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March 31, 2007

Lack of Network Expertise Holding Back VoIP?

Note:  I would say having a solid background in Networking, Linux and Telephony helps greatly

Network engineers are concerned that enterprise networks are not ready to handle voice over IP traffic and worse, they aren't sure they have the skills to make the situation better. Those are among the conclusions from a survey of 273 network engineers across the United States fielded by network testing vendor Network Instruments.

While a survey by a test vendor that essentially concludes IT staffs need more testing tools and skills must be taken with a grain of salt, the results nonetheless highlight some important issues surrounding enterprise VoIP deployment.

Of the enterprises surveyed, nearly half said they had implemented VoIP, but 32 percent of those organizations said they lacked the ability to monitor VoIP performance. Other findings:

- Almost 50 percent of those surveyed said they were concerned with their ability to monitor the quality of VoIP service

- 41 percent were unsure of their network's ability to handle the extra bandwidth consumption from VoIP calls

- 36 percent were concerned with the reliability of their VoIP application during periods of heavy use

The full survey results are available for download (PDF) from the Network Instruments Web site.

Click Here for the Full Survey 

Posted by Dal at 04:41 PM | | TrackBacks (0) Posted to VoIP News

March 29, 2007

Call Recorder - Easy Call Recording for Skype

 
 
If you want to record, for example, a podcast or an important phone call over Skype, you’ve probably looked at either WireTap Pro or Audio Hijack Pro. These apps are great and powerful for much more than just Skype, but if you want absolute simplicity and all you really want to do is record both sides of a Skype conversation, Ecamm’s Call Recorder for Skype, just upgraded to version 2.0, may be the best option for you.
Call Recorder is a simple plugin for Skype that sits as a little floating window when you open the app. When you’re in a conversation, either Skype-to-Skype or Skype-to-phone, simply hit the big red record button to save the call until it ends or until you hit the stop button.
 
The resulting recording gives you both sides of the conversation at the full quality you originally heard it in, which is just what you want for a podcast or personal phone call for your own records. With the newest version of Call Recorder, you can record one or both sides of video chats as well.
 
 

Posted by Dal at 10:38 AM | | TrackBacks (0) Posted to Skype

March 26, 2007

Polycom Completes Aquisition of Spectralink

Polycom, announced the expiration of the tender offer by its wholly-owned subsidiary, Spyglass Acquisition Corp., for all outstanding shares of SpectraLink. The subsequent merger to finalize the acquisition will close today, Monday, March 26, 2007. All remaining outstanding SpectraLink shares, other than those held by stockholders who properly perfect appraisal rights under Delaware law, will be converted into the right to receive $11.75 in cash.
Following the merger today, SpectraLink will become a wholly-owned subsidiary of Polycom. "We are delighted to complete our acquisition of SpectraLink," said Robert Hagerty, chairman and CEO of Polycom. "SpectraLink's technology, products, partnerships, and people provide Polycom with immediate leadership in the wireless telephony market.
 
Now, by leveraging Polycom's proven strength in voice and video over IP (V2oIP), we have the unique ability to provide fixed and mobile solutions that seamlessly encompass voice, video, and content collaboration solutions from the desktop, to the meeting room, to the mobile individual."
 
Source: Polycom 

Posted by Dal at 10:23 AM | | TrackBacks (0) Posted to Polycom

March 23, 2007

VoIP Still on FCC's Open Road

A federal appeals court today upheld a 2004 Federal Communications Commission (FCC) ruling that Voice over IP services are interstate in nature and not subject to state public utility regulation. The ruling gives the FCC the responsibility and obligation to decide which regulations apply to Internet telephony.
The ruling is a significant victory for Vonage and other VoIP providers, which have been fending off state efforts to regulate their services. While the FCC's order pre-empted those efforts, numerous states have challenged the ruling.
 
"The end-to-end geographic locations of traditional landline-to-landline telephone communications are readily known, so it is easy to determine whether a particular phone call is intrastate or interstate in nature," the Eight Circuit Court of Appeals ruled. "Conversely, VoIP-to-VoIP communications originate and terminate at IP addresses which exist in cyberspace, but are tied to no identifiable geographic location."
 
Similarly, the court ruled, in VoIP-to-landline or landline-to-VoIP calls, the geographic location of the landline part of the call can be determined, but the geographic location of the VoIP part of the call could be anywhere in the universe. The court did not rule on the status of fixed VoIP services offered by cable companies, claiming it was not yet "ripe for review." "[The decision] allows Vonage to continue growing our business unfettered by outdated pre-Internet regulatory structures,"
 
Vonage CEO Mike Snyder said in a statement. The case stems from a 2003 Minnesota Public Utility Commission decision that Vonage's service was a traditional telephone service and subject to state rules, regulations and tariffs. Vonage won an appeal before the Eighth Circuit, which stayed any state actions until the 2004 FCC order was appealed.
 
 

Posted by Dal at 10:47 PM | | TrackBacks (0) Posted to VoIP Politics

Skype to Offer Money Transfer System via PayPal

Web telephone calling service Skype will shortly begin allowing users to send money to other Skype users via the PayPal online payments system, Skype co-founder Niklas Zennstrom said on Tuesday.
Speaking at a technology conference in Silicon Valley, Skype Chief Executive Zennstrom said the company is working with PayPal, but declined to say exactly when the service would be made available to Skype's tens of millions of global users. Both Skype and PayPal are units of online auction leader eBay Inc.
 
PayPal is already the most popular way that Skype users pay for long-distance Skype phone calls to other phones. "You can send money over Skype," Zennstrom said of the upcoming service plan. "This is basically connecting the Skype community over PayPal. All the user needs is a PayPal account."
 

Posted by Dal at 10:35 PM | | TrackBacks (0) Posted to Skype

March 21, 2007

Asterisk 1.4.2 Released

The Asterisk and Zaptel development teams have released Asterisk 1.4.2.

In addition to minor bug fixes, this release includes:

- Improved SLA support, sample configurations and documentation

- Fixes for incoming DTMF handling in the IAX2 channel driver

There are also two security-related changes in this version:

- A fix for a SIP channel driver remote DoS vulnerability
(http://bugs.digium.com/view.php?id=9313)

- A fix for a SIP channel driver remote DoS vulnerability discovered by INRIA Lorraine
(http://voipsa.org/pipermail/voipsec_voipsa.org/2007-March/002275.html)

All users of Asterisk 1.4 with the SIP channel driver loaded and connected to an untrusted network are urged to update to this release to avoid the possibility of experiencing these problems.

Thanks for your support of Asterisk and Zaptel!

Posted by Dal at 01:03 PM | | TrackBacks (0) Posted to Asterisk Releases

Polycom Adds IP 320 , IP 330 and IP 550 to SoundPoint Family


 
Polycom, Inc. expanded its line of standards-based IP desktop phones with three new models, including two enterprise-grade entry-level phones (IP 320 and IP 330), and a high-performance four-line phone with Polycom HD Voice technology (IP 550).
Polycom will demonstrate its entire line of VoIP phones, including the new models, at the VON Conference and Expo taking place March 20-22, 2007 at the San Jose Convention Center in San Jose, Calif. (Polycom booth #625) "We are listening to customers and delivering VoIP phone solutions that meet their complete range of business telephony needs," said Sunil Bhalla, senior vice president and general manager of voice communications at Polycom.
 
