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.

Click Here for the Full Article 

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."

Click Here for the Full Article 

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.

Click Here for the Full Article 

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

The OmniAnalysis Platform offers network engineers real-time visibility into every part of the network, simultaneously from a single console, including Gigabit, 10/100, 802.11 wireless, VoIP, and WAN links to remote offices. Engineers can use the OmniAnalysis Platform's local capture capabilities, centralized console, distributed engines, and analysis to troubleshoot faults, fix problems, and restore essential services maximizing network uptime and user satisfaction.

Click Here for the Full Article 

Posted by Dal at 10:35 AM | Direct Link Posted to VoIP Hardware

“Our ultimate vision as a company is to create one contiguous organization and share resources fluidly across the entire organization, irrespective of location or geography, which required strong standards and communication capabilities,” said Mark Dietrick, Burt Hill CIO, in the Communication News article.
 

Click Here for the Full Article 

Posted by Dal at 10:26 AM | Direct Link Posted to VoIP News

The VoSky call center includes a computer answering software. It supports Sype speed-dial and SkypeOut service. Other features include call waiting, caller ID and echo cancellation.

The device has 16-bit, full-duplex linear audio signal support. It is compatible with USB 1.1 and 2.0. It has two RJ-11 ports for connecting with analog phones and PSTN line.

Click Here for more Information 

Posted by Dal at 10:19 AM | Direct Link Posted to Skype

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 www.asterisk.org.

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

 This is the tour plan:

* Amsterdam April 26
* Copenhagen April 27
* Oslo April 28
* Paris May 3
* Brussels May 4
* London May 5
* Stockholm May 19 (Close to Von Europe)

MeetAsterisk is the one-day training that introduces Asterisk for a beginner, both from a business perspective and a technical perspective. You will get insights in how to use Asterisk in your business, as well as an introduction in how to install and set up Asterisk. It's a day filled with information to give you a quick-start with Asterisk.

Find out the complete schedule at http://www.meetasterisk.com and register today!

See you at MeetAsterisk!

/Olle

PS. MeetAsterisk will also contain a brief introduction to the new functions in the coming version of Asterisk - Asterisk 1.4 - to be released this summer.

Posted by Dal at 12:53 AM | Direct Link Posted to Asterisk News

Last fall, I brought Asterisk to SIPit for the first time. I survived, Asterisk survived - but I had a long
list of things to fix. Some very serious errors, some smaller syntax errors and some minor things.

In 30 minutes, we're closing SIPit 18 in Tokyo. This week makes me very proud over the Asterisk
development of the version 1.4 SIP stack. The errors I've found are all very minor and easy to fix.
I have some larger issues that need to be resolved, but those does not affect normal communication unless you have some seriously advanced networks. I even found errors  in other product's SIP stacks!!!

So the Asterisk SIP team is making progress, proven by this week's tests. And now the SIP community here expect us to do even better next time - adding new SIP features 
and functions.  New stuff for you to test!

Due to all the changes I made in the SIP code this week, mostly integration functions from the SIPtransfer branch, and the changes made in other modules of Asterisk the test branch is now temporarily out of synch with svn trunk. I will do my best to restore it early next week, if not earlier.

As soon as I've finished integrating the SIP transfer code, the SIP channel will go into bug fixing mode for the 1.4 release - only adding functions that exist in the bug tracker (pending review) and has been tested. At that time, I'll fork and create a parallel development  branch for chan_sip3. I will start that branch with integrating the sipregister and sippeers branches, as well as some other new code that I have in other branches. After that, we'll look into transactions, transport layer awareness (UDP/TCP) and adding new features. More about chan_sip3 later.

Let me finish with a big thank you to all the people that have supported the work with the SIP stack
- all bug reporters, sponsors, all testers, coders and users. We are moving forward together.
Asterisk 1.4 will be a great product!

Greetings from Tokyo!
/Olle
 

Posted by Dal at 12:49 AM | Direct Link Posted to Asterisk Development

The ability to make calls over the Internet has been around for years but is just now really catching on. Its popularity is growing as more consumers trade dialup Internet service for broadband, almost the only essential for VoIP service. Investors are bullish over the VoIP market because they see the Internet as the gateway to all voice and data communications. It’s created quite some chaos in terms of investments.

Vonage raised $200 million in one round of funding last Spring and has now filed to go public. Skype was bought by eBay Inc. for $2.6 billion in cash and stock in September. According to VentureOne, a unit of Dow Jones & Co., more than $700 million was invested in VoIP startups in 2005, with 76 companies getting funding.

Despite the heightened interest, venture capitalists think there’s room to make money. As the VoIP market expands, the companies needed to support and bring services to the market will be numerous.

Click Here for the Full Article 

Posted by Dal at 10:30 AM | Direct Link Posted to VoIP News

Skype, a division of eBay in San Jose, Calif., is also the latest Internet company to talk openly about the real price of doing business in China.

For Skype, it means that its partner, Tom Online, a subsidiary of leading Chinese language media company TOM Group Limited of Hong Kong, can pluck out certain words from text chats to comply with local laws.

