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January 31, 2007

Shipments of Wi-Fi Meshing Access Points to Double by 2010

Wi-Fi Meshing, which allows wireless access nodes to achieve a longer range by using each other as repeaters, is becoming increasingly common in several markets.

Shipments of Wi-Fi mesh AP is expected to grow from 50,000 units in 2006 to almost 100,000 units in 2010. However, In-Stat predicts that incompatibility among systems may be a potential hurdle for the growth of the technology.

The market research firm also reports that although Tropos has the largest mindshare in municipal mesh networking, Nortel, Strix, BelAir, and SkyPilot made aggressive pushes into the market in 2006. Cisco is perceived as a wildcard in this market, as it just launched Wi-Fi Mesh Access Points in late 2005, but it is using its strong IT channels into businesses and government networks. Throughout 2006, most Wi-Fi mesh vendors tied their growth to municipal network build-outs.

Source:  Global Sources 

Posted by Dal at 07:58 PM | | TrackBacks (0) Posted to Wireless Hardware

Skype 2.5 For Mac OS X is Ready For Download

 
 
Note:  I actually prefer Mac Skype over iChat because of Skype's better NAT traversal ability. 
 
Skype finally released the new version of their free voip client application for Mac OS X users: Skype 2.5. 

The beta’s been around since mid-November, but the final version was officially released today. And aside from a slightly updated look, there’s a new birthday reminder feature and the abilities to send SMS messages to any mobile phone in the world and hold conference calls with up to nine other people have been added.

Skype 2.5 for Mac

Source: CrunchGear 

Posted by Dal at 07:42 PM | | TrackBacks (0) Posted to Skype

Coming Soon: More Big Brother to Love

Note: I found this article in the Huffington Post today about some new revisions into the eavesdropping rules allowing the different agency to"tap" into data streams and voice IP packets.  I do believe we should have some basic privacy even if it means we are a little less safe.  The key is to respect each other and we will get along fine.
 
"Over the past several months, the FCC and Justice Department have been working overtime, and fighting hard to tap not only your landline phone, your cellphone, but to tap Internet phonecalls, as well.  Effective in May, those who provide "voice transmission," and broadband services will have to ensure that their equipment that is wiretap-ready, and accessible to your local police force, and the FBI." 
The new legislation is modeled after the 1994 Communications Assistance for Law Enforcement, or CALEA, which was designed primarily to facilitate wiretaping of mobile phones. This new legislation is intended to expand governmental surveillance powers to cover companies like Vonage, so the progression evolves thus: first we can tap Ma Bell, then Cingular Wireless, then Yahoo emails, then Vonage.
 
The rules set to go into effect in a couple of months have been challenged by a U.S. appeals panel, back in July, at which U.S. District Judge Harry T. Edwards called courtroom arguments made by the FCC "goobledygook." (VoIP News Net) He was, in my opinion, being kind. Civil liberties groups have expressed outrage over the FCC expansionism claiming that this legislation doesn't take into account the fundamental difference between the telephone, a vehicle for conversation, and the Internet, a tool by which information is acquired and conveyed.
 
Lawyers for the government argued only that the 1994 intended to be applied to future technology; the Judge wasn't buying that, and neither are we. Moreover, sophistic claims by the Justice Department that not increasing wiretapping capability to encompass the rapidly proliferating Internet phone industry will transform the Web into a refuge for "criminals and terrorists" are not only hackneyed, they're transparent enough for a six year old to see through.
 
 

Posted by Dal at 02:18 PM | | TrackBacks (0) Posted to VoIP Politics

Adtran Execs Defect To Digium, Plan Channel Push

Note:   This should help their management team and hardware support and integration.  Now we need some high profile security and data backup execs :P 
 
Open Source VoIP vendor Digium on Tuesday named a new CEO and a new vice president of worldwide sales, bringing on two former Adtran executives to fill the roles. Danny Windham, former president, COO and director of networking vendor Adtran, is joining Digium as its chief executive. Huntsville, Ala.-based Digium is the creator and primary developer of the Asterisk open-source VoIP platform.
Steven Harvey, former vice president of enterprise networks and competitive service provider sales at Adtran, has been named vice president of worldwide sales at Digium. With the executive additions, Digium Founder and President Mark Spencer is taking on the newly created position of chairman and CTO.
 
Windham, who has been a member of the Digium board for the last seven years, will be responsible for Digium's corporate strategy and day-to-day operations. A 16-year Adtran veteran, he served as president and COO since September 2005. He was named to Adtran's board a year ago, a seat he is now vacating. Harvey will drive Digium's channel strategy and business development activities.
 
He has overseen Adtran's channel partner program for nine of his 11 years at the company, ushering in recent updates to the vendor's Advantage Partner Program, including the launch of a deal registration program, the rollout of a new partner relationship management system and a more than $1 million investment to build a channel telesales group. Harvey said he and Windham are moving to Digium as part of a plan to build a multitiered channel, much as they did together at Adtran.
 
Harvey said he plans on working with the channel team at Digium to craft the company's channel program. When they began working together at Adtran, the company was primarily a direct-sales culture within the enterprise business, which the duo moved to a 95 percent channel sales model with $125 million in enterprise sales for 2006.
 
With the planned release this quarter of a new Asterisk hardware appliance for small businesses, Digium is poised to appeal to a group of VARs that might not have shown interest in Asterisk historically because of the difficulty in configuring and supporting the product.
 