"Through our focused product development efforts, breakthrough technology advancements like HD Voice, and thorough integration efforts with our IP PBX and Softswitch partners, Polycom is able to offer standards-based phone solutions that are second to none in interoperability, quality, performance and value."
 
 
 
SoundPoint IP 330 and 320 - Best-in-class entry-level VoIP desktop phones The SoundPoint IP 330 and 320 are two new entry-level models that feature a full-duplex speakerphone with Polycom Acoustic Clarity Technology for natural, hands-free conversations, an easy-to-read graphical LCD, and integrated Power over Ethernet (PoE) support enabling flexible powering options.
 
These phones are designed to meet the demand for a cost-effective VoIP phone that delivers excellent sound quality and enterprise-grade features. The SoundPoint IP 330 features a dual-port 10/100 Ethernet switch for LAN and PC connection, presenting a cost-effective solution for cubicle workers, as well as call center operators who use a "hard" phone in conjunction with a PC client.
 
The SoundPoint IP 320 has a single 10/100 Ethernet port, making it ideal for common areas such as lobbies, hallways, and break rooms as well as various wall-mounted deployments.
 
"By continuing to expand its desktop VoIP phone line, Polycom is giving customers a broad range of phones for different applications within an organization," said Lynda Starr, senior analyst with the Frost & Sullivan Information & Communication Technologies Practice. "In addition, by increasing the number of models that include Polycom's HD Voice Technology, the company will continue to help differentiate IP telephony offerings, and provide customers with a significant benefit for choosing VoIP over traditional phone networks."
 
 
 
SoundPoint IP 550 - Cutting-edge SIP feature set and Polycom HD Voice The SoundPoint IP 550 four-line VoIP desktop phone provides a powerful tool for users requiring an advanced feature set. The phone includes Polycom HD Voice technology that delivers lifelike, high-fidelity voice richness and clarity which can improve comprehension and reduce listener fatigue.
 
It also features a high-resolution, backlit, graphical LCD; integrated Power over Ethernet (PoE) support; and an XHTML micro browser for productivity-enhancing web applications. The phone offers advanced functionality such as shared call / bridge line appearance, busy lamp field (BLF), and presence*.
 
The SoundPoint IP 550, 330 and 320 all support IEEE 1329 Type 1 full- duplex audio to allow for natural conversation even in double-talk situations. This is the same advanced technology used in the award-winning Polycom conference phones and provides best-in-class performance.
 
"People are so accustomed to the quality of phone calls today that the value of HD Voice may not be immediately clear ... until you hear it," said John Guillaume, senior vice president of sales and marketing at New Global Telecom.
 
"After rigorous testing and side-by-side comparisons in our lab, it became evident that HD Voice quality and clarity surpasses that of legacy systems and can dramatically improve user experience. In fact, with Polycom HD Voice, we can now demonstrate that VoIP quality is substantially higher than traditional networks." The new phones will be supported by most Polycom IP PBX and softswitch partners.
 
Pricing and Availability
 
The Polycom SoundPoint IP 550 is now available worldwide (available in China, Korea, and Brazil in Q2) through Polycom certified channel partners for a MSRP of US $369. The Polycom SoundPoint IP 330 and 320 phones will be available for order worldwide in April (available in China, Korea, and Brazil in Q3) through Polycom certified channel partners for a MSRP of US $179 and $139, respectively.
 
Polycom reserves the right to modify future product plans at any time. Products and/or related specifications referenced in this press release are not guaranteed, and will be delivered on a when and if available basis. *Some described features must be supported by the call server and may require updated software.
 
Source: Polycom Inc. 

Posted by Dal at 11:48 AM | | TrackBacks (0) Posted to Polycom

March 20, 2007

New Networked Fax Servers (1 and 2-port) Announced from Multi-Tech



Multi-Tech announced one and two-port fax servers designed to simplify and lower the cost of networked faxing. The FaxFinder fax servers (Models FF120 and FF220) are turnkey units that include all the hardware and software necessary for sending and receiving faxes via a PBX equipped with digital line (i.e. T1 or PRI) support.
When receiving a fax via the PBX, the appliances route pre-assigned fax numbers to standard analog ports connected to them. The FaxFinder fax servers receive and convert the fax to a PDF or TIFF file, which is attached to an email and sent to the proper recipient. Send fax operations can be from many Windows application using the bundled fax client software for Windows.
 
The FaxFinder fax server supports "Super G3" ITU V.34 standard fax communications, along with two-dimensional compression, real-time fax compression conversion, and ECM (Error Correction Mode). The new FaxFinder fax servers are shipping now. "These two new fax servers provide a major upgrade of our current line,"states Chip Harleman, Vice President of Sales and Marketing for Multi-TechSystems. "We have added significantly more power and memory to the units.
 
The extra processing and memory allows new features and capabilities, such asbeing able to convert incoming faxes to PDF files before sending them in emails, and being able to handle longer documents as a single email. These more robust units also prepare the way for future improvements.
 
The new units still have the same business mission: To make network faxing available to a wider range of companies, including those without a dedicated IT staff by bundling everything needed in a single, easy-to-use package. Considering that most other network fax solutions require expensive network server hardware, the new FaxFinder fax servers will both lower fax-networking costs and simplify the installation process."
 
Source: Multi-Tech 

Posted by Dal at 01:10 PM | | TrackBacks (0) Posted to VoIP Hardware

SIP Application Layer Gateway with Redundancy Released By Aspen Networks

Aspen Networks, today announced the addition of a SIP Application Layer Gateway (ALG) to its Aspen 365-VOIP Multi-link WAN Switch. The product release marks the industry's first SIP ALG that utilizes multiple ISP links to provide VoIP failover, redundancy and SIP survivability.

"We want to accelerate the adoption of VoIP in the SMB market by eliminating concerns about reliability and lowering costs of implementation," said Dan Berger, President and CEO of Aspen Networks. "Our solution protects VoIP from WAN instabilities without requiring you to maintain PSTN lines for failover. The Aspen 365 with our SIP-ALG gives hosted VoIP providers, IP PBX vendors and business users the ability to build-in redundancy over an all-IP, all-broadband network."

The Aspen 365-VOIP is an ideal solution for hosted VoIP providers. The product is installed at the customer premise and supports VoIP traffic on any mix of 2-4 active links (T1, DSL, cable, WISP), eliminating a single link as a point of failure and ensuring continuous service. In the event of failover, Aspen's SIP ALG ensures application consistency for call transfer, voicemail, conference calling, and other features. In addition, Aspen's transparent SIP proxy architecture gives VoIP providers more robust IP phone configuration control.

"We sell reliability and customer support," said Steven T. Francesco, Chairman and CEO of Cohere Communications, a New York City based hosted provider of enterprise grade VoIP telephone services. "There was a gap between disparate broadband technologies and local access providers-- and Aspen has filled that void with integrated ISP/VoIP centric technologies that will raise the bar for next-generation applications. The Aspen 365-VOIP helps us provide our customers with outsourced telecommunications services that they can depend on."

The Aspen 365-VOIP is compatible with Broadsoft, the industry's leading VoIP application platform, and many IP PBX vendors including Allworx, Asterisk-based systems, and others. Aspen also supports direct connection to session border controllers such as Acme Packets and Juniper Networks (Kagoor). With Aspen's SIP survivability feature, outages or "reachability" problems are detected within seconds. Fail-over is immediate and transparent to the user, often with no dropped calls, and ensures that IP phones always get dial-tone.