Click Here for the Full Article 

Posted by Dal at 10:24 AM | Direct Link Posted to Skype

End-to-end connectivity anytime, anywhere in the world, has become a fundamental need in today's digital age. The resounding calls for seamless connectivity have motivated the development of cellular technology and broadband wireless access (BWA).

While prior proprietary BWA solutions have mostly suffered early demise or slow adoption as a result of the lack of standardization and poor interoperability, WiMAX (Worldwide Interoperability for Microwave Access has generated much interest as the next evolutionary data-voice enabler.

Those interested in participating in the web/tele-conference should send an e-mail to Mireya Castilla at mireya.castilla@frost.com with the following information: title of analyst briefing, your full name, media/company name, title, telephone number, e-mail address and country. Upon receipt of the above information, the registration details for the live briefing will be e-mailed to you.

"WiMAX is poised to cause a major upheaval in the telecommunications industry," says Sin Siew Teyew, Asia Pacific head of telecoms research at Frost & Sullivan. "The added mobility and farther range it offers over its predecessor Wi-Fi (Wireless Fidelity), and greater bandwidth over 3G (Third Generation) further strengthens its appeal." 

Click Here for the Full Release

Posted by Dal at 10:08 AM | Direct Link Posted to WiMax

Click Here for more Information 

Posted by Dal at 10:05 AM | Direct Link Posted to Asterisk Software

Posted by Dal at 11:06 AM | Direct Link Posted to Asterisk News

Maybe the fast-talking Swede, who launched Skype with his business partner Janus Friis in 2003 and sold it to Ebay last year for an upfront payment of $US2.6 billion ($3.5 billion), is making a point to those of us who have yet to cast off earth-bound telephonic shackles for unlimited conversation in cyberspace.

In Skype's London office, tucked away in a featureless Soho side street near Piccadilly Circus, Zennstrom laughs at the notion and explains that operating a telephone switchboard does not fit the business model.

"We have more than 80 million users and we're getting 250,000 new users a day," he says. "We provide a service for free. If we had full customer service for everyone, we'd have so many people calling us that it would be too costly."

Zennstrom says the office does have "a few" landlines, including one for faxes, but inquiries are dealt with most quickly by filling in a form on the website. In my experience, trying to contact companies this way is ineffective. Then again, Skype is not like most companies.

Zennstrom's admirers in the venture capital community say he is a potential Michael Dell, Jeff Bezos or even Bill Gates, to whom he bears a physical resemblance. If you did not know what he looks like, however, it would be hard to spot the boss in the open-plan office.

Click Here for the Full Article 

Posted by Dal at 10:41 AM | Direct Link Posted to Skype

With Skype refusing to answer emails promptly or even to update their ad-hoc customer service style, Danny Wirken ponders the paperless world of Skype.

Ah, Skype. Time was, my father would often say, that one could correct most problems if you found the right paperwork. While I don't share my father's passion for record keeping (I enjoy paying my bills online and doing my bit for the environment) I must agree that the old man had some sense.

If you want to pay for something like your Skype credit online but don't receive any sort of billing or billing information from Skype as part of the process, then you are surely asking for problems and misunderstandings. But perhaps that is what Skype want?

"I paid on Tuesday. The money left my card on Tuesday. I waited on Tuesday. On Friday, I was able to make a call. No emergency calls is right!!" - anonymous forum user.

While the complaints about Skype's customer service post Ebay-takeover continue to flood in, we can only look at the specific parts of their service which are failing to deliver. So far, that list is pretty long. I sent a test email to Skype as part of my research - and I received an automatic response that surely many of you are already all too familiar with: 'If your inquiry requires a response, we will get back to you within three days.' Uh huh. It's now 13 days and counting.

Click Here for the Full Article 

Posted by Dal at 10:38 AM | Direct Link Posted to Skype

The release of an Apple-branded iPod mobile phone is "logical and inevitable", according to analysts at Visiongain.  Analysts there believe that 2006–2007 will see Apple's product offerings accelerate, reflecting its focus on convergence technologies. They believe Apple will open for business as a virtual mobile phone  network operator, hiring bandwidth from existing network operators.

The findings are contained in Visiongain's latest report: 'Apple in Wireless: Strategic options in a converged marketplace'.

The report looks at Apple's failed partnership to deliver a mobile handset (the ROKR) with Motorola, which the analysts called a "lowest common duct that compromised all of Apple's strengths".

They predict that Apple will embrace mobile more fully and pose a greater threat to the mobile phone industry itself as a virtual network, challenging carriers and a mobile phone handset makers.

The analysts reiterate the persistent rumour that soon-to-launch US network, Helio, will be part of an entry strategy for an 'iPhone'.

Click Here for the Full Article 

Posted by Dal at 10:26 AM | Direct Link Posted to VoIP News

The acquisitions of software companies by VoIP hardware vendors, such as the CopperCom Inc. addition of Switchmaxx about a week ago, illustrate how important having a broad range of application software, beyond phone features, is becoming for these hardware companies and their service provider customers as well.

Other acquisitions that illustrate this trend are Genband’s buy of Syndeo, a developer of softswitch technology, about the same time it changed its name from General Bandwidth to Genband.

Even a leading software provider, BroadSoft Inc., has augmented its offering with the acquisition late in 2005 of Carbon Twelve, a software and multimedia applications developer based in Sydney, Australia.