"We're simplifying the product into one discreet unit, helping VARs cross the chasm of voice/data convergence," Harvey said. "This helps them simplify their convergence challenges." Adtran and Digium already have strong ties.
 
In addition to Windham's seat on Digium's board, Adtran holds an equity interest in Digium, and Spencer formerly worked as a co-op student at Adtran, also based in Huntsville. Harvey is being replaced at Adtran by Ted Cole, who has been named vice president of channel sales. Adtran will not be replacing Windham, according to a company spokesperson.
 
Source:  CRN 

Posted by Dal at 01:41 PM | | TrackBacks (0) Posted to Asterisk Development

Nelson, Clinton, Snowe Re-Introduce Voice Over IP (VoIP) E-911 Legislation

Senator Bill Nelson (D-FL); Senator Hillary Rodham Clinton (D-NY), Co-Chair of the Congressional E-911 Caucus and Senator Olympia Snowe (R-ME) today announced that they have reintroduced the IP Enabled Voice Communications and Public Safety Act. The bill addresses the need to ensure the growing number of Voice Over Internet Protocol (VoIP) telephone service subscribers have full access to 911, including Enhanced (E)-911 capability that allows 911 dispatchers to trace the phone number and location of calls for help.
Unfortunately, we ve seen the tragic consequences when consumers can t connect to 911 services through their Internet phone company, Senator Nelson said. VoIP subscribers should feel confident that they will have access to emergency services it could be a matter of life or death. It is critical that the millions of households using this technology can reach 911 when tragedy strikes.
 
All emergency calls, whether made on a land line, cell phone or Internet-based phone service, need a rapid response. It could truly make the difference in saving a life, said Senator Clinton. The inability of the emergency response network to keep pace with voice over Internet protocol technology has left millions of VoIP subscribers without guaranteed access to emergency services, Senator Snowe said.
 
Innovation and technological advances should improve the lives of Americans, not endanger them. VoIP subscribers should not be susceptible to substandard emergency service simply because they are on the cutting edge of in home telecommunications technology. VoIP telephone customers are connected to broadband internet lines instead of traditional phone lines. Ensuring that 911 calls made from VoIP phones are properly routed and responded to has presented new challenges to public safety officials.
 
There have been several tragedies in which VoIP 911 calls were either routed to closed business offices instead of emergency dispatcher or could not be connected. The Clinton-Snowe-Nelson bill will allow VoIP companies to patch into the 911 networks operated by the traditional phone companies. The bill also ensures that consumers are fully informed if their VoIP provider cannot ensure that their 911 call will be properly routed in an emergency.
 
Furthermore, the legislation tasks the National E-911 Implementation Coordination Office -- created under the ENHANCE Act introduced by Senator Clinton and signed into law in 2004 -- to develop a plan for a nationwide network and make recommendations to Congress in order to ensure that all 911 VoIP calls are responded to properly.

Posted by Dal at 01:07 PM | | TrackBacks (0) Posted to VoIP Politics

January 30, 2007

AudioCodes and FaxBack Partner to Deliver Best-in-Class IP Fax Solutions

AudioCodes, a leading provider of Voice over Packet (VoP) technologies and voice network products, and FaxBack, developer of NET SatisFAXtion IP, a T.38 VoIP fax solution, today announced that they have completed interoperability testing between NET SatisFAXtion 8.1 and AudioCodes TrunkPack PCI and cPCI form factor SIP media gateway products.

By combining NET SatisFAXtion and AudioCodes SIP media gateways, designers can build high performance and scalable fax solutions for applications that need to retain their existing PSTN infrastructure and avoid purchasing expensive dedicated fax blades. With this combined solution, a broad range of enterprise and service provider applications can be deployed leveraging industry-standard T.38 and SIP protocols.

Service provider applications as large as multiple DS3s can be implemented using the AudioCodes TP-6310/SIP cPCI media gateway blade, while at the other end of the spectrum enterprise applications as small as a single fractional T1 can be configured using the TP-260/SIP PCI media gateway. Our partnership with FaxBack enables a new range of high-performance and scalable fax offerings to customers, says Moshe Tal, General Manager of AudioCodes Blade Business Line.

Integrating our T.38-enabled media gateway products will provide industry leading AudioCodes media gateway performance along with the powerful FaxBack application platform. Implementing the NET SatisFAXtion IP and AudioCodes solution helps companies complete their business communication strategy beyond voice and data by easily adding fax capability to IP environments, leveraging investments in converged networks, says Mike Oliszewski, CTO of FaxBack. Whether organizations seek to eliminate fax machines, automate and track inbound fax traffic or integrate fax with core business applications such as email, NET SatisFAXtion IP and AudioCodes media gateways are a cost-effective and reliable choice.

AudioCodes TrunkPack line of SIP media gateways, including the TP-260, TP-1610 and TP-6310, offer OEMs and value added resellers both PCI and cPCI digital media gateway platforms for enhanced voice services and enterprise applications. Offered in densities from one E1/T1 on a single PCI slot up to a full OC3 on one cPCI blade, the TrunkPack line is an excellent building block for a diverse range of applications including fax solutions. The TrunkPack line supports a range of central office and PBX TDM protocols, including ISDN, CAS and E1R2 signaling.

Leveraging an on-board SIP stack, the TrunkPack line eliminates the need for PCI drivers and associated operating system compatibility issues. NET SatisFAXtion IP leverages converged network investments and, unlike traditional fax server solutions, do not require expensive fax blade hardware, complex configurations and ongoing support costs. This software streamlines and automates the faxing process, allowing inbound and outbound fax routing from the desktop while increasing employee productivity. Installation is quick and easy using the web-based LaunchPad that is installed on the fax server system.