Source: Aspen Networks 

 

Posted by Dal at 01:00 PM | | TrackBacks (0) Posted to SIP

EarthLink To Unveil WiFi Phone Beta

 
 
EarthLink is expected to announce a beta test for Accton Technology Corp’s WiFi phone as early as tomorrow. Beta testing on the phone has been in the works on EarthLink’s municipal wireless network in Anaheim and it seems to be working very well.
Like any WiFi phone, you’ll be able to make VoIP calls from your home network or on EarthLink’s Municipal WiFi network. As soon as you power on the phone you’ll be able to make quick and cheap phone calls over the internet without sacrificing voice quality.
 
Source:  Crunch Gear

Posted by Dal at 12:52 PM | | TrackBacks (0) Posted to Wireless Hardware

March 19, 2007

Trixbox Asterisk Appliance

 
The trixbox Asterisk Appliance is the only solution designed for trixbox by trixbox. It comes pre-loaded, tested, and configured with the latest version of the trixbox telephony application platform. It’s stocked with hand-selected hardware to give you the highest quality phone calls with VoIP, digital, analog connectivity options.

trixbox Asterisk Appliance Features:

• Rackmount Options – The trixbox Appliance can be mounted five different ways: 19” telephony relay rackmount, cabinet slide-mount, wall mount and free-standing floor or tabletop.

• Telephony Expansion Capability – trixbox Appliance can scale up to 4 T1s or 48 analog lines. The trixbox appliance supports Asterisk®-compatible cards from virtually any telephony card manufacturer.

• Enhanced Reliability – With the dual power supply, the trixbox Appliance Enterprise Edition offers high reliability with the ability to run on multiple power sources; a feature you won’t find on any other telephony appliance.

• Data Security – Mirrored hard drives ensure that your data is always safe from a hardware failure.

• Easy Networking – Featuring network connectivity on the motherboard and an onboard 4-port switch, the trixbox Appliance can be networked in a variety of configurations to meet your specific business needs.

• Increased Air Flow – What’s that loud noise you hear? It’s not your trixbox Appliance! With its intriguing one-of-a-kind custom front panel and limited fan noise, the trixbox Appliance was designed to be seen, not heard.

• LCD Display – With a back-lit 4 line LCD included prominently on the front of your trixbox Appliance, you can check the status of the system, including call and queue metrics/details, all in a single glance.

• Quality Hardware – The trixbox Appliance Enterprise Edition comes with pre-configured Sangoma line cards with industry-leading Octasic echo-cancellation hardware inside to give you high quality phone calls.

click to pre-order - Available June 2007

 

Posted by Dal at 10:42 AM | | TrackBacks (0) Posted to Asterisk Hardware

The Mad Scramble over VoIP Patents

Anthony Cataldo, chief executive of Internet-calling provider VoIP, Inc. (VOII), closely watched the recent patent dispute between Verizon Communications and Vonage Holdings.
 
Upon learning of the Mar. 8 decision by a jury that Vonage infringed on Verizon patents, Cataldo asked lawyers to start proceedings against companies that he says are using his company's technology. "You are going to see a lot of demand letters going out from us," he says.

Letters will go out from him—and a lot of others. On the heels of the Verizon verdict, expect a flurry of lawsuits by holders of patents on the technology that delivers phone calls the same way e-mail travels over the Internet, legal experts and industry executives say.

"We were waiting to see how [the Verizon-Vonage case] was going to work out," says VoIP Chief Operating Officer Shawn Lewis. "[The verdict] has opened up the validity of protecting the patents." There are plenty of VoIP (voice-over-Internet Protocol) patents to protect. The Patent & Trademark Office says there are 2,273 such patents. Many belong to telecom stalwarts like Verizon, AT&T, Motorola, Broadcom, and Cisco Systems. More than 150 patents, filed as early as 1999, were granted as recently as 2007.

Click Here to Continue Reading 

 

Posted by Dal at 10:31 AM | | TrackBacks (0) Posted to VoIP News

March 18, 2007

F3000 - The Latest VoIP Headset from UTStarcom

 
 
We're not talking Skype here. This is a bit different. It's the F3000, the latest VoIP clamshell from UTStarcom. It gives you cellular and Wi-Fi capabilities in the same phone, with a host of protocols thrown in, including the prevalent 802.11b/g. Perhaps the most exciting feature is the Auto-Search function, which finds Wi-Fi networks nearby and stores them for later use.
Users can access the interface in four languages (English, French, Chinese, and Spanish). Other common features include a security suite including caller ID blocking and call rejection (whether caller is known or unknown).

Other than that, the F3000 is a new version of the F1000, which was a hit when it first came out.
 
Source: HT Lounge 

Posted by Dal at 02:15 PM | | TrackBacks (0) Posted to Wireless Hardware

March 16, 2007

Is Wireless VoIP Practical?

Note:  I tend to agree that VoIP via cellphones is not very practical without the major carriers doing some major upgrades to support the huge amount of additional traffics that will be generated via VoIP.  Before that happens I believe we will see "Flat Rate" affordable cell plans so you will just be able to pay a monthly fee and make unlimited calls period. 

VoIP is clearly a hit on conventional data services, but what about wireless services? Can it compete with cellular?

Wireless communications companies like Verizon Wireless, Sprint, and AT&T have been moving as fast as they can to roll out data services. As useful as they can be and as slick as they clearly are, they're much more expensive to operate than conventional wired data services.

Therefore, it's tempting to ask whether they are a practical vehicle for VoIP (Voice over IP). There is good reason to believe that VoIP is not a practical application for these networks for the foreseeable future, and not just for cost reasons.

Even if cost were not an issue, there are technical problems, not to mention the business problem: running Skype or some other free VoIP network could compromise the interests of the companies operating the networks. These companies—which, after all, are first and foremost telephony companies, and their networks first and foremost are cellular telephone networks—are not going to take kindly to low-cost hacks eating into their revenue and using their own networks to do it.
 
In the long term, they may not have a choice if both customers and government demand open access to other voice providers on mobile networks, i.e. "net neutrality." In the shorter term, the telcos don't have to be bullies about it. Market forces will make VoIP on mobile data networks a quixotic effort.
 
As T-Mobile CEO Hamid Akhavan noted recently, VoIP's successes have been on landline networks that have low, flat costs. While costs on mobile networks are dropping, they are certainly not low yet, and definitely not flat. Renegade VoIP companies like Skype aren't giving up on the idea.
 
About a year ago, Skype and Hutchison 3G announced a partnership for bringing Skype to mobile phones, but it's hard to see it being successful any time soon. When customers take advantage of the free Skype service but see a large mobile data charge at the end of the month, they'll switch back to their cellular calls in a hurry. Even Skype concedes this point now, and has put development in this area on the back burner.
 
 

Posted by Dal at 11:08 AM | | TrackBacks (0) Posted to WiFi / Wireless

Nintendo Talks about VoIP Capability on New Pokemon Game

Note:  This should be fun for the gamers.

Nintendo put out a press release today discussing the VoIP features of the latest Pokemon game for the DS.

Pokemon Diamond and Pearl will ship to the US on April 22 along with a headset that can be used to speak with friends. This feature is used for online play and allows players who have each other's friend codes to talk during and after battles using the DS' built in mic or headset. 