CopperCom bought Switchmaxx for its capabilities in customer portals, which is viewed as a competitive advantage at a time when IOCs are starting to get competition from both the broadband voice providers and cable companies. “Independents are starting to realize, as they put together bundled services, that they need to create a distinct brand and drive people to their services rather than a cable company or Vonage (Holdings Inc.),” says Chuck Harris, vice president of marketing at CopperCom.

While the addition of Switchmaxx will not greatly expand CopperCom’s addressable market the way its host-remote capabilities did, says Joe McGarvey, principal analyst of carrier IP telephony at Current Analysis Inc., it may well tip the scales in a sale. “Service providers might be swayed by things like the self-care and Web portal capabilities, so there might be differences in selling one piece of hardware over another,” McGarvey says.

Also, CopperCom customers might not be offering broadband telephony to all customers, but with the addition of the CopperCom switch with Switchmaxx software next to a legacy TDM system, all customers can use a Web portal to manage their services and the portal will not change if they are migrated to VoIP. “That was one of the selling points, the fact that it was also compatible with (Nortel Networks Inc.) DMS 10s and other legacy equipment,” says McGarvey.

Competitors in the same space as CopperCom, such as Tekelec and MetaSwitch, also have worked to maximize the application capabilities of their hardware offerings. MetaSwitch has taken advantage of its parent firm, Data Connection Ltd., while Tekelec acquired hosted PBX software company VocalData. Although there are distinct differences in the hardware offerings of all three companies, they all target similar markets, namely IOCs, CLECs, ISPs and cable companies adding voice.

Click Here for the Full Article 

Posted by Dal at 10:19 AM | Direct Link Posted to VoIP News

Technology from Ottawa's PointShot Wireless is a key component of the system, which allows VIA passengers to surf the Web while riding the rails.

Click Here for the Full Article 

Posted by Dal at 09:14 PM | Direct Link Posted to WiFi / Wireless

For the past six months, Vonage has been turning up on average over 112 calling centers in more than 45 new counties each week. Over the past two weeks, Vonage equipped an additional 260 calling centers in over 45 new counties with E911 -- bringing the total number of calling centers across the nation with emergency service to over 4300.

Click Here for the Full Article 

Posted by Dal at 09:10 PM | Direct Link Posted to Vonage

Posted by Dal at 09:08 PM | Direct Link Posted to Skype

The forecast underscores the challenges faced by traditional wire-line telephone service providers as broadband services, portals, ISPs, and other providers increasingly plan to bundle voice services with other offerings, such as cable or satellite television, mobile phone service, and Internet service. Wireline service is traditionally telephone companies mainstay revenue: communications trends may come and go, but regional telephone companies have always been able to rely on revenue from their (often exclusive) monopolies on local telephone service.

Click Here for the Full Article

Posted by Dal at 09:05 PM | Direct Link Posted to VoIP News

It plans to utilize a so-called WiMax system, using television channels that the Federal Communications Commission allocated for educational purposes.  James Davis, MPS director of technology, said he sees WiMax as the means to provide Internet access to students from families too poor to afford a phone line.

The channels have been used to broadcast educational programs into classrooms in a one-way exchange of information, but the WiMax system would provide for a two-way exchange of data.

Click Here for the Full Article 

Posted by Dal at 02:09 PM | Direct Link Posted to WiMax

CCI is continuing with the project and is converting its existing customers to the new service. With the conversion, CCI will be able to lower costs and increase revenues by selling features not available with the older analog switches. With our existing customer revenue base of 1.2 million, CCI's margins are expected to increase from 22% to over 60%. With the newer features, CCI expects to grow revenue to more than $10 million over the next year, with margins exceeding 50% on new business. With the new state-of-the-art services CCI will be able to expand its customer base to underserved regions, such as Alabama, Mississippi, and Kentucky, where CCI already has business presence.

Click Here for the Full Article 

Posted by Dal at 02:01 PM | Direct Link Posted to VoIP News

Posted by Dal at 01:52 PM | Direct Link Posted to VoIP News

Click Here for the Full Article 

Posted by Dal at 07:49 AM | Direct Link Posted to Asterisk Help

Strix Systems introduced a new, third-generation Outdoor Wireless System and a new High Performance Modular Architecture for large-scale mesh networks.The new OWS 2400-30 offers increased user density, provides enhanced transmit power and receive sensitivity, a 30% smaller form factor and lower cost per radio compared to earlier products.

The new design doubles from 64 to 128 the number of WLAN associations permitted per radio, allowing up to 768 concurrent users per node. Modularity lets the OWS 2400-30 scale to up to six radios as needed (802.11a/b/g/j, 802.16d/e, 2.4 GHz, 4.9GHz and 5 GHz) in any access and backhaul configuration.

Posted by Dal at 07:45 AM | Direct Link Posted to WiFi / Wireless

Posted by Dal at 07:41 AM | Direct Link Posted to VoIP News

"In addition, the new release 8.0 features strategic voice and video quality monitoring and real-time compression technologies that will enable customers with converged networks to handle increasing volumes of voice and video traffic between sites without the need to add costly bandwidth," says the company.