Source: CRM2Day 

Posted by Dal at 10:12 AM | | TrackBacks (0) Posted to VoIP News

Danny Windham to Replace Marc Spencer as Digium CEO

According to Alec Saunders, Mark Spencer the Founder of Digium and creator of Asterisk, has stepped down as CEO of Digium. Adtran’s Danny Windham will be assuming the reins at Digium in Mid-February. According to Nufone’s Jeremy McNamara, Mark Spencer will remain Chairman of the Digium Board, and also take on the role of CTO.

This is interesting news and further signs that Digium and Asterisk is positioning itself to take on the enterprise PBX space. For month’s they have been filling out their executive management team, and with Danny coming on board, it looks like this has been completed.

This is excellent news for everyone involved. It is scary to think what will come out of Digium now that Marc Spencer has been freed-up to focus on the development of Asterisk and other ancillary technology….

Source: Smith on VoIP 

Posted by Dal at 10:07 AM | | TrackBacks (0) Posted to Asterisk News

VoIPowering Your Office: Powering Call Centers

Note: VoIP Planet has another great article this time covering Call Centers and how to setup on up on an extreme budget.
 
"Annoying pests who invade your home. Inmates memorizing your credit card numbers and home address. Customer service reps who do not speak your language, or quite possibly any human language. Long hold times. Long annoying commercials instead of pleasant hold music. Byzantine call-routing menus designed to make you go away. Voice-stress analyzers that reward yelling and swearing." 

I doubt you'll find many people with positive things to say about call centers. And why should they? The poor things are misused and abused in all kinds of ways. But it's not the fault of the technology. Which, like all telephony gear, has long been overpriced and under-featured. Voice over IP has revolutionized call centers—just as it has practically everything else in the world. Can you build a call center on free software? Yes, you can. Call centers fall into two general categories: customer service, and annoying phone spammers. Customer service call centers typically handle tasks like:

* Taking orders

* Fixing problems

* Providing information and general assistance

Those are the functions that many businesses need—someone to answer the phone and be helpful. Perhaps these should be called answer centers instead of call centers. Then there are the call centers that are literally call centers—these are the folks that pester us as though we were paying for our phone service just so we could serve as extensions of their marketing. It's rather amusing browsing publications that target this type of call center.

They use benign phrases like "outbound dialing," "predictive dialing," "customer care," "marketing relationships," "interactive intelligence," and "text-to-speech systems for totally automated collections". Now that's progress—completely eliminating the humans. You can complete the cycle by setting up an Asterisk server at home to talk to the telemarketer's telephony server, and never have to touch a telephone yourself. Not all phone spammers are really spammers, of course.

Volunteer organizations benefit from using automated dialers to remind members of meeting dates and other events. Businesses that offer genuine opt-in for certain services might as well reap the benefits of automation as well. For example, my bank calls me when they have specials on things I'm interested in. That is a good thing. Not like some businesses that elevate a trivial one-time purchase into a lifelong intimate relationship. Read my lips: OPT-IN.

Click Here to Continue Reading 

 

Posted by Dal at 09:44 AM | | TrackBacks (0) Posted to Blogsphere News

January 29, 2007

VoIP security: Scenarios, challenges, and counter measures

VoIP combines the worst security vulnerabilities of IP networks and voice networks. This article discusses vulnerabilities, challenges and countermeasures in securing a VoIP network from the application right down to the hardware.

Spoofing

Spoofing poses another level of challenge for VoIP that is creation of TCP/IP packets using someone else's IP address. Hackers use a variety of techniques to find an IP address of a trusted host and then modify the packet header (Source IP address field) so that it appears that the packets are coming from that host, a technique popularly called as Caller ID Spoofing in VoIP domain. Pranks on friends and loved ones are the most common application of spoofing.

Websites such as: Spoofcard, Nufone, and Spooftelprovide caller ID spoofing services, and eliminate the need for special hardware. Caller ID spoofing is often used by those who bug stolen credit card numbers. They will call a service such as Western Union, setting Caller ID to appear to originate from the card holder's home, and use the credit card number to order cash transfers that they then pick up. Exposing a similar vulnerability, Caller ID is used by credit-card companies to authenticate newly issued cards. The recipients are generally asked to call from their home phones to activate their cards.

In August, Secure Science Corporation warned that hackers can use Caller ID spoofing to break into voice mail boxes of T-Mobile subscribers. A U.S. wireless company with 15.4 million customers, T-Mobile permits users to check voice mail without entering a passcode, as long as they're calling from their own phone--an easy matter to fake with caller I.D. spoofing.

Caller ID Spoofing and SPIT are threats that are one or the other form of more generic term "Man-in-the-middle" attack. This is the name given to a situation where an attacker inserts himself between the originator and recipient of the call, without either of them knowing that their communication medium has been compromised. To either participant in the call, the attacker appears as the other, intended participant. Thus the attacker can intercept, modify and insert messages in the conversation. Obvious consequences include loss of confidential information and changing the meaning of the information conveyed.

Call hijacking is a form of the man-in-the-middle attack in which the attacker replaces one of the participants in the call. Such attacks can be accomplished in a variety of ways. One, is the manipulation of registration records maintained by the registrar/proxy server in a SIP-based VoIP network. This allows a malicious user to register as a valid user and further carry out toll fraud etc. Another means to launch such an attack is to manipulate the 3xx SIP response codes.