“The amazing wireless and voice chat features of Pokémon Diamond and Pokémon Pearl offer gamers something completely new,” says George Harrison, Nintendo of America’s senior vice president of marketing and corporate communications.
 
In the press release, Nintendo compares this feature to a child's first cell phone. With an emphasis on easy use and protection, this feature will hopefully appear on many more DS games to come. The headset will be going for $14.99.
 
Source: DS Advanced 

Posted by Dal at 10:55 AM | | TrackBacks (0) Posted to VoIP News

March 15, 2007

Introducing Phone Genie 2.0 for Asterisk

Excerpt: Phone Genie for Asterisk is an all-purpose web utility that lets you reconfigure virtually anything and everything on your Asterisk system on the fly using an incredibly powerful HTML command language that you can master in a matter of minutes. All you’ll need is a web browser including most cellphone web browsers as well.
What Phone Genie 1.0 provided was a web interface to the complete Asterisk Manager API. With version 2.0, you now get HTTPS secure web access as well as a web interface to both the Asterisk Command Line Interface (CLI) and the Linux command prompt with bash command and script support as well.
 
Phone Genie 2.0 also provides complete control over Asterisk’s internal database: dbget to query the Asterisk internal database, dbput to add and update information in the database, dbdel to delete records from the database, and dbshow to display all or a subset of the Asterisk database contents.
 

Posted by Dal at 11:03 AM | | TrackBacks (0) Posted to Asterisk Help

APIOTEK VP-0002 USB Skype Phone Review (Slim)

 

Note:  Now thats what I call design.  Sleek and cool.

As a VoIP phone, the APIOTEK VP-0002 could be the thinnest in the world with only 7.5mm thickness; it has been 100% developed for use with Skype. Unlike other regular Skype phone handsets, the APIOTEK VP-0002 has a shiny, elegant design due to its stainless metal back casing. It reminds me of the Apple iPod and Nokia 8810 GSM cell phone.

Features：
- ‘STYLISH Metal Mobile Phone Design
- Ultra Slim Design - Only 7.5mm
- Compatible with Skype
- Supports MSN/Yahoo Messenger…
- Compatible with USB V1.1/2.0
- Display LED function
- Supports Record Function(PC/Windows Only)
- Supports PC & Mac

Model Name
 ‘STYLISH VoIP Phone : VP-0002
Description VoIP Phone for Skype / MSN / Yahoo Messenger …
Dimension 7.5D x 36.5W x 123H (mm)
Operation Voltage +3.3V ~+3.0V
(+/-5% wide range power supply)
Interface USB 1.1/2.0
System Environment Operating Temperature: 0°C ~50°C
Operating Humidity: 10% ~90%RH
System Requirements Supports Windows 98, 2000, ME, XP and VISTA ready
Mac 8.6, 9.x, 10.1.2, and above
Design Taiwan Design


Click Here For More Photos and Commentary

 

Posted by Dal at 10:21 AM | | TrackBacks (0) Posted to Skype

March 14, 2007

OctWare Offers Echo Cancellation for Asterisk IP PBX

OctWare introduced its "SoftEcho", a new per channel upgrade option for Asterisk-based IP PBX systems. SoftEcho is a Line Echo Cancellation (LEC) application for Asterisk systems. The software is available as a download for enhancing Asterisk with sound quality.
OctWare said its transparent algorithm that works better than off-the-shelf echo cancellation solutions. The algorithm has been improved through years of testing in large-scale carrier deployments worldwide through Octasic, OctWare’s parent company.
 
That carrier grade quality, already obtainable with interface cards by Digium, Sangoma, and other vendors, that include Octasic’s OCT6100 echo cancellation device, is now available in software form for small to midsized businesses using an Asterisk PBX. SoftEcho’s algorithm is auto-tuning meaning that no adjustments are necessary. SoftEcho learns its environment and quickly adapts to it.
 
Its long, well-covered echo path tail ensures carrier proven quality on any calls in trunk side use. SoftEcho handles double-talk and background noise better than the generic echo cancellation solutions currently integrated into popular Asterisk PBX systems. OctWare’s SoftEcho for Asterisk based IP PBX includes performance and voice quality statistics. Downloads are available for $10 per channel through distributors worldwide.
 
Click Here For More Information: 

Posted by Dal at 10:17 AM | | TrackBacks (0) Posted to Asterisk Software

WiFi Design Claims 300Mbps Throughput

Note: LD has a nice write on some new high throughput wireless design.

Fabless network processor vendor Cavium has teamed up with WiFi software specialist Arada on a Linux-based 802.11n WiFi networking reference design claimed capable of 300Mbps throughputs. The design uses Atheros's "XSPAN" technology, and targets both enterprise and in-home applications, including video streaming and triple-play gateways.

Several years ago, Cavium collaborated with WiFi chip vendor Airgo on a demonstration of what was claimed at the time to be the first 108Mbps WiFi implementation. Now, Cavium claims that the performance attributes of its Octeon CN30xx family of network processors enable 802.11n performance that outstrips the competition in WiFi throughput, while leaving lots of cycles for packet inspection or other application processing.
 
The chips cost $20 to $125, have one or two MIPS64 cores, and have previously been marketed with Linux-based network routing software stacks from Team F1. The 300Mbps design incorporates Atheros's AR5008 WiFi chipset, said to feature "draft 802.11n XSPAN technology." XSPAN appears to be a MIMO (multiple input, multiple output) strategy based on a "3 x 3 transmitter/receiver architecture."
 

Posted by Dal at 10:11 AM | | TrackBacks (0) Posted to Wireless Hardware

March 13, 2007

New Enterprise Class Wireless Access Point (MP-422) Released From Trapeze Networks

Trapeze Networks, provider of Smart Mobile wireless solutions, today announced they are shipping the new MP-422 dual radio 802.11a + 802.11 b/g Access Point (AP) with Smart Mobile Intelligent Switching, the latest addition to an extensive line of high performance, scalable, and secure enterprise Wi-Fi infrastructure products.
As large organizations transition from pilot and field trials to real enterprise installations, they need to ensure their Wi-Fi infrastructure meets stringent requirements in security, management, performance, and scalability. Trapeze Networks is uniquely qualified to deliver on each of these critical enterprise requirements. Most Secure Enterprise Wi-Fi System Available Today Enterprise-class Wi-Fi security systems require four key elements -- endpoint integrity assurance, strong authentication and encryption, application-based mobile firewall enforcement, and an integrated Wireless Intrusion Detection System.
 
Only Trapeze Smart Mobile products can deliver on these requirements, and as such, Trapeze Networks is the recognized leader in Wi-Fi security as cited by industry analysts. Most recently, Trapeze Smart Mobile products were praised by analyst firm ABI Research, which named Trapeze Networks #1 worldwide in enterprise Wi-Fi security in their December 2006 report.
 
The recognition put Trapeze ahead of Cisco Systems, Aruba Networks and other firms. This highly coveted award was the result of several Trapeze security initiatives, including the Trapeze integration of AirDefense(TM) technology, the award-winning Wireless Intrusion Detection System, and the only one that is Common Criteria certified by the Department of Defense.
 
Trapeze Smart Mobile products protect against 250 types of wireless network attacks compared to just 24 for Cisco and 40 for Aruba. Smart Mobile is the only solution that includes Dynamic Threat Response, which turns every Access Point on the network into an intrusion detection system when a potential threat is identified. The new MP-422 extends this leadership by implementing these security features. By contrast, Aruba Networks was cited as insecure by a US security watchdog group.
 