Release 8.0 features new voice and video quality-monitoring functions that examine Real Time Protocol (RTP) traffic for jitter, delay and packet loss, says Packeteer, going on:

Posted by Dal at 07:32 AM | Direct Link Posted to VoIP Hardware

The VoIP adapter has a compact and ultra-lightweight design for traveling. It supports SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DHCP, NTP, PPPoE, STUN and TFTP. It also features password enabled VoIP-to-PSTN and PSTN-to-VoIP call routing.

GATE 104 also supports remote device configuration via TFTP server with encrypted configuration files, caller ID display, 3-way conference calling, call blocking, call hold, call waiting/flash, call transfer, call forward, DTMF and dial plans.

Click Here for more Information 

Posted by Dal at 07:27 AM | Direct Link Posted to VoIP Hardware

NEW Energie expects to launch new broadband wireless services based on Nortel WiMAX technology in the second quarter of 2006 to areas which have little fixed broadband coverage. WiMAX is a next-generation technology that uses advanced wireless transmission techniques to bridge the 'last mile' connection between an operator's network and a user, eliminating the need for fixed copper or cable in the ground.

Click Here for the Full Article 

Posted by Dal at 11:16 AM | Direct Link Posted to WiMax

The benefit of VoIP go way beyond the savings, write Lia Timson.  Less cabling, no hardware maintenance and the ability to easily manage the communications needs of fluctuating staff numbers were enough incentive for small business partner Andrew Wall to consider a hosted Voice over Internet Protocol service.

The cost savings achieved by no longer subscribing to landlines were a bonus he realised after the AAPT service began.  As partner in the 10-staff Sydney advertising agency Jam Communications, Mr Wall had not been concerned about the company's phone or internet needs, until it came time to move premises.

Then having gathered some technical knowledge from working with telecommunications clients, he decided to investigate VoIP as a way to minimise set-up costs in the new building.

"We were fitting out the whole office. It was attractive to have all phones, broadband internet and voice mail on the same service, not have to buy any equipment and do all the cabling in one," Mr Wall says.

Click Here for the Full Article 

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

"We tried Vonage at first and it worked for a while," says Jeremie Desautels, CIO of C360. "It worked for a while but as soon as we had more than 4-5 of those Vonage lines, the quality really suffered."

With Vonage, Desautels says, C360 had no effective way of transferring calls, and administration was a headache. "It was a nightmare for me to manage. I had to remember five user names and passwords while making configuration changes." That's because, in Vonage, each user has an individual login to his or her control panel. That works fine in the consumer world, but C360 rapidly outgrew it.

However, having tasted the joys of VoIP, the company didn't want to go back to traditional telephony. It ended up getting an enterprise VoIP solution from specialty vendor Zultys, which sold C360 an IP PBX box. "It's about the size of a pizza box. We screwed it to our server rack and connected it to our T1; 48 hours later, we were up and running," recalls Desautels.

Click Here for the Full Article 

Posted by Dal at 11:06 AM | Direct Link Posted to Vonage

EMC Smarts VoIP Manager provides service providers and enterprises with unprecedented insight and real-time views into the VoIP network by automatically:
o Discovering network elements such as voice switches, IP PBXs (private branch exchanges), media gateways and VoIP servers, as well as all telephony and network application services
o Building a topology of relationships and dependencies among VoIP and the infrastructure
o Establishing device relationships and interconnectivity within the VoIP infrastructure and with all overlying applications

EMC SmartsVoIP Manager is built upon the Smarts common data model that maps devices, relationships, behaviors and interactions across all layers of the IP network, revealing the relationships between network, core applications and business services.

Click Here for the Full Article 

Posted by Dal at 08:58 AM | Direct Link Posted to VoIP News

This is a huge test lab - imagine a large conference hall filled with almost 200 people, each one with
laptops - one or several - phones, video cameras, hubs, coke bottles and other pieces of important stuff that I am not allowed to tell y'all about (sorry for the Huntsville accent  there)...

I will spend one week testing Asterisk with many, many SIP devices to make sure we pinpoint problems and hopefully fix them too. There are teams here with impressive test equipment that tes  almost every possible construction in the SIP protocol and addons. Stressing and fun!

I hope to merge the final parts of the SIP transfer branch into svn trunk today, so that it exists bot in trunk and the test branch. During this week, you will propably see some other bug fixes being integrated into the svn repositories, if needed both in 1.2 and trunk, in some cases only in trunk.

This week of testing will lead to an even better Asterisk SIP stack!

My participation is generously sponsored by:
Digium and Voop - Thank You!

/Olle

PS. This does not mean that you're off the hook. Keep testing!
       A special thank you to Max Bressel for test reports for the 
"sipregister" part of the test branch!!!
       Max - you're the Asterisk Tester of the week!

Posted by Dal at 12:17 AM | Direct Link Posted to Asterisk News

"We have always planned to recruit an experienced CEO as the company entered the mass production and distribution phase of the VisiFone. The proper candidate will enhance our telecom carrier and MSO sales relationships as well as introduce the VisiFone to new markets both in the U.S. and abroad.