This allows the rogue user to redirect the voice traffic through them. There are some legal methods too, i.e., 'Footprinting' that is the easiest and safest way to go about finding information about a company that is available to the public, such as phone numbers, addresses, etc. Performing who is requests, searching through DNS tables, and scanning certain IP addresses for open ports, are other forms of open source footprinting. Most of this information is fairly easy to find, and obtaining it is legal.

Most companies post information on their website which can be very useful to hackers--and the companies don't even realize it. Footprinting this is most convenient way that hackers use to gather information about computer systems and the companies they belong to. Footprinting allows a hacker to know as much as they can about a system, its remote access capabilities, ports and services, and aspects of its security. Many administrators now post false phone numbers to protect themselves from footprinting.

Click Here to Continue Reading 

Posted by Dal at 12:25 PM | | TrackBacks (0) Posted to VoIP Security

Linksys SPA 921 IP Phone Review

Note:  I found this browsing VoIP-Info.  Personally we use a couple of the Linksys SPA-921 in the office.  For a value phone I think they are a pretty good deal.  But for a heavy phone user I would recommend something a higher quality like a Polycom 501/601 or Cisco 7960+.

" First Sipura introduced the SPA-841 which was a decent IP phone but nothing to get excited about, it looked bad and did not have a high quality feel to it when you hit the buttons, felt the handset, etc.  The phone was quite popular though with the Asterisk crowd, in large part due to its low price and I would assume the general popularity of Sipura SPA3000s within the Asterisk community." 

After Linksys bought Sipura (Linksys itself is now owned by Cisco) they re branded the next generation of Sipura products, the SPA9XX series, as Linksys products.  This series improves upon the original 841 in terms of features (Power over Ethernet on some models and a built in switch on some models are the main improvements) as well as build quality, the phones now feel like a high quality product.

The Linksys SPA921 is the cheapest of the Linksys SPA9XX series but offers the same quality as other models in the series, just not all of the same features.  For instance the SPA922 is similar to the 921 but includes Power over Ethernet (which means if you do not already have PoE compatible equipment you must purchase either a PoE injector or a power cube for the phone) as well as a built in 10/100 Ethernet switch so that if you only have a single Ethernet drop to a desk you can plug that into the phone and the computer into the phone, versus having to purchase a separate switch and plugging both the phone and the computer into the switch and the switch into the single drop.  
 
The one nice feature for a home office that the SPA922 does have that the SPA921 does not is a back light for the LCD.  If you want a back lit phone for home use you could purchase the SPA922 but keep in mind the additional cost of having to purchase a power cube or a PoE injector.  I find it very odd that Linksys would provide a back light on a PoE phone but not on a phone that plugs directly into the wall for electricity, usually it is the other way around.

All of the SPA series of products only work with Session Initiation Protocol (SIP).  SIP is of course the dominant protocol out there for both IP phones and servers but also for Internet Telephone Service Providers (ITSPs).  Personally I like to have my SIP devices all connect to Asterisk and then have Asterisk make IAX2 protocol connections out to my ITSP since not only does IAX2 support encryption (who knows how good it is, it was thrown in a few versions back but rarely used by anyone) but I also don't have to have as many ports open on my firewall.
 
 

Posted by Dal at 09:33 AM | | TrackBacks (0) Posted to VoIP Hardware

10 things you need to know about VoIP

Note:  Network World is running a piece that covers some of the areas you want to make sure and cover for any new VoIP initiative for your business.  The two areas I thought were the most important were getting people on board with the project and 911.  Remember that traditionally 911 is handle through your local carrier based on their 911 database.  When you choice a carrier make sure they have the ability to add your 911 entries into the database or hire a external company that provides this service.

"Before rolling out voice over IP in a business, it pays to tap into the lessons others have learned. Anybody working on a VoIP project should stand on the shoulders of those who have gone before to avoid their mistakes and glean tips that can make their own deployments go more smoothly." 

In the interest of promoting this knowledge sharing, here is a list of 10 tips you should follow if you want to roll out VoIP with as little pain as possible.

1. Buy time.

Even with the smoothest deployments, things don’t always happen as planned, so build a buffer into your timeline, says Lauren Johansson, IP telephony manager for MedQuist, a medical records firm in Mt. Laurel, N.J. For example, in Johansson’s case, getting an OC-3 from her carrier took an extra six months during which MedQuist had to make do with a DS-3, a lot less bandwidth than it wanted.

2. Get everybody onboard.

Make sure business-unit leaders are on the VoIP project team so they know the details and can communicate them to their employees, giving all users a stake in the project. “This reduced switchover time and made for little need for user training,” says Randy Hillman, customer care manager for Sovran Self Storage, headquartered in Buffalo, N.Y., who oversaw a Shoretel VoIP deployment.

3. Know what you’ve got.

Along with traffic, businesses need to figure out exactly what hardware makes up the network infrastructure and more important, whether it will support technology that can improve voice quality. For instance routers and switches that support virtual LANs and traffic shaping go a long way toward carving out enough reliable bandwidth to prevent degradation of VoIP connections. “If you don’t have an accurate network diagram, you can’t do a project like this,” Johansson says.
 