According to warning bulletins issued by the US CERT, the United States Computer Emergency Readiness Team which operates in conjunction with the Department of Homeland Security, Aruba Networks WLAN systems have several security vulnerabilities rated as "high severity" including those that allow "remote attackers to cause a denial of service (crash) and possibly execute arbitrary code," and attacks that allow "remote attackers to gain access to the WLAN or administration interface." Overall, Aruba Networks was cited for five security violations in just the last twelve months versus zero for Trapeze Networks. Details can be found on the US CERT website at: http://www.us-cert.gov .
 
The new MP-422 integrates seamlessly with the Trapeze RingMaster management suite, long recognized as the leading software system for managing large enterprise Wi-Fi deployments. As a true enterprise management tool, Ring Master provides a single view of the entire wireless network from the management console. RingMaster is the only management tool that enables large organizations to perform RF planning, client and rogue location, configuration, verification, monitoring, and reporting of the entire Wi-Fi network all from a single management console.
 
RingMaster features Virtual Site Survey which automates coverage, capacity, and voice planning for both indoor and outdoor areas. By contrast, Aruba and Cisco's software show only pieces of the network and have limited functionality. In the same report that established Trapeze as the leader in Wi-Fi security, ABI also said, "Ring Master remains the gold standard in this area (wireless network management) with its tightly integrated RingMaster software."
 
In head to head comparison tests created by the independent research firm VeriWave, the MP-422 with Smart Mobile technology significantly outperformed APs by Aruba Networks, Trapeze's closest competitor. In a 25 AP test designed to simulate real world large enterprise Wi-Fi networking conditions of 500 clients, VeriWave found that the MP-422 performed significantly better in throughput, latency and jitter, making Smart Mobile technology the most scalable, best choice for large deployments. 
 
At the core of Smart Mobile is a breakthrough architecture that combines the benefits of centralized control and management with the performance efficiencies of optimized traffic through intelligent switching. Better Range and Valuable New Features The MP-422 Access Point introduces significant improvement in range at all data rates, due to higher output power and an improved antenna design resulting in greater than 40% improvement in rage over prior generations of access points.
 
The MP-422 also features wireless mesh networking, and is draft 802.11s compliant in its implementation. Mesh networking enables the wireless distribution of traffic among access points and is extremely valuable in environments where wire cannot easily be run to the AP such as outdoors and across the campus.
 
Smart Mobile technology's significant advantages in security, manageability, scalability, and performance make Trapeze Networks the best overall choice for large scale enterprise Wi-Fi deployments. By contrast, Aruba Networks' technology falls far short in meeting these critical enterprise requirements.
 
Pricing and Availability
 
The MP-422 is shipping worldwide now. List price is $599.00.

Posted by Dal at 11:29 AM | | TrackBacks (0) Posted to Wireless Hardware

Canadian Yellow Pages Distributor Switches to Asterisk

For more than 90 years National Sales and Distribution Inc. in Ontario, Canada, has been serving the Yellow Pages Group as an exclusive distributor of telephone directories for Quebec and Ontario. The company sells advertising and other directory products primarily through a 20-person call center. As the company’s Nortel Norstar PBX began to show its age, Allan Kobelansky, a network consultant to the company, began to search for options to replace the system with an IP PBX.
He wanted a system that would allow transparent migration for the call center agents and reduce the company’s total monthly operational expenditures. When he completed his analysis, most of the options were beyond the economic reach of his client. New handsets, wiring, upgraded LAN gear, disruption, and retraining proved expensive and cumbersome.
 
Based on its attractive low-investment and robust functionality, Kobelansky chose an Asterisk open-source IP PBX. But new handsets and other associated expenses were still an inhibitor. “I looked at the cost of new IP phones and saw it would be a significant expense,” he said. “What I really wanted was a device to connect the existing Norstar phones into the Asterisk IP PBX.
 
I searched the web and came across Citel’s Portico TVA. It was exactly the solution I was looking for.” National Sales and Distribution Inc. purchased two Citel Portico TVA (Telephone VoIP Adapter, formerly branded Handset Gateway) units, which were installed after the office closed on a Friday evening. “It took me about an hour to connect the device. Installation was incredibly simple and, quite frankly, a joy,” said Kobelansky. “Configuring the Portico TVA via the intuitive web interface was also extremely easy.”
 
No End User Training Required
 
“When the call center employees arrived for work Monday, everything was the same to them except for a slight change to the phone display,” continued Kobelansky. “No one even noticed I’d made the change to an Asterisk platform – it was completely transparent to the employees.
 
We literally didn’t have to do any end user training.” Kobelansky estimates new IP phones comparable in quality to the Norstar handsets would have cost about $200-$250 per unit. And total deployment costs, including new LAN switching gear, wiring, and retraining would have added substantially more to the “per port” price of the deployment.
 
“Portico really is about making the finance group happy by extending the amortization of your existing phones,” said Kobelansky.
 
The Bottom Line
 
“I’m a telecommunications engineer and this is one of those products that you dream about developing yourself,” said Kobelansky. “This product is exactly what I was looking for and it does everything as advertised. You literally could set up a call center overnight with this technology.”
 
Source: Telephony World 

Posted by Dal at 10:38 AM | | TrackBacks (0) Posted to Asterisk Case Studies

March 12, 2007

Cisco Launches 'Subnetting Game' to Help Students

Cisco today announced the availability of "The Subnetting Game," a new educational game designed to help make the highly technical topic of subnetting easier for students to learn. The game was developed in response to a recent survey by Cisco in which a majority of students responded that network addressing, and particularly subnetting, remains one of the most difficult skills to learn.
"The Subnetting Game" is available to anyone who registers at the CCNA Prep Center, a website dedicated to networking professionals working toward completion of the CCNA certification: http://www.cisco.com/go/prepcenter.

A foundational skill for networking professionals, subnetting is the process of dividing an organization's growing network into separate subnetworks to provide for better manageability, efficiency and security. These subnetworks can be distributed to separate departments throughout the organization for unique access points, resources and levels of security.

The immersive game allows students to test their subnetting knowledge while developing the necessary skill sets through an entertaining challenge. The game begins with a small network and a simple problem, then becomes increasingly difficult as the levels progress so that students with even a minimal knowledge of subnetting can utilize the learning tool.

"Network addressing and subnetting are crucial skills for networking professionals to master," said Jerry Bush, program manager at Cisco. "With the new 'Subnetting Game,' students quickly increase their subnetting expertise while having fun. Most students find they are learning new skills without even realizing it."

Source: Cisco 

Posted by Dal at 09:41 AM | | TrackBacks (0) Posted to Cisco

Digium Preps Open-Source VoIP Appliance

Open-source VoIP vendor Digium this week unveiled its first formal channel program, a move that foreshadows its forthcoming launch of a line of channel-friendly appliances.

Digium, the primary developer of the Asterisk open-source IP-PBX platform, is about 30 days to 45 days away from launching its first VoIP box for end users, said Steve Harvey, vice president of worldwide sales at Digium, during an interview with CRN this week at the VoiceCon Spring 2007 conference in Orlando, Fla. The company currently offers an appliance-based developer kit.