Click Here for the Full Article 

Posted by Dal at 10:48 PM | Direct Link Posted to

"We had fears of a temporary restraining order being imposed on us as we implemented our [VoIP] guidelines before. As of now, the telcos [telecommunications companies] are not stopping it," NTC Deputy Commissioner Jorge Sarmiento said.

Sarmiento said, however, that the NTC has received only five applications from potential VoIP providers. These applications do not include local telephone companies who have also launched their respective VoIP services.

Click Here for the Full Article 

Posted by Dal at 10:39 AM | Direct Link Posted to VoIP Politics

First, I will install the EQO plug-in on my PC. During that configuration process I will be asked several questions intended to configure my mobile phone for Skype.  After I go thru the configuration, the plug-in will send the necessary info to my cell.  So if this works, and I am moving around, I am supposed to be able to use my cell to place SkypeOut calls.

Click Here for the Full Article 

Posted by Dal at 11:08 PM | Direct Link Posted to Skype

The public Wi-Fi service, set up with $1.2 million in donated equipment, has been "a lifeline" in the recovery from the deadly storm, serving residents, businesses, public safety officials and building inspectors who have vastly increased their numbers of inspections to help the rebuilding process, CIO and CTO Greg Meffert said in an interview this week. Tropos Networks in Sunnyvale, Calif., and Intel Corp. donated the labor and equipment for the public Wi-Fi after the storm;

Posted by Dal at 11:05 PM | Direct Link Posted to WiFi / Wireless

EarthLink CEO Garry Betty told the Wall Street Journal, "We still don't believe in the free model much," due to the cost of building a network and supporting it afterward.  "We're not in business to break even. We're in the business of making money for our shareholders," Betty said.

Breaking even means earning $7 in ad revenue per customer per month in a municipal WiFi system, and Betty doesn't see search advertising generating that much money. In the past, Microsoft's chief software architect Bill Gates has said each search user is worth about $50 a year to Google.

Click Here for the Full Article 

Posted by Dal at 11:03 PM | Direct Link Posted to WiFi / Wireless

Posted by Dal at 10:59 PM | Direct Link Posted to VoIP Politics

A recent report from eMarketer calls this a battle in the large war to capture the up-for-grabs market, with major players in the cable, ISP, Internet portals and pure-play VoIP space all heading for the front lines to capture at least a part of $190 billion US fixed-line telephony market, which the telecoms have basically had to themselves.

“A fierce battle is emerging in the VoIP market, and is only one part of a larger war that is being waged on three (or more) fronts between telecoms and cable MSOs. At stake is the so-called triple-play, which is a market worth approximately $300 billion,” says Ben Macklin, a senior analyst for eMarketer.

Click Here for the Full Article 

Posted by Dal at 10:57 PM | Direct Link Posted to VoIP News

The National Association of Realtors (NAR) exists to help its 1.2 million members "become more profitable and successful." The NAR provides buying power, education, government policy influence, and the latest technology. In fact, NAR has its own IT department, dedicated to making a real estate agent's job easier through the use of open source technology, called the Center for Realtor Technology (CRT).

CRT has four full-time staff members, and they all do everything open source, says Keith Garner, strategic architect. "My boss, the vice president of CRT, codes as much or maybe more than I do on an average day," Garner says. "It's open source just for the fact of rapid development." Garner and his boss, Mark Lesswing, have worked together at other companies. Garner says back in 1997, Lesswing used to be a "big OS/2 guy. I slowly got him converted over and he started to see the value, to see the 'way,' to understand the value of open source.

When Lesswing applied for a job heading up the CRT, the exact function of the department was still up in the air. "They didn't know exactly what it was going to do," Garner says, but Lesswing came in preaching advocacy, education, and implementation, and that got them both a job. "We advocate to our members the technology they should be using and looking at, and we advocate to the vendors on their behalf. We go to the national, state, and local association meetings for anyone who wants to hear us yap and do educational talks," he says. "We always said we would do stuff as open source, and we would do proof of concepts, full projects, find holes, and explore areas to help the industry as a whole."

A large part of what the CRT team does is proof of concept. They create a low-cost, open source solution for realtors, and make the technology available to IT vendors that brokers hire to outfit their offices, and to the Multiple Listing Service (MLS). For example, Garner and his team coded something they call Messenger, a VoIP-based technology that routes email from potential clients to realtors cell phones. According to Garner, a study found that the average response time from realtors communicating with clients via email was seven days. "By that time, the customer has moved on," he says. Messenger, built on Asterisk, translates text entered at the realtor's Web site into voice and phones it in so the agent can respond quickly. "The guy who was working on it before over-engineered it, and we're simplifying it. The MLS in Texas is providing Messenger as a service to its members."

Click Here for the Full Article 

Posted by Dal at 10:27 PM | Direct Link Posted to Asterisk News

Each day of the conference is filled with presentations and Q&A sessions with many of the leaders in the industry including hardware engineers, programmers and project leaders.  No short course or even full semester can deliver as much information and knowledge as this concentrated exposure to the front lines of Telephony.  If you are looking for information on how to get involved in Telephony to start a new product, to engage in the development of a Telephony application or to learn how to equip your home with things you never imagined were possible this conference will not dissapoint.