 

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

January 28, 2007

Bill Gates discusses the future of Web 2.0 and PBXs

Note:  I found this article about Gates recently speaking about where all this new technology is headed in the coming future.  It sounds like he does actually "get" where the PBX market is going.  Here is some bits from the article:

"Speaking to a group of business leaders at the World Economic Forum in Davos, Bill Gates gave some insight into where he believes technology is heading. During the conference, which played host to several big names in the industry such as YouTube's Chad Hurley and Flickr's Caterina Fake, Gates touched on topics such as the Web 2.0 craze, IPTV, and a possible micropayment model for the web." 

Gates stressed that software is becoming more advanced and capable every year for hosting multimedia content, saying, "Every year we just move to more of a digital environment. We take away the older approaches." In the next few years, for example, we should expect to see the disappearance of the Private Branch eXchange (PBX) in telephone systems. "In voice telephony, you have a thing called a PBX. You won't have those anymore. You'll have a communications system that is using your Internet network and it's a far richer, more flexible software-drive system."
 

Posted by Dal at 07:03 PM | | TrackBacks (0) Posted to Blogsphere News

January 27, 2007

Rich Tehrani - the man who breathes IP Communications

Note:  I was browsing on Tom Keating site this afternoon while I wait for lunch in between a weekend conference I am attending and I found this little write-up from Rich Tehrani about his time at ITExpo.  Here is a snippet from the post.

"Rich is a man who eats, sleeps, breaths IP communications. In fact, he barely slept this week at ITEXPO since he was so busy meeting with VoIP service providers and VoIP vendors, as well as speaking at the show and walking the show floor." 

I'm at the show myself, but I always keep one eye on Rich's blog to see what he's writing about. This morning, Rich woke up at an ungodly hour of 3:30am and then at 5:29am Rich posted an excellent recap of all the happenings at ITEXPO. He potentially has the scoop about a new Skype offering called Skype Pro that is worth checking out.
 

Posted by Dal at 12:53 PM | | TrackBacks (0) Posted to Blogsphere News

January 26, 2007

NEC Develops Technology to Prevent IP Phone SPAM called VoIP Seal

Note: Thank you NEC for getting started on this problem early in the game. 

NEC Corporation announced the development of new technology for the prevention of Spam over Internet Telephony (SPIT). VoIP SEAL, the new technology, which defends against the threat of rapidly increasing spam IP phone calls, is expected to contribute significantly to the realization of safe voice over internet protocol (VoIP) phone networks in the future. VoIP SEAL will be exhibited at NEC's booth at the 3GSM World Congress 2007, taking place in Barcelona, Spain from February 12 - 15.

The main features of VoIP SEAL are as follows:

(1) Calls arising from spam-generating-software and calls from real individuals are separated by a Turing test. Before connecting the call, VoIP SEAL detects and blocks the unauthorized access based on the communication pattern observed during a call. This enables the detection and blocking of SPIT and prevents the user's phone from ringing unnecessarily.
(2) By adopting a module structure, VoIP SEAL enables rapid response to new kinds of SPIT attacks, without adjusting the system, by adding and updating modules to respond to new and different kinds of SPIT.
(3) The adoption of a module structure also realizes response to a broad range of applications by enabling flexible and easy customization of systems to meet the needs of a variety of hardware, such as SIP servers, SBC, home network equipment and terminal equipment.

NEC carried out a SPIT attack simulation project employing VoIP SEAL to verify the technology's ability to protect against SPIT. This project showed that 99% of SPIT was detected and blocked, preventing users from receiving unwanted and bothersome calls.

In recent years, the spread of low-cost IP phones has advanced significantly in comparison to fixed-line phones as a new method of communication in the next-generation network environment. However, although IP phones offer cost advantages, they also act as an easy platform for generating spam calls. The cost of generating a spam call over the internet is cheaper than in a traditional network by a factor of 1000.
 
As a result, the existing infrastructure for producing spam e-mails (so called "botnets") can easily be modified to also produce spam telephone calls. Today, the number of spam emails is higher than the number of regular emails produced jointly by all of the users in the internet. If unsolicited marketing and spam calls become as frequent as spam email, constantly-ringing VoIP phones may hinder the spread of their use.

VoIP SEAL can protect and defend against SPIT, and is expected to contribute to the realization of a safe and secure NGN. The modular platform provides the flexibility required to defend against smart attackers and spammers, who are continuously enhancing their spamming software and techniques. NEC will continue to develop this technology toward its early commercialization as a VoIP solution.
 
Source: NEC Inc. 

Posted by Dal at 10:39 AM | | TrackBacks (0) Posted to VoIP Security

IEEE Committee Members Unanimously Approved draft standard version 1.10 for 802.11n

Things may be smoothing out for the much-embattled 802.11n wireless standard. IEEE committee members unanimously approved draft standard version 1.10 and laid the groundwork for draft 2.0 at a meeting in London last week, paving the way for WiFi Alliance compatibility specifications, and a wave of product announcements.
The first 802.11n draft, in May of last year, was voted down, leading to some confusion in the market, and several vendors marketing "pre-draft" 802.11n products. According to a report on eWEEK.com, the newly approved draft clears the way for the WiFi Alliance to publish specifications.
 
The Alliance's specifications in turn allow manufacturers to build products with full assurance of compatibility or, in many cases, upgrade the firmware of existing products for full compatibility. With the first draft of the 802.11n spec squared away, and a second draft likely to follow quickly, eWEEK expects a raft of new product announcements, both from chip vendors Atheros and Intel, and access point vendors Asus, Belkin, Buffalo, D-Link, and NetGear.
 

