeek Enterprise analyzer, which supports analysis on an unlimited number of nodes and from multiple NICs, including WildPackets Gigabit and WAN analyzer cards for maximum capture performance. OmniPeek Enterprise, available at $6995, can connect to an unlimited number of OmniEngines and includes support for media VoIP analysis. The Enhanced Voice Option, available for $2000, adds support for comprehensive VoIP call analysis and call playback to OmniPeek Enterprise.

The OmniAnalysis Enterprise family also includes OmniEngine Enterprise, a remote analysis engine for monitoring a variety of network topologies, including 10/100, full-duplex Gigabit, WAN, and wireless. OmniEngine Enterprise runs on Windows and is available for $5995. OmniEngine Enterprise is also available in an Omnipliance 3U rack-mount unit for ease of deployment.

WildPackets OmniAnalysis Platform

WildPackets' OmniAnalysis Platform is a distributed network analysis platform for optimizing network services and maximizing uptime on enterprise networks. The OmniAnalysis Platform uses advanced analytical techniques, including network forensics and application performance indexing, to troubleshoot network problems -- even those that have occurred hours or days ago. Compliance officers can use WildPackets' network forensics features to discover and verify compliance violations, such as those related to server or network access or illicit transmission of data.

Posted by Dal at 10:11 AM | Direct Link Posted to VoIP Hardware

Citel, The VoIP Migration Company, today announced general availability of the EXTender IP6000, enabling enterprises with multiple locations to take a cost-effective, phased approach to IP telephony migration with minimal business disruption.

Rather than supporting multiple PBX systems and remote connections, Citel's 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 the business case evolves and the enterprise is ready to complete the migration to SIP, the EXTender IP6000 can be software upgraded to accommodate a 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, without having to "rip and replace" existing infrastructure.

Hundreds of blue chip clients already rely on Citel's robust EXTender product line to seamlessly distribute the features and applications of a central digital PBX to remote call centers, home workers, and branch offices. The EXTender IP6000 expands this product line with a new lower price point and an assured upgrade path to SIP-based hosted or premise IP telephony in the future.

"Although the market for new telephony infrastructure is shifting rapidly toward IP, many enterprises with existing PBX infrastructure require a more solid business case for immediate migration to VoIP," said Mike Robinson, CEO of Citel. "The EXTender IP6000 addresses this reality across a multitude of PBX platforms. This solution provides an immediately cost-effective and productivity enhancing business case as the first step, followed by a simple, flexible path to complete the migration to IP telephony in the future."

Citel's 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.

Posted by Dal at 10:56 AM | Direct Link Posted to VoIP Hardware

Soyo Group entered the VoIP phone market Tuesday with the introducton of the U201, a dedicated USB handset designed for Skype. Interestingly, Soyo apparently believes that users will prefer to use a wired USB phone but not use the PC to manage the call, as the $45.99 phone (when purchased directly from Soyo) includes a small LCD screen and the ability to manage calls from the handset.

The phone includes a full 16-bit sound card, as well as standard features including a 2.5-mm headphone jack and speakerphone, although the company didn't specify whether it was full- or half-duplex.

Other functions include personalized ring tone, caller ID, Skypeout balance display, call history and more. The phone also displays the SkypeOut number being dialled and, more usefully, the balance remaining in the user's SkypeOut account.

Click Here for more Information 

Posted by Dal at 10:52 AM | Direct Link Posted to Skype

The Aspect Software packaged offering includes:

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.

Click Here for the Full Article

Posted by Dal at 10:43 AM | Direct Link Posted to Asterisk News

* AEL - The Asterisk Extension Language ---------

Last summer, Mark Spencer created a new language for creating your Asterisk dial plan. Before that, many developers tried making the current dial plan "language" into a script language by adding if/then/else and do/while constructs - and it all seemed very strange and, well, not really like a script language.

So Mark decided to take another route and implemented a new language, that was interpreted into the old. You could suddenly create a dial plan in a language that looked more like C, and let the AEL parser create a dial plan based on the old language. This first version was experimental and had a lot of problems. Writing a language parser is not an easy task.

Remember that what you write in the AEL file and what you see when you do "show dialplan" in the CLI is very different. AEL is still interpreted into the old dial plan language.

The new AEL is implemented using Bison, which leads to a much more robust parser. Steve Murphy has put a lot of work into implementing AEL2 and it looks very good. So good, so Kevin removed the "experimental" flag on AEL, making it a standard feature in Asterisk.

* AUTOCONF and MENUSELECT - Installation now is easier! ---------

Since I joined the Asterisk community, I have seen regular requests for a "./configure" script for Asterisk. The Asterisk Makefile replaced some of the functionality of the "./configure" script, trying to find out what functionality was available on the host system.