Click Here for More Information 

Posted by Dal at 11:54 AM | Direct Link Posted to Asterisk News | VoIP News

Features:

• At your finger tips, easy switching between locations/connections.
• Balloon Tips
• Call Records, searchable too!
• Hides in the System Tray and does not bother you with annoying pop-ups or other nuisances. You don’t even notice it’s there until you need it.
• Intuitive interface. Fast learning curve.
• Letters are automatically dialed as numbers
• Redial button
• Simple Installation
• Supports automatic updates

Click Here for more Information

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

A survey of 522 small to medium-sized enterprises by communications specialist Inclarity in February found that of those questioned, the 12 percent making voice calls over the internet had seen their annual communications bills fall by 23 percent.

IT costs had fallen by 13 percent, largely through the convergence of separate voice and data networks that yielded savings on the cost of equipment, said Inclarity marketing manager David Larkin.

Click Here for the Full Article

Posted by Dal at 11:17 AM | Direct Link Posted to VoIP News

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

According to Datamonitor analyst Thomas Jowitt, ongoing industry debates over future standards are helping to generate investment in the wireless sector. This includes discussions over technologies like Wireless Fidelity (WiFi), WiMax and other emerging technologies like location-based services for mobile phones and content deployment for mobile TV.

This week Santa Clara, Calif.-based Beceem Communications, a provider of chipsets for Mobile WiMax technology, disclosed that it had received an investment from DoCoMo Capital, the venture-capital investment arm of NTT DoCoMo, the Japanese telecom company. The two companies are collaborating on the evaluation of the performance of Mobile WiMax and its desirability as a wireless broadband solution. 

Click Here for the Full Article 

Posted by Dal at 10:46 AM | Direct Link Posted to WiFi / Wireless

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

Arialink, based in Lansing, Michigan, provides a range of telecommunication services, including local and long distance telephone, dial-up, and broadband Internet and metro Gigabit Ethernet access. The planned WiMax service will be offered in Michigan's Muskegon County, Samsung said.

Click Here for the Full Article 

Posted by Dal at 10:26 AM | Direct Link Posted to WiMax

According to a new report by Juniper Research, the market for VoIP business services is set to reach $18 billion by 2010. In a separate report by Integrated Research, based on a survey of 1,232 executives worldwide, 78 percent of large companies say they are deploying IP telephony, largely to enhance communications with IP applications and services such as video conferencing. With business critical applications like voice and video now running on IP networks, guaranteeing uptime, performance and service levels is vital.

While the excitement and promise of VoIP is real, organizations have been cautious in their deployments due to a lack of end-to-end visibility into network performance, and the inability to manage the call quality and reliability that users have come to expect from traditional public switched telephone networks (PSTN). Robust performance management tools will allow enterprise IT departments to aggregate data and use it for baselining, trending, capacity planning and Quality of Service (QoS) monitoring; enable reporting from the service and device levels so that IT organizations can correlate business metrics to IT performance; and provide faster problem resolution.

Click Here for the Full Release 

Posted by Dal at 10:21 AM | Direct Link Posted to VoIP News

Minutes trader Arbinet-thexchange Inc. has made a commercial arrangement with wide-area peering company PacketExchange so VoIP service providers can connect to its voice exchange using a private IP backbone from 24 PoPs in Europe.

The PoPs will function as Arbinet VoIP Exchange Delivery Points, and Arbinet will extend its voice services to PacketExchange customers connecting to the company’s network from all of its PoPs.

"Our agreement with PacketExchange will be extremely beneficial to our VoIP members, allowing them to interconnect at any of these PacketExchange PoPs and transport their VoIP calls to Arbinet’s voice on thexchange platform," said Steve Heap, Arbinet CTO.

Arbinet said the secure network will provide high QoS, and minimize latency and packet loss. "This is especially beneficial for VoIP service providers looking to purchase retail-quality termination through thexchange," Heap added.

Click Here for the Full Article 

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

EQO Communications on Thursday announced the availability of EQO Mobile for Mac OS X users of the Skype VoIP service. The solution enables Skype subscribers to place and receive calls, as well as exchange text messages with other users, on their cell phones. EQO Mobile is compatible with over 45 models of cell phones from Nokia, Motorola, Palm, Sony-Ericsson, and others.

EQO Mobile is currently in Beta testing and is free.

Posted by Dal at 12:06 AM | Direct Link Posted to Skype

Posted by Dal at 12:02 AM | Direct Link Posted to Asterisk Releases

So if you're contemplating VoIP and want to minimize the tasks for compliance with the FCC's e911 mandate, a look at a product called Locate911 is advised. Locate911 is a plug-and-play appliance from eTelemetry that provides real-time VoIP location tracking by automatically linking the VoIP phone to building/room and person.

Contact info is dynamically pulled from the organization’s system of record and may be shared with your call management system or other enterprise solutions via XML Web services. This provides the accurate ALI (Automatic Location Identification) the FCC requires for e911 compliance.

Click Here for the Full Article 

Posted by Dal at 11:55 PM | Direct Link Posted to VoIP Hardware

Nexus' service will be called 'LLU Stream' and will offer wholesale access to Easynet's IP network. LLU Stream will deliver full key stream telephony functionality, primarily to small businesses. Nexus claims that the VoIP platform will be able to completely replace traditional telephony systems and reduce communications costs.