Posted by Dal at 10:28 AM | | TrackBacks (0) Posted to WiFi / Wireless

Setup MV-370 GSM Gateway with Asterisk

Note: Found this handy help article on how to setup a approx. $150 GSM gateway from ebay to work with your Asterisk PBX.  Internally on our office we are going to be deploying a GSM gateway to get a couple employees' cellphones without the big bills and the ability to test this concept out.

"The GSM gateway MV-370 is manufactured by http://www.portech.com.tw/.
It's main interest in comparison with other gateways is that it is GSM to SIP (not FXS), so voice quality is very good.
Another point of interest is the price (around $150 on ebay).
With that gateway properly configured, you are able to receive calls from GSM to Asterisk (including DISA) and to give calls from Asterisk to GSM network.
" 

Usage

A typical usage of such a gateway is to be able to give a call with your normal mobile to any destination at voip cost :
Your mobile <----gsm network----> MV-370 <--lan--> Asterisk <--internet--> VOIP provider <--whatever--> landline

To do such a call, you just call your MV-370 number (it has its own simcard), then you get an invitation tone, then you dial the number which is handled by Asterisk.
If you have some special deals with your mobile operator, like free special number, you can call your MV-370 for free.  You can then call all around the world from your mobile at voip cost :-)
 


Posted by Dal at 10:02 AM | | TrackBacks (0) Posted to Asterisk Help

January 25, 2007

Digium full-court press (release)

Note: Alec Saunders has posted this informative interview with Bill Miller at ITEXPO. 

Posted by Dal at 09:17 AM | | TrackBacks (0) Posted to Blogsphere News

PIKA Connect for Asterisk adds support for PrimeNet T1/E1 Gateway Board

Note: Danny Sullivan sent this release from PIKA.  Thanks Danny.
 
PIKA Technologies Inc., announced today the release of a new version of its PIKA Connect for Asterisk software package.  Among the improvements and new features it contains, this release allows Asterisk users to take advantage of the DSP-quality software-based echo cancellation offered by PIKA's PrimeNet T1/E1 Gateway board and PIKA Connect for Asterisk software.
The PrimeNet T1/E1 Gateway board was first released to the market in the spring of 2006. Providing support for up to four T1/E1 spans, this 5.3-inch PCI card was specially designed and optimized to work with PIKA's patent-pending AllOnHost media processing software technology. The number of spans the board supports is field upgradeable; if a single span board is initially purchased, it can be expanded to a dual, triple or quad with the simple purchase of a software license.
 
Installation is as simple as downloading and installing the PIKA channel driver for Asterisk with the AllOnHost media processing software and plugging in the board. A PCIe version of this card, offering the same features and benefits, will be available later this year. PIKA Connect for Asterisk is a channel driver distributed under the GNU Public License.
 
PIKA's prior version provided Asterisk interoperability with its high-density, four- to 24-port analog boards and low density, one- to four-port analog boards, as well as with Skype. This new version adds interoperability with PIKA's new T1/E1 Digital Gateway board, thus offering a full line of network connectivity options to Asterisk developers and end users around the world. "The Asterisk development community continues to benefit from advanced features like reliable fax and DSP-quality echo cancellation in both analog and digital applications, made possible by PIKA's advanced media processing technology," said Terry Atwood, Vice President of Sales, Marketing and Customer Care at PIKA.
 
"Finally, the Asterisk community has a cost-effective answer when customers and users demand traditionally high levels of voice and fax quality and reliability." Asterisk users can now shop online for PIKA hardware PIKA also opened its new e-commerce web store today.
 
Available through the company's main site at www.pikatechnologies.com, the store gives Asterisk developers and end users a fast, easy method to procure the hardware they require. PIKA's analog and digital boards for Asterisk are available through the store. PIKA Connect for Skype, an Asterisk plug-in that allows incoming Skype calls to be handled by an Asterisk system in the same manner as calls received from the traditional telephone network, is also available.

Posted by Dal at 08:54 AM | | TrackBacks (0) Posted to Asterisk Hardware

January 24, 2007

Digium and Polycom Release More Details on their Asterisk Partnership Program

Note: Very interesting move by these two.  Personally I have been waiting for an announcement like this to see where this partnership is going.  I think this is a great first step and I look to see more integration in the future with more product offerings. 
 
Digium and Polycom today announced a VoIP solution for small and medium-size businesses that simplifies the technical aspects of purchasing, configuring and deploying a complete VoIP phone system. The integrated solution, featuring the Digium AsteriskNOW software appliance and Polycom SoundPoint IP desktop phones can significantly lower the time and technical expertise needed to deploy a high quality VoIP solution to enjoy the cost savings, enhanced quality and productivity benefits of advanced IP telephony.
The simplified purchasing, configuration and deployment process is enabled by new capabilities in the Digium AsteriskNOW software appliance, including a one-click function called BuyNOW that enables customers to purchase Polycom SoundPoint IP phones, and an intuitive configuration process that automatically provisions the phones for immediate customer use. In addition, Digium's Asterisk software now supports Polycom HD Voice technology that enables calls with twice the clarity and sound quality of traditional analog calls.

"This expanded offering with Polycom makes it even easier for SMBs to deploy and manage an Asterisk-based solution -- from the beginning stages to deploying the phones," said Mark Spencer, president and CEO at Digium. "This development is part of our overall strategy to offer an easy, rapid migration to VoIP in an enterprise environment. Partnering with Polycom further emphasizes our commitment to providing users with a simple Asterisk installation, based on only the best quality products."