Digium channel partners have to do a lot of integration work to build an Asterisk IP-PBX from scratch, using their Linux expertise to build solutions that tie off-the-shelf hardware, the Asterisk software and Digium's network interface cards together, Harvey said. Currently 85 percent of the vendor's business comes from sales of interface cards, which connect systems built on the Asterisk platform back to the public telephone network for dial-tone, he said.

"There is a whole group of resellers who love to add value through their Linux expertise and integration expertise and who are willing to do that level of integration. That market is growing," Harvey said. "But most [SMB] customers don't want to mess around to get their hands that dirty to make a phone system work," Harvey said.

That's why the Huntsville, Ala.-based company is planning the roll-out of an appliance family that should prove attractive to a larger number of customers and channel partners, Harvey said.

"The most important thing we'll be getting out of the program is more sales support," said Chad Agate, co-founder of NeoPhonetics, formerly known as SIPbox, a Digium channel partner in Tinley Park, Ill.

The new channel program splits partners into three categories: Authorized Resellers, Premier Authorized Resellers and Elite Authorized Resellers, based on sales volumes. Digium offers program members a starter kit that includes a demo kit, sales collateral and tools. The vendor is also offering training and certification for solution providers that want to become Digium-Certified Asterisk Professionals, a required designation to sell the forthcoming appliances.

Click Here to Continue Reading 

Posted by Dal at 09:24 AM | | TrackBacks (0) Posted to Asterisk Hardware

March 08, 2007

Clearwire Nets $600 Million in IPO

Clearwire, the company that is building WiMax, a new wireless technology boosts the performance of wireless broadband, has raised $600 million in its initial public offering. It begins trading today.

Clearwire, of Kirkland, Wash., reported a net loss of $240 million last year, largely because it is building out its WiMax network, and has yet to start selling it. It remains a big risk. Sprint-Nextel plans to roll out a competing WiMax network next year, and there are many other competing offerings outside of WiMax.

WiMax differs from WiFi in that it has much longer ranges — as much as 10 miles vs. WiFi’s reach of a few hundred feet.

The IPO priced this morning at $25 a share, at the top of its planned range. The IPO is signficiant because the stock markets have been jittery lately, the Nasdaq having lost more than 6 percent of its worth since WiMax filed its IPO papers. Telecom IPOs have been rare. Clearwire’s biggest backers are Motorola ($300 million invested) and Intel ($600 million invested).

Source: Personal Bee 

Posted by Dal at 04:18 PM | | TrackBacks (0) Posted to WiMax

VoiceCon: Microsoft to the VoIP industry: We're Here

Note:  VoIP Industry to Microsoft: So?  I am not trying to give Microsoft a hard time and not trying to say that Microsoft couldn't produce a quality product (take xbox 360).  I am just saying VoIP has been here for some time and it is not news that it is going to be bigger in the coming future. 

Microsoft introduced the public beta version of Microsoft Office Communications Server 2007 at VoiceCon. Following the announcement, Gurdeep Singh Pall, corporate vice president, of Microsoft's Unified Communications Group, discussed with Network World Senior Editor Phil Hochmuth how Microsoft's SIP-based VoIP, messaging and collaboration server fits in, and competes, in the enterprise convergence market.

How are you presenting Office Communications Server 2007 to enterprise voice managers and IT managers?

The key message is to see what it can do for you and see what the limitations they may have. We see a lot of folks going down these one-way streets [with PBX and IP telephony vendors]. They might find themselves in a situation where they've deployed a solution, and because it is not an open solution, it is slow in terms of innovation.

Are you talking IP PBX products from companies such as Avaya, Cisco, Nortel, and so forth?

Yes. These are closed systems. They're just like mainframes. Once you bought the computer, or IP PBX in this case, you pretty much every component you buy from that vendor. They'll tell you about openness, and say "you can buy any SIP phone, sure," but when you call product support, they'll tell you, "sorry, if you're not using their phone, we can't guarantee the voice experience." It sort of builds on the fear that voiced cannot be delivered in an open platform.

Our approach to building a solution was we didn’t try to look at it that way. We didn't go back and say here is a list of 300 features on a PBX, and that we need to start matching each one of them. We looked at what people want from their communications systems. For example, IP PBXs today have all these features, but if you ask a user to do anything more than answer a call, or add a third person into a call, is very hard. Most users have unmet needs today.

Many of IP PBX vendors at VoiceCon are calling Microsoft a partner. Is OCS a complementary product, or a competitive product for these companies?

Enterprises which have a TDM PBX today and are looking to move to an IP solution. Then you have some enterprises which have some TDM PBXs and some IP PBX and their goal is to replace all of their TDM PBXs with IP PBXs. What we are telling both groups of users is that we believe, over time, you can be totally based on Office Communications Server. For now, we also want to help customers deal with missing features they may not have, or to help along those who are saying, 'oh, can I trust my voice entirely to Microsoft.' They can keep their current system in place, and put Office Communicator next to it, and slowly phase out the old one.

This resonates with customers, but what about the partners? I'm a big believer in the force of the customer. If customers are educated and aware and they know what they want, they will make the right choices. If there is merit to our approach, then the partners who are in the [IP telephony market] today will have to transform themselves, similar to the way IBM transformed itself form a mainframe company to a great services company. They will provide what the customer is asking right now, which is interoperability with OCS. Over time, they will figure out how to create a good business in this new market. Nortel certainly has joined with us to do that. The question is, will other players do it, or will they push their vertically integrated stack.

Click Here to Continue the Interview 

Posted by Dal at 03:48 PM | | TrackBacks (0) Posted to VoIP News

March 07, 2007

British Telecom to Invest in FON?

Note:  Om Malik from GigaOm sent over this link about BT possible investing in the free wifi company FON.

FON, the Spanish share-your-Wi-Fi services company, is close to announcing a new round of funding that could total to as much as 10 million Euros (shade over $13 million.)  While some of its existing investors – Index Ventures, Skype and Google - are coming back with more cash, the word from telecom circles in Europe is that British Telecom is going to invest in the wireless router company. It is one of the new strategic investors in the company.

FON had raised about $22 million in a previous round of funding. With this round the company has raised close to $35 million in venture capital funds.

FON is the latest start-up by Spanish telecom entrepreneur, Martin Varsavsky, who has made a name for being a thorn in the side of incumbents. Ah the sweet irony that is life! We have not been able to confirm the rumors so far, but will update the story with more info as it surfaces. Varsavsky declined to comment.

Click Here to Continue Reading 

Posted by Dal at 12:16 PM | | TrackBacks (0) Posted to Blogsphere News

Skype Clears 500 Million Downloads Worldwide

Skype  today announced that its software has been downloaded more than 500 million times by users around the world, extending its mark as one of the most popular free downloads of all time.  Skype makes it easy for anyone with a broadband internet connection to make free and very cheap unlimited calls worldwide.
This milestone was reached in slightly over 42 months time, counting from when Skype was first introduced in beta at the end of August 2003.

"The entire Skype team owes its success to the global community of registered users we have today who tell their loved ones how much they enjoy using Skype. We’re absolutely delighted that so many people are embracing Skype and speaking to friends or family all over the world for free or very little cost." said Skype CEO and co-founder Niklas Zennström.

A little over three years ago, Skype developed a little piece of software that allowed anyone to experience high-quality, unlimited voice communication over the Internet. Today, people are using Skype to express themselves via free voice, video, conference calling and instant messaging communication, as well as Skype’s paid-for products including Skype Pro.