Yesterday, we finally got an auto-configuration system. The Makefile now creates a configure script, runs it to check what you have - MySQL, OSP, PostgreSQL, CURL etc - and make sure the optimal Asterisk is created on your system. Additionally, you can run "make menuselect" to be able to select what modules you want. No app_dial.so? Just disable it! Menuselect also marks clearly modules that can't be installed on your system due to lacking third party libraries.  And to top it off, we now have ASCII art embedded into Asterisk!

* Making life easier for the Asterisk administrator ---------

hile these additions does not really change the functionality of your favourite PBX, they make installation and configuration of your Asterisk system easier. It's a big step forward and an important part of Asterisk 1.4. Now, I have to learn the inner workings of this and adopt my branches to it... Always good to have something to do ;-) Greetings from the Asterisk Developer Community!

/Olle

Posted by Dal at 01:28 AM | Direct Link Posted to Asterisk Development

Companies or individuals can call the specific language live interpreter and then conference-in the third party on a Skype call (PC to PC) or on a SkypeOut call (calling traditional land or mobile lines). Digitrad has been partnering with Skype to offer enhanced services to Business customers. 

Live Interpretation Service is provided by Digitrad on the IVR-platform Stand4u, a state-of-the-art application that enables any operator to create value added and premium voice services without any previous technical knowledge in record time. "The old telecom business model which used to make money out of communications is about to end", said Micha Benoliel, founder of Digitrad. "In the future, only services will remain as a source of telecommunication revenue. With Stand4U, we provide an easy way to handle the future and make VoIP profitable." To find out more about Digitrad's Live Interpreter by 1TouchConnect go to Skype for Business' home page or on stand4u's website.

Source: Digitrad Visions

Posted by Dal at 05:13 PM | Direct Link Posted to Skype

Highdeal Transactive is a leading-edge modular software suite that provides carrier-grade pricing, rating, charging and billing functionality for today's new emerging services, i.e., VoIP, IPTV and mobile data. Highdeal Transactive prices and rates thousands of transactions per second, thus providing much desired carrier-grade service convergence.

"Our new relationship with Highdeal supports our company's strategy of aggregating 'Best of Breed' VoIP network applications," stated Kevin Burke, COO and CMO of VoIP Logic, "Highdeal Transactive is the most powerful pricing & rating platform in the marketplace; its highly flexible architecture allows us to customize deployments based upon each customers' unique business needs. 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."

Click Here for the Full Article 

Posted by Dal at 05:02 PM | Direct Link Posted to VoIP News

VoIP (or Internet telephony which is almost the same thing) is any one of several technologies that allow you to make phone calls over the Internet instead of over the telephone network. Some more advanced and secure systems use a private data network instead of the Internet. This technology has been around since the 1970s but hasn't been practical until recently because for it to be effective you need a broadband/high-speed connection.

Specifically you need a bit more than 100kbps per connection using modern VoIP transmission technologies. This has only recently become common among residential broadband subscribers. That kind of bandwidth has been available in businesses for longer and the technology is already well established in the business market – but even there the necessary broadband has only been commonly available for three or four years.

In addition, improvements in standards, protocols and underlying hardware and software have also made the required broadband speeds more feasible and have reduced costs to where the decision to move to VoIP is more about the timing and the implementation for a business rather than if it should switch or not.

Typically any VoIP system – residential on up – offers slightly lower operating costs (contrary to advertising claims the cost savings are small) but offers a big step forward in available features and functions. For example, it is now perfectly feasible and cost effective for a 20 person small business to run a call center of its own and to have one system manage main and branch offices and even remote and telecommuting workers.

Click Here for the Full Article 

Posted by Dal at 04:57 PM | Direct Link Posted to VoIP News

 
Decades ago, the government allocated a portion of the 2.5-GHz spectrum to schools nationwide for educational television programming, but much of it hasn't been used. In 2004, the FCC issued a proposal: Any portions of the spectrum not in use or leased by 2008 could be auctioned.

Milwaukee Public Schools--where three out of four kids get the free lunch program for low-income students--hopes to build a WiMax network by next summer to give students free Internet access. "We don't want to lose precious bandwidth that can be used to benefit our low-income students," says James Davis, the Milwaukee school district's director of technology.

The district is treading into uncharted territory. A dozen or so municipalities nationwide are setting up free and low-cost wireless broadband networks, but they're using well-established Wi-Fi on the unlicensed 5.4- to 5.8-GHz spectrum. The Milwaukee school district says it's talking to several WiMax vendors with equipment that works with its 2.5-GHz spectrum, but none of it is certified by the WiMax Forum, an industry standards group. Certified equipment is just now becoming available, and it's initially only for 3.5 GHz, a spectrum used in Europe, Asia, and other places where WiMax is gaining momentum, but one that's not yet approved by the FCC for wireless broadband.

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Posted by Dal at 04:56 PM | Direct Link Posted to WiMax

Posted by Dal at 11:25 AM | Direct Link Posted to Asterisk Software

African ICT market research and analysis company, BMI-TechKnowledge, has announced the publication of its latest report entitled the South African Telecoms Business Customer Analysis (Corporate and SME).