"Many small and medium businesses want an advanced, high quality and affordable VoIP solution, but the technical challenges can be a barrier," said Sunil Bhalla, senior vice president and general manager, voice communications at Polycom. "We are working with partners like Digium to deliver unique capabilities like Polycom HD Voice and to simplify the deployment process. As open source continues to play an important role in the evolution to VoIP, we look forward to working with Digium to provide innovative, cutting-edge solutions that help customers make this important transition."

The BuyNOW feature, within AsteriskNOW's Digium-designed GUI, greatly simplifies the phone purchasing process by immediately connecting users to NETXUSA, a recognized leading distributor of Voice over Internet Protocol (VoIP) products and services. With regional distribution offices and in-house certified engineers, NETXUSA assists business customers with the entire Digium-Polycom implementation. Polycom's full line of award winning and industry leading standards-based phones are available through BuyNOW including: the SoundPoint IP 650 with HD Voice, SoundPoint IP 601, SoundPoint IP 501, SoundPoint IP 430, SoundPoint IP 301 and SoundPoint IP Expansion Module. The line also includes the SoundStation® IP 4000, Polycom's market leading SIP-based conference phone.

The AsteriskNOW GUI also includes a simplified process for configuring and provisioning Polycom's full line of SoundPoint IP phones. The interface provides a step by step process that helps the user select from the broad array of features that are available in a combined Asterisk and Polycom solution. Once the selection process is complete, the configuration is automatically downloaded to the phone which immediately becomes active on the system.

In addition to the updates in the AsteriskNOW GUI, the Asterisk open source software (release 1.4.0) now supports Polycom's breakthrough HD Voice technology delivering the ultimate communications experience. Polycom's HD Voice includes wideband audio, enhanced signal processing, Acoustic Clarity Technology which includes next generation technologies for transparent full duplex, echo cancellation, dynamic noise reduction, automatic gain control and microphone management) and specialized system design to deliver unrivaled clarity and richness. This enables significantly better voice clarity and improved intelligibility of information, which significantly improves comprehension and productivity while reducing listener fatigue. Polycom HD Voice is currently available on the SoundPoint IP 650 telephone.

These new capabilities are the result of collaborative efforts between Digium and Polycom and follow the recent announcement of their partnership to supply Polycom SIP-based phones as the exclusive phone in Digium's currently available Asterisk Appliance Developer Kit (AADK). Combined with Digium's Asterisk Business Edition, SMB customers benefit not only from a rich feature base capable of rapid deployments; but also, from a more affordable telephony solution as compared to proprietary systems.

Source: Polycom Inc. and Digium Inc. 

Posted by Dal at 01:41 PM | | TrackBacks (0) Posted to Polycom

The 10 Best Video Conferencing Solutions for 2006

Note: Found this breakdown of all these top-notch video conferencing and telepresence solutions 
 
The experience of the user of video conferencing systems has become better and better over the last five years. "In 2006 there was a giant step forward with major vendors offering new High Definition Telepresence and Videoconferencing Solutions," comments online Specialist Newsletter Videoconferencing Insight at VC Insight when announcing The Editor's Choice of "The 10 Best New Video Conferencing Systems of the Year 2006" in the 15 January 2007 issue.
"There were two enormous changes in the New Videoconferencing Systems produced in 2006. We provide a snapshot of a fast-changing industry. The first is the arrival of new High Definition VC systems from Aethra, LifeSize, Polycom, Sony and TANDBERG; the second is the release of HD Telepresence solutions from Cisco, HP, Polycom and Teliris. Both these developments made videoconferencing more lifelike and took the industry to a new high level in 2006," said Richard Line, Editor of online Specialist Newsletter Videoconferencing Insight at VC Insight.

In 2005, the widespread use of LCD and plasma screens made for a more elegant design, a slimmer look and a smaller footprint of all 10 videoconferencing systems selected as the Best New Videoconferencing Endpoints of 2005.

In 2006, the 10 Best New Videoconferencing Systems all offered High Definition (HD) video and audio. In 2006, there were new HD Videoconferencing Systems from Aethra, LifeSize, Polycom, Sony and TANDBERG; and four new HD Telepresence suites from Cisco, HP, Polycom and Teliris.

Telepresence suites are designed for an entire room dedicated to videoconferencing and video meetings and include suitably designed furniture as well as three or four large screens to provide realistic lifesize images of persons at the far end.

The Editor believes that most users will prefer a High Definition (HD) videoconferencing system when they see one in operation. One way to understand why HD is superior is to visit www.lifesize.com and view the high resolution that HD videoconferencing provides.

The philosophy of the Newsletter Videoconferencing Insight has long been that no single videoconferencing system is the best for each and every situation. Therefore 10 systems are included in the Editor's choice of "The 10 Best New Videoconferencing Systems of 2006" provided in the 15 January 2006 issue. The systems are listed in alphabetical order below:

1. Cisco TelePresence System with 1080p HD video and HD audio. The best room and furniture design. Proprietary non-standard technology. As yet, no multipoint links. Cisco has plenty of orders and will install over 100 systems for its own use.

2. HP Halo Collaboration Studio TelePresence System with high quality video and audio. Proprietary technology. Halo will link to standards-based systems in Q2 2007 thanks to collaboration with TANDBERG. HP has installed 90 systems worldwide so far, including many for its own use.