Skype Pro is a new Internet communications package offering zero cents per minute calls to domestic landlines along with a series of premium Skype features and discounts on Skype Certified hardware. Connection fees apply.

Skype in Numbers:

* Six people download Skype every second
* Skype to Skype minutes in Q4 2006 alone totaled 7.6 billion minutes
* In September 2005, Skype had 54 million registered users and today it has over 171 million registered users worldwide
 
Source: Skype

Posted by Dal at 11:40 AM | | TrackBacks (0) Posted to Skype

March 06, 2007

Portable Wireless Video Streaming Solution Demonstrated at APCO

 
 
As part of the Association of Public Safety Communications Officials International (APCO) Western Region Convention, HauteSpot Networks Corporation demonstrated a new product which allows any video camera to send a standard definition resolution (740x480 NTSC) digital video stream, wirelessly, at long range, and through walls and other obstructions. The demonstration took place at the Long Beach Fire Department's Emergency Communication and Operations Center.
The demonstration showed the HR-IXPSXPi-SD wireless video streaming encoder feeding a video signal from a cameraman carrying the unit in a small bag back to the LBFD's state of the art mobile command vehicle. The video signal was then integrated with other fixed video feeds providing the incident commander with a complete picture of the entire area around the vehicle.

The HR-IXPSXPi-SD represents a new advance in communications technology. The device incorporates a high performance 4.9 GHz broadband transceiver (other unlicensed and licensed frequencies are available) with MPEG2 video and MPEG1 stereo audio encoders and a simple to use browser interface. Any analog camera can be directly attached to the HR-IXPSXPi-SD through either composite or S-Video connectors. The video is then compressed and streamed using RTP over Unicast or Multicast which is sent out wirelessly.

The wireless signal is then received by a HR-IXPSXPi unit which is attached to a wired network. The video can be received and played back through any laptop computer using freely available software, or through a low cost IPTV set top box which can be attached to a standard definition television, VCR or DVR.

The operating range of the HR-IXPSXPi-SD is approximately 300 meters through walls, around trees or around buildings and about 2 kilometers line of sight with no obstructions, using compact multipolarized antennas. Range can be extended using larger, higher gain antennas. During the demonstration the image quality was outstanding and there were no artifacts or interference.

The compact size of the HR-IXPSXPi-SD allows it to be carried in a small camera bag, together with a standard 12VDC battery to power the unit. The entire system, including battery, weighs less than 3 lbs.

"The main application we were thinking about for the HR-IXPSXPi-SD," said Bob Ehlers, CEO of HauteSpot Networks, "is for first responders to provide incident assessment and video feed back to incident commanders using the new 4.9GHz broadband spectrum." "Of course the HR-IXPSXPi-SD could also be mounted into vehicles, into helicopters, or in fixed surveillance cameras," Ehlers concluded. Optional pan-tilt-zoom control is offered through the serial port of the HR-IXPSXPi-SD using the Pelco-D protocol.

The HR-IXPSXPi-SD is available in both unlicensed 900MHz, 2.4GHz and 5GHz models, as well as in the 4.9GHz public safety band. Other bands for export and government use are also available. Optional on-board mass storage is also available so the unit can be used as a digital recorder.

Pricing for the HR-IXPSXPi-SD starts at just $999.
 

Posted by Dal at 08:21 AM | | TrackBacks (0) Posted to Wireless Hardware

BlackBerry 8300 Daytona Has WiFi

 

Okay, so it might be the most reliable source, given that it is a Wordpress-hosted personal blog, but the prospect is just too juicy to pass up. Not only does Research in Motion have a new BlackBerry coming down the production chute, they have the evaluation unit out and it's got built-in WiFi. Ladies and gentlemen, may I present to you the BlackBerry 8300, also known in some circles as the "Daytona".

I'm not sure how the Nascar reference comes into play, but it might be that blazingly fast wireless transfer speed you'll enjoy with the WiFi-ness found within.
 
The rest of the specs are still up in the air, but the current rumblings indicate that it has a QWERTY keyboard (as you can see from the picture), a 2-megapixel camera with LED flash, GPS, improved speakerphone audio, and a multi-touch display like the iPhone. Okay, that last part is just wishful thinking on my part.
 
Source: MentesDigital 

Posted by Dal at 08:12 AM | | TrackBacks (0) Posted to Wireless Hardware

March 05, 2007

Clearwire: Taking WiMAX to "The Street"

 
 
On Mar. 6, Clearwire, the company headed by wireless pioneer Craig McCaw, is expected to sell shares in what could be one of the most talked about—and sought-after—tech initial public offerings of the year. The Kirkland (Wash.) company, which provides services through the wireless broadband technology known as WiMAX, plans to offer up to 23 million shares, at $23 to $25 each.
When it debuts, the stock may fetch more, says Scott Sweet, managing partner at consultancy IPOBoutique.com. The shares are likely to debut at $25 to $27, fetching Clearwire as much as $621 million, compared with the $513 million the company initially expected, Sweet estimates. "There's enormous demand," despite a recent equity slump that analysts believe could roil IPO markets, Sweet says.
 
"Unless the market absolutely crumbles, Clearwire is ready to go," he adds. Trusted Pioneer Part of the optimism is based on the track record of McCaw, who more than a decade ago stitched together the first nationwide cellular empire, and then sold it to AT&T for $11.5 billion in 1994. McCaw is hoping to re-create such successes with Clearwire, which has amassed the second-largest chunk of the airwaves best suited to WiMAX services after Sprint Nextel.
 
In February, the company shelled out $300 million for an additional swath of that prime spectrum from AT&T. "McCaw knows what he is doing," says Philip Solis, principal analyst in mobile broadband with consultancy ABI Research. "Spectrum is the most valuable asset in wireless."
 
 

Posted by Dal at 12:42 PM | | TrackBacks (0) Posted to WiMax

Digium Launches "Authorized Reseller Program" for Asterisk PBX Platform

 
 
Digium®, Inc., the Asterisk® company, today announced the first formal reseller program for the Asterisk community. The Digium Authorized Reseller Program is designed to provide new and existing resellers with access to cutting edge Asterisk open source telephony solutions and attractive incentives for bringing the proven solution to new markets.

“Digium has a track record of being very channel centric, with more than 90 percent of our business flowing through the channel,” said Steve Harvey, vice president of worldwide sales. “As end user interest in Asterisk continues to grow, we must build a network of regionally based resellers capable of installing and maintaining open source IP telephony solutions. The Digium Authorized Reseller Program is designed to provide resellers with the tools they need to excel in the market, with particular emphasis given to reseller profitability.”

The cost for an Asterisk solution is typically one-half to one-third the cost of a traditional legacy system. As part of the program, Digium will assist resellers in promoting, marketing and selling Digium’s products, solutions and services.

Approved resellers will be able to sell Digium’s hardware and software solutions including all telephony interface boards, Asterisk Business Edition™ (Digium’s professional grade version of Asterisk), and the Asterisk Appliance Developer Kit, which provides early access to the upcoming Asterisk Appliance for value added resellers to develop their own Asterisk-based solutions.

Digium’s Asterisk solutions also enable VARs to install hybrid VoIP solutions, allowing the use of legacy phone equipment when deploying a VoIP solution. Hybrid solution alternatives are especially important for small- and medium-sized businesses looking to decrease the initial, up-front cost of equipment when deploying a VoIP solution.