Click Here for the Full Article 

Posted by Dal at 11:33 PM | Direct Link Posted to VoIP News

ESWG further recommends that the Commission order eight specific tasks with sequential milestones to assist with the orderly deployment of i2:

1. CISC should be ordered to deliver within 6 months a preferred PSAP funding model for VoIP E9-1-1 addressing regional/provincial variances and practices to produce a common national standard.

2. CISC should be ordered to deliver a comprehensive architecture for the implementation of VoIP E9-1-1 to deliver within 9 months specifying roles and responsibilities of all emergency services industry participants.

3. All 9-1-1 Service Providers ordered to provide MSAG for the purposes of LIS validity checking within 12 months subject to amended agreements.

4. All Broadband Internet Service Providers be ordered to provide LIS capability within 12 months at their own expense.

5. All 9-1-1 Service Providers be ordered to provide ALI/ANI capability consistent with NENA i2 implementation within 15 months at their own expense.

6. All local VoIP service providers be ordered to provide Call Servers and/or Proxy Gateway capability within 15 months at their own expense.

7. All 9-1-1 Service Providers be ordered to provide ESGW capability within 15 months at their own expense.

8. All VoIP 9-1-1 calls to be E9-1-1 delivered to the correct PSAP within 18 months (Full Production).

ESWG also recommends the establishment of at least one pilot program / test region in Canada to evaluate and determine the best method and practices for transition from legacy to IP emergency services. Finally, ESWG requests Commission continue their practise of fostering advancement in emergency services by providing deadlines for the accomplishment of specific tasks through decisions and order the commencement of this deployment as quickly as is prudent.

Posted by Dal at 01:22 PM | Direct Link Posted to VoIP Politics

And so is the battle in the larger war to capture the up-for-grabs market, with major players in the cable, ISP, Internet portals and pure-play VoIP space all heading for the front lines to capture at least a part of the $190 billion U.S. fixed-line telephony market, which the telecoms have pretty much had to themselves, the report adds.

"A fierce battle is emerging in the VoIP market, and is only one part of a larger war that is being waged on three (or more) fronts between telecoms and cable MSOs. At stake is the so-called triple-play, which is a market worth approximately $300 billion," says Ben Macklin, a senior analyst for eMarketer.

Click Here for the Full Article

Posted by Dal at 07:25 AM | Direct Link Posted to VoIP News

Kansas Gov. Kathleen Sebelius signed a bill into law Thursday that will tack a 50-cent-a-month charge onto Internet-based phone services in the state to pay for enhanced 911 service.  The bill, a watered-down version of legislation introduced during last year's session, had no opponents testify against it this year, according to a conference committee report.

Click Here for the Full Article

Posted by Dal at 07:14 AM | Direct Link Posted to VoIP Politics

Sonorit, based in San Francisco with offices in Denmark and Sweden, was formed in 2005 by ex-employees of Global IP Sound (GIPS). Skype already uses GIPS' SoundWare product, which embeds voice processing technology for IP networks, according to Skype spokesperson Lisa Hempel. AOL, MSN and other VoIP players also use GIPS technology.

The deal, which includes 700,000 shares of eBay stock, is part of Skype's "ongoing efforts to create the best possible audio and video experience for users," Hempel said. Jupiter Research analyst Joe Lazlo said the buy means Skype is ramping up its quality of service.

As Skype moves beyond the stage where the VoIP service was tethered to a computer and expands into a commercial offering with SkypeOut and SkypeIn services, quality of service has become a crucial factor. Lazlo noted that Skype is trying to move closer to VoIP leader Vonage with the deal.

Click Here for the Full Article
 

Posted by Dal at 01:58 PM | Direct Link Posted to Skype

Excerpt: Over the past six months, we've covered lots of territory in building an Asterisk PBX for home or small office use. While most of our projects have relied upon Asterisk@Home, many are easily accomplished using any Asterisk system running the Asterisk Management Portal (AMP) or freePBX.

Today we offer a fresh catalog of available projects on the Nerd Vittles site and some great links to other Asterisk resources on the web. We'll keep the list updated as we add new articles down the road ...

Click Here for the Full Nerd

Posted by Dal at 11:08 AM | Direct Link Posted to Asterisk Help

Posted by Dal at 02:14 PM | Direct Link Posted to Asterisk Software

Yesterday, I implemented a test scenario described to me by Roy Karlsbakk, a Norwegian friend and Asterisk user. I connected two PCs with Linux and Asterisk. Placed one call from the console over SIP to the other server. That server raised the extension with one and dialed back. My server raised the extension with one and dialed back - 200 times. One call going back and forward 200 times before Allison started talking about weasels. Quite a lot of signalling. Quite a lot of files
open.

The freshly installed Ubuntu desktop ran out of file handles and a lot of interesting things happened. One server started sending re-invites, which confused
everything and I ended up with a big mess. A few patches later, things started
working as expected.

A simple test that hopefully improved the Asterisk 1.4 release a tiny bit.

So what happens if I start adding codec translations? If I add a third server? With two or three small systems, you can cause a lot of havoc and stress test. Realtime on one server, two other servers that send registrations with one minute expiry and 200 calls each? 200 authenticated calls that rings for two secs, then hangs up and tries again?