3. Polycom HD Telepresence Solution - Polycom RPX HD with 720p HD video on large cinematic walls and its own Siren HD audio. Eye-level HD cameras. Standards-based technology. Multipoint with Polycom MGC HD MCU.

4. Teliris VirtuaLive HD unified and customized Telepresence Suites with 1080p video at 60fps and HD audio. Flexible designs for rooms. Multipoint available. Supports SD video conferences.

5. The Aethra Vega X7 high definition (HD) visual communication systems is optimised for 768 Kbps; it provides 720p HD video, stereo HD audio, a nine-site MCU and a third-party HD camera.

6. LifeSize Room HD videoconferencing system with 720p HD video and LifeSize own proprietary HD audio system. Multipoint for four sites. Meeting scheduling from Microsoft Outlook.

7. The Polycom HDX 9004 high definition (HD) visual communication system provides a HD video format of 720p resolution (1280 x 720 pixels), Polycom HD audio, Polycom HD camera and more.

8. The Sony PCS-HG90 high definition (HD) visual communication system provides a HD 720p video format at 60 fps and a video transfer rate up to 8 Mbps over an IP network. Multipoint for 4 sites.

9. The TANDBERG 95 MXP, 85 MXP and 75 MXP high definition (HD) visual communication systems provide HD video at 720p resolution, stereo HD audio, and a TANDBERG HD camera.

10. The TANDBERG Centric 1700 MXP high definition (HD) visual communication system gives busy executives at the desktop HD video at 720p, stereo HD audio, and a TANDBERG HD camera.
 
Source: VC Insight

Posted by Dal at 12:04 PM | | TrackBacks (0) Posted to Blogsphere News

Introducing Version 3 of the Plug-and-Play Asterisk IP PBX for the Intel Mac

Excerpt: Thanks to the work of literally hundreds of developers, there is a terrific Asterisk IP PBX with an incredible array of additional bells and whistles. That product which we have tested extensively is TrixBox 1.2.3. It’s so good, in fact, that we chose it as the base system for all of the Nerd Vittles applications that we write about each week.
What was missing unfortunately was a way to run this same system on a Mac. So today we have not one but two special treats for the Mac enthusiasts of the world.
 
First, it’s now possible to run our standard Version 3 system using the new VMware beta for the Intel Mac. And, thanks to one of our great contributors, there’s now another alternative: a Parallels image of our Version 3 Asterisk system. Today, we'll show you how to install both of them...
 

Posted by Dal at 10:19 AM | | TrackBacks (0) Posted to Asterisk Help

The Asterisk Appliance Developer Kit - Overview

Note:  I found this on another blog.  It goes a little more into Digium's Asterisk Appiance and has a photo of the back with a diagram.  I am still on the fence if I should purchase one of these to play around with.   Today I plan to install AsteriskNow on an old Dell to see how it does.  I hope its as easy as Mark said it was in his recent YouTube apperance.

"The Asterisk Appliance is a standalone embedded PBX. Targeted for small to medium businesses (2-50 users), and remote branch offices of larger organizations (2-50 users per site), the Digium Asterisk Appliance will feature the commercially licensed Asterisk Business Edition software and the first Digium-developed Asterisk GUI framework."

The Asterisk Appliance appears to take advantage of the Blackfin DSP's microcode programmability by implementing echo cancellation, and possibly other telephony functions, in hardware.
 
The Appliance's I/O includes eight analog ports (FXS, FXO), a WAN port, four LAN ports, hardware echo cancellation, and a "craft port" for debugging. Expansion is available through a CompactFlash slot suitable for voicemail storage cards or wireless radio peripherals.
 
 

Posted by Dal at 09:42 AM | | TrackBacks (0) Posted to Asterisk Hardware

January 23, 2007

Azimuth Systems Introduces Next-Generation WiMAX Channel Emulator

Azimuth Systems, introduced the new ACE™ 400WB, a comprehensive channel emulator platform for testing multiple-input-multiple-output (MIMO) and single-input-single-output (SISO) WiMAX solutions. The ACE 400WB is the industry’s first WiMAX channel emulator – a purpose-built, single box solution that provides sophisticated channel modeling capabilities in a complete test automation platform. The Azimuth solution enables real-time performance testing of MIMO devices, streamlining the testing of WiMAX chipsets, clients and infrastructure.

The ACE 400WB accurately emulates multipath characteristics with channel correlation to determine the effect of multi-channel RF interactions. The advanced signal processing technology is integrated in a platform that automates device control, traffic capture and results display to deliver accurate, repeatable and fast results. The ACE 400WB can be used to test and debug MIMO algorithms, optimize the performance of WiMAX devices in MIMO and SISO environments, streamline QA processes and run competitive performance benchmark tests. The ACE 400WB also tests interoperability between MIMO and SISO implementations from multiple vendors and can be used to define industry-wide mobility performance test suites for future WiMAX products.

“The growing WiMAX development market provides an exciting opportunity for Azimuth,” said Jeff Abramowitz, vice president of marketing at Azimuth Systems. “The ACE platform redefines customer expectations for channel emulation and we’ve seen tremendous interest in the ACE 400WB, particularly from vendors familiar with our award-winning Wi-Fi channel emulator.”

In next-generation MIMO systems for both Wi-Fi and WiMAX technologies, multipath emulation with correlated channels is required to determine whether MIMO algorithms are working properly and to predict performance of MIMO-based products in a real-world environment. However, multipath is difficult to predict and control because it is affected by everything from building construction to the movement of people, so accurate channel emulation is a critical element of any MIMO test