The report identifies the telecommunications trends and spending among the local top corporate and SME businesses, with emphasis on business solutions, emerging technologies, current trends, and telecommunications spend within this sector.

Tertia Smit, senior telecoms analyst at BM-T and author of the report, says that although “least cost routing (LCR)” was the most strategically important technology for the corporate sector, VoIP followed, and then mobile data using a laptop or notebook PC.

In the case of VoIP, the relative degree of its perceived strategic importance was 63% in the medium corporate group versus 68% in the large corporate group.

Click Here for the Full Article 

Posted by Dal at 11:20 AM | Direct Link Posted to VoIP News

eBanx (UK) Limited is a Scottish company that was set up over 4 years ago specifically to hold a 49% share in Beijing Huyang, a Chinese company that was issued the first VoIP license in Beijing. Beijing Huyang now also owns other VoIP licenses and has the rights to resell China Telecom, China Unicom and China Netcom services. It also has the right to develop/acquire its own VoIP technology and sell that in the Beijing area. Beijing Huyang has also been approved for a China-wide national VoIP license subject to further capitalization.

Alan Boyd, a Director of visionGATEWAY, who has facilitated the transaction, said that the opportunity to work with Beijing Huyang will enable visionGATEWAY to expand the distribution of its solutions to the Chinese market, particularly the new secure VoIP solution bundle that visionGATEWAY has in conjunction with Centile's VoIP IntraSwitch solution. Mr. Boyd has been involved in the Chinese Technology market for over 20 years. He co-founded St Banks International Group, a wholly owned intellectual property management company in Shanghai and Tian Na TV and Media Company, the first Western-owned television company to be licensed in China.

A pioneer of the US personal computer industry in the 1970s, Boyd became the first Product Development Manager at Microsoft in 1980, where he reported directly to Bill Gates. At Microsoft, he was responsible for the development of many software products that have since become household names and sold hundreds of millions of copies. He was subsequently responsible for the formulation and implementation of Microsoft's successful acquisitions strategy.

Posted by Dal at 11:18 AM | Direct Link Posted to VoIP News

Said Gary Meyer, data analysis product manager for the Service Assurance Solutions business unit of JDSU's Communications Test and Measurement Group: "JDSU's DA platforms provide the tools needed to ensure reliable and efficient call delivery, a key requirement as VoIP enters the mass deployment stage."

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Posted by Dal at 11:14 AM | Direct Link Posted to VoIP News

With support for up to 20 stations (an optional upgrade supports 40 stations), the BCM 50 amalgamates a wealth of features including voice and unified messaging, contact centre apps, call routing, voice over IP (VoIP) and internet access. With the BCM 50, administrators get the same features found in the products for larger enterprises, but Nortel has reduced costs by using a simple Linux kernel and cheaper components. We reviewed the base BCM 50 unit, which is aimed at network environments that have no requirement for, or already have, data routing installed. Other models are available offering ADSL or Ethernet WAN connections along with integral routing.

Click Here for the Full Review
 

Editor's Note: I hope companies in the future will move away from the "per seat" license model in the pbx market.

Posted by Dal at 11:05 AM | Direct Link Posted to VoIP Hardware

Posted by Dal at 10:20 AM | Direct Link Posted to WiFi / Wireless

Calling it the first law of its kind, Westchester County Executive Andrew Spano said the new law would cut down on identity theft while allowing businesses to avoid the "public relations disasters" that accompany data breaches. He's right about the second part, anyway. When CardSystems was hacked after deciding to contravene its agreement with Visa and keep names and credit card numbers used in transactions it processed, the result was an avalanche of bad press along with a lot of lost business.

According to the county's CIO, county officials found that almost half of the 248 WiFi networks discovered during a 20-minute wardriving session were wide open. That led to the new mandatory security measures for certain businesses, along with a requirement that businesses operating open WiFi networks to post signs to warn their customers about the perils of surfing unprotected networks.

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Posted by Dal at 10:18 AM | Direct Link Posted to WiFi / Wireless

Note: Armijn Hemel has posted details about an upcoming conference in the Netherlands

Hello All,

First of all, sorry for this slightly off-topic post. I asked Mark Spencer permission at FOSDEM in february for posting this mail (which he granted).

On september 14 the NLUUG, the organisation of professional Unix users in the Netherlands, will organize a conference about IP Communication. Possible topics include IP telephony, but also TV over IP, triple play, video conferencing, electronic whiteboarding and so on.

The program committee is looking for speakers with a (technically) interesting story about positive and negative experiences with IP Communications.

The call for papers (deadline May 7) with more information can be found, in both Dutch and English, at:

http://www.nluug.nl/events/nj06/index.html

regards,

Armijn Hemel
program committee NLUUG fall conference 2006

Posted by Dal at 02:15 PM | Direct Link Posted to Asterisk News