We have things to do, tests to run.

* NEWS IN THE TEST BRANCH

This week, the PostgreSQL driver was committed to SVN trunk. It still needs testing, as the LDAP ARA driver we have in the test branch. A lot of small features was implemented in the test branch, among them a SQLite 3 CDR driver. I am preparing the SIPtransfer branch to be merged into the testbranch, so hopefully that will happen the coming week. Feel free to test it from the siptransfer branch today.

And of course, Mark's HTTP server that was committed directly to svn trunk for testing is also available in the test branch. See it as AMI over HTTP. It's not a full featured Apache, it's a way to reach manager over HTTP as an alternative to TCP. Many script languages can send HTTP requests and receive responses. Mark implmented a tiny test interface in javascript to prove the usefulness of this. I am sure that this will make Asterisk manager development much easier for everyone that works with PHP, JavaScript, Visual basic or any other language with HTTP objects and methods. Play around with it and have fun! You will see that the manager interface needs your love and attention. There are many missing commands, things you still have to use the CLI for.

* A NICE WEEKEND = A TEST WEEKEND

So I wish you a nice weekend, testing the test branch and taking care of your plants. Spending time on both, means a fruitful and colorful summer! Your house is the Asterisk greenhouse!

Read all about the test branch here

README.test-this-branch.html

Thanks for testing, your work is important to all of us!

Regards,
/Olle

Posted by Dal at 08:19 AM | Direct Link Posted to Asterisk Releases

Posted by Dal at 11:12 AM | Direct Link Posted to Asterisk Software

One: you can listen in on another phone extension. This is the method preferred by siblings everywhere. If you have the right access, it's the easiest. While it doesn't work for cell phones, cordless phones are vulnerable to a variant of this attack: A radio receiver set to the right frequency can act as another extension.

Two: you can attach some eavesdropping equipment to the wire with a pair of alligator clips. It takes some expertise, but you can do it anywhere along the phone line's path -- even outside the home. This used to be the way the police eavesdropped on your phone line. These days it's probably most often used by criminals. This method doesn't work for cell phones, either.

Three: you can eavesdrop at the telephone switch. Modern phone equipment includes the ability for someone to listen in this way. Currently, this is the preferred police method. It works for both land lines and cell phones. You need the right access, but if you can get it, this is probably the most comfortable way to eavesdrop on a particular person.

Four: you can tap the main trunk lines, eavesdrop on the microwave or satellite phone links, etc. It's hard to eavesdrop on one particular person this way, but it's easy to listen in on a large chunk of telephone calls. This is the sort of big-budget surveillance that organizations like the National Security Agency do best. They've even been known to use submarines to tap undersea phone cables.

Click Here for the Full Article

Posted by Dal at 10:50 AM | Direct Link Posted to VoIP Security

Note: After helping break the story about SIP to Skype progress it is nice to see products coming out in a more refined form.



SIP protocol is regarded as the industry standard for VoIP. A number of telephone companies, IP phone manufacturers and virtual IP based PBX systems have been using this protocol for connecting calls. On the other hand, Skype has been using a proprietary protocol and whenever one wants to call from a SIP based VoIP service to Skype, one requires an adapter which acts as a gateway so that calls be made as SIP and Skype are not interoperable.

NCH Swift Sound has come with Uplink which connects SIP based voice calls to Skype and it works in both directions. It can be easily configured and used. Once the software has been installed, go to the options to give the path to Skype executable, enter SIP service setting and dial options when Skype calls SIP and vice versa. If configured in the right manner, it works perfectly and provides a good voice quality.

Click Here for more Informaion

Posted by Dal at 03:33 PM | Direct Link Posted to Asterisk Software

iotum's Asterisk integration module connects its award-winning Web 2.0 call management applications to Asterisk IP-PBX's to help users prioritize which calls are important, and which can wait, based on who's calling, and what the user is currently doing. The iotum Asterisk integration module consists of source code, installation instructions and access to an iotum test account.

"Today's announcement represents the next step in iotum's platform strategy", said CEO Alec Saunders. "As of today, members of the world-wide community of Asterisk developers and integrators can use our platform to build new and compelling relevance-enabled revenue generating applications."

Once the integration is complete, this non-commercial beta will allow Asterisk users to filter, rank, and prioritize incoming calls using iotum as well as offer users the ability to easily schedule conference calls from within Microsoft Outlook using iotum's Pronto Conference Calling feature. With, Pronto Conference Calling participants join a conference call by dialing the organizer's usual phone number eliminating the need for everyone to dial into a conference bridge or remember pass codes.

The iotum Relevance Engine API allows Asterisk developers to embed advanced, contextually driven call management capabilities into their own applications. The API allows services to be built which use the iotum Relevance Engine, the world's first smart platform to intelligently assess the relevance of a phone call, and route it to the most appropriate device on any network.. The API is for use in a variety of different kinds of applications, including collaboration, CRM, and personal productivity applications.

For more information about iotum, visit the company's web site:
http://www.iotum.com

Posted by Dal at 11:19 AM | Direct Link Posted to Asterisk Development