“We joined the Digium Authorized Reseller Program to broaden our VoIP practice and support the most widely used open source PBX on the planet,” said Michael Tanenhaus, president and CEO at Mavenspire, an IT consulting, sales and hosted technology services company. “Being a Digium reseller has opened opportunities for integration and bundling that are simply not available with proprietary products.”

The Digium Authorized Reseller Program offers resellers three program levels to choose from depending on anticipated sales levels:

  • Authorized Reseller (up to $120K annually)
  • Premier Authorized Reseller (up to $500K annually)
  • Elite Authorized Reseller (over $500K annually)

Resellers can register to join the Digium program by completing an online application at www.digium.com. Once a reseller joins the program, they will have an opportunity to purchase the Digium Reseller Starter Kit which includes a demo kit, sales collateral and access to reseller tools. With entrance into the program the reseller also has six months to complete the Digium Asterisk Boot Camp to become a Digium Certified Asterisk Professional.

Posted by Dal at 12:00 PM | | TrackBacks (0) Posted to Asterisk News

March 03, 2007

SIP Based Paging Server Using VoIP Released

CyberData announces its new SIP-based Paging Server. The Paging Server is a SIP-endpoint that multicasts out to CyberData VoIP speakers. Because the vast majority of IP-PBXs are SIP enabled, the Paging Server allows IP-based zone paging on nearly all VoIP platforms.
To conserve network bandwidth and resources, the Paging Server is configured through its web interface with unique multicast address and port number combinations that represent specific paging zones. Paging to different buildings or organization-wide announcements are now possible anywhere on your network or through the Internet with the CyberData Paging Server.
 
 

Posted by Dal at 01:52 PM | | TrackBacks (0) Posted to VoIP Hardware

Asterisk PBX 1.4.1 Released

The Asterisk and Zaptel development teams have released Asterisk 1.4.1.

This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability.
It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety of phones and provides much more flexibility and configurability (along with actually working properly in most scenarios, which the original implementation failed to do).
 
Users who are interested in SLA functionality should update to this version and try it out; we welcome bug reports and test reports. Because of the security vulnerability fix present in this version, all users of Asterisk 1.4 are urged to update as soon as they can schedule it.
 
 
Thanks for your support of Asterisk and Zaptel!

Posted by Dal at 01:42 PM | | TrackBacks (0) Posted to Asterisk Development

March 02, 2007

FCC Clarifies VoIP-PSTN Interconnection Rules

Note:  Can I say w00t?  Choice is Power.

 

The FCC granted a petition from Time Warner petition that clarifies rules of how voice traffic can be exchanged between broadband providers and the PSTN. By granting the petition, the FCC affirmed that competitive local exchange carriers (CLECs) are entitled to interconnect with incumbent local exchange carriers (ILECs) pursuant to section 251 of the Telecommunications Act for the purpose of exchanging traffic on behalf of VoIP-based service providers.

FCC Chairman Kevin Martin stated: "Our decision will enhance consumers' choice for phone service by making clear that cable and other VoIP providers must be able to use local phone numbers and be allowed to put calls through to other phone networks."
 
Jim Kohlenberger, Executive Director of VON Coalition, stated "The VON Coalition applauds today’s FCC decision to answer the call for voice competition by ensuring that voice traffic can be exchanged between broadband and telephone networks. The FCC was right to find that states cannot unilaterally twist the meaning of the Telecom Act to hang up on broadband enabled voice competition.
 
Today’s decision helps ensure that millions more American consumers can take advantage of power and potential of Internet based communications. The problems that Time Warner faced in South Carolina and Nebraska were not isolated, they threatened the competitive availability of VoIP services overall, and they effectively created “digital divides” between those Americans who can enjoy the full potential of broadband and those who cannot.”
 
 

Posted by Dal at 08:56 AM | | TrackBacks (0) Posted to VoIP Politics

Web-Call-Back Module the Integrates Joomla CMS and Asterisk PBX

Web-Call-Back is the module for Joomla! that allows, in cooperation with a Asterisk PBX server, to perform a call back to a visitors provided phone number directly from your website. To make use of this module a Asterisk PBX server or compatible software PBX is required. The modules settings allow fully customization of the messages, in multiple languages with JoomFish installed, and styling via CSS.

It requires JavaScript on the client browsers side as it performs its work in client-server communication with the server side module in Joomla.
 
As overview Web-Call-Back allows:
 
* To enter any international phone number on the frontend

* Initiation of the call-back, either by internal extension first or outbound call with transfer to internal extension

* Fully customizeable messages for phone number form and status messages of the module

* Date & Time rules to be defined to show the module only during office hours or any specified period of times

* Show messages or hide the module when the internal extension or the PBX system is offline and the call back is unable to get performed.

* Output of debug information on communication with the Asterisk PBX.

Prerequisite for this module to perform its functions is a Asterisk PBX system for call initiation and transfer. This module requires knowledge and ability to configure a Asterisk PBX system, without these individual prerequisites this module will become difficult to startup. Support for the module is limited to the module itself, any issue that is dealing with Asterisk will be the users self responsibility.

Download:  Click Here 

Source: Joomla Feed 

Posted by Dal at 08:45 AM | | TrackBacks (0) Posted to Asterisk Software

Live from Etel - The Future of Asterisk

Note:  VoiP-News has a nice write from Mark Spencer's address to the E-Tel conference audience.

"We’re welcoming Mark Spencer, the “Father of Asterisk.”  In Japan, he says, it’s customary to begin with an apology.  He apologizes for the presentation graphics, and also for app_voicemail.  At the time he wrote it, it seemed like a simple concept, he says.  He also apologizes for app_queue/chan_agent." 

He says it’s great to get our concepts and projects “out of the box” but sometimes we just get ourselves into another box.

One of the important things he wants to do is build bridges between the community of developers with end users and customers.  He wants to make it easier to build, configure, and customize it so that it’s accessible to those who aren’t experts in the technology.

Spencer has hired a new CEO for Digium so he can focus on software and community relationship development.  He’s also expanding developer programs and the core development team.  A goal is to reach out to end users with improved graphical configuration, that doesn’t cover up or hide asterisk, and improved desktop experience.

One audience question is whether they’ve considered integrating voice recognition.  The answer is yes - in v.1.4 it’s included.  It integrates with Sphinx.

Another is, what is the fate of the asterisk device?  It’s available today with 4 ports of LAN, 1 WAN, same interface as PCI cards, compaq flash.  It’s been released as a developer kit and will eventually be a final customer product for end user purchase.

Click Here to Continue Reading 

Posted by Dal at 08:41 AM | | TrackBacks (0) Posted to Asterisk News

March 01, 2007

Introducing Nerd Vittles Phone Genie for Asterisk

Excerpt: Today we introduce an all-purpose web utility that lets you reconfigure virtually anything and everything on your Asterisk system on the fly using an incredibly powerful HTML command language that you can master in a matter of minutes.
All you'll need is a web browser including most cell phone web browsers as well. And, of course, it's free. What the Nerd Vittles' Phone Genie does is provide a web interface to the complete Asterisk Manager API. If you also happen to be using the Asterisk Management Portal or freePBX, then the sky's the limit since there are dozens of functions you can execute by sending web commands to manipulate data stored in Asterisk's internal database. These functions include call forwarding, call waiting, call tracing, do not disturb, and many more.
 
 

Posted by Dal at 09:15 AM | | TrackBacks (0) Posted to Asterisk Help

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