If you were ever wondering why your Cisco Gear if not performing properly, better check where you got them and verify with Cisco if it is Genuine. US and Canadian Authorities have seized more than US$78 million worth of counterfeit Cisco Systems networking equipment in an ongoing investigation into imports from China, the U.S. Department of Justice and other agencies announced Friday.
It is more than just counterfeit hardware, "Counterfeit network hardware entering the marketplace raises significant public safety concerns and must be stopped," Assistant Attorney General Alice Fisher of the DOJ's Criminal Division, said in a statement. "It is critically important that network administrators in both private sector and government perform due diligence in order to prevent counterfeit hardware from being installed on their networks."
You can read more about it at the source and always you can check with Cisco.
tag: cisco, Cisco Counterfeit Gear, Cisco Gear
Friday, February 29, 2008
If you were ever wondering why your Cisco Gear if not performing properly, better check where you got them and verify with Cisco if it is Genuine. US and Canadian Authorities have seized more than US$78 million worth of counterfeit Cisco Systems networking equipment in an ongoing investigation into imports from China, the U.S. Department of Justice and other agencies announced Friday.
Zimbra, which I use at work and Skype that I some time help to remove (remove Skype), (I do use Skype!) has come together to create a unified communications solution. This solution is from OnState Communications, The Callcenter Guys, as they call them selves.
Newburyport, Mass. — February 12, 2008 —OnState Communications, innovating new-generation contact center solutions, today announced OnState Unified Messaging for Skype™. This single-platform Skype messaging solution integrates multiple business-communication modes such as: email; voicemail; business chat; and customer contact messaging and recording systems. Starting as low as $10/per month, the solution enables communications-enhanced business processes for small- and medium-sized businesses that use Skype.
“Today, business interactions happen via a multitude of modes - verbal and written, real-time and asynchronous,” said Pat Kelly, CEO of OnState. “By mashing-up Zimbra, Skype and OnState, we’ve taken the first step in melding these communication modes to create a unified messaging solution.” Skype is a leading provider of real-time, Internet-based conversations; Zimbra offers open-source, next-generation messaging and collaboration; and OnState’s contact management software ensures customer requests are delivered to the right resource at the right time.
OnState Unified Messaging for Skype routes, manages, stores and archives a range of business communications to expedite business processes and drive improved customer-contact management. Features include:
- Web 2.0 Ajax client
- integrated Skype voicemail
- dynamic call recording
- online business chat logging
- enterprise-class email and calendaring
- push-mail for mobile devices
- support for Outlook™, Thunderbird and other mail clients
- cross-mailbox search and compliance features
“With this latest offering, OnState advances its charter to deliver Web-based, enterprise-class, customer-contact management solutions at an industry-redefining price point,” noted Kelly.
OnState Unified Messaging for Skype is available now on the OnState web site and through select partners.
tag: managed solutions, skype, Unified Communications, Zimbra Collaboration Suite, The Callcenter Guys, Unified Messaging for Skype, Skype messaging solution, Zimbra,
Thursday, February 28, 2008
If you have not yet signed up, I think now you have another reason to join TringMe. TringMe allows you to make calls from Google Talk (Gtalk), or Googletalk (and to some gooletalk ;) ). Gtalk doesn't support making phone calls natively, basically you can't dial anyone from within Gtalk. Enter Tringme, if you already have Gtalk installed, then making a call to any phone or SIP URI from Gtalk is as simple as sending a message using Gtalk. Calls made from Gtalk can be terminated on all devices that TringMe currently supports.
Is that sweet or what? Ok here how it really works, straight from TringMe blog;
"To make a call, all you need to do is send a message to email@example.com from Gtalk. Please note that you would have added TringMe to the list of friends in Gtalk already.
- To make a call to a number, merely send a message - call <number> - to firstname.lastname@example.org. For e.g. call 18585551212 would initiate a call to 1-858-555-1212.
- You can also call other TringMe users by merely using their email address. For e.g., call email@example.com would connect user firstname.lastname@example.org to whatever destination the user has chosen to terminate upon.
- You can reach more than 40 million worldwide SIP users using Gtalk just by typing their SIP URI. For e.g call email@example.com will connect you to Greg's SIP server.
In all cases, you will get an inbound call from TringMe and once you answer, the call to the intended party will be initiated. This also opens a new way for developers to integrate core TringMe functionalities using Google or Jabber APIs."
Although it required invites to join TringMe, I just signed up today without any problems at TringMe site. I certainly will be writing more about TringMe, once I had a good taste of it.
Posted by ravenII at 2/28/2008 01:22:00 PM
Posted by ravenII at 2/28/2008 12:44:00 PM
When I first read the article, I was aghast, getting a call in my own voice? (How many times I had to ask, Is that my voice when I hear on the answering message!).
Nerd Vittles' Telephone Reminders for Asterisk 1.4 not only lets you schedule reminders by phone using your own voice, but now you can use a clever (if we do say so) web interface as well. Just fill out a simple web form to set your reminder or recurring reminder in motion, and Telephone Reminders for Asterisk will swing into action with Flite or Cepstral's Allison to deliver your typed message to the phone of your choice at the appointed time.
They are also celebrating the new FreePBX Training Seminar being held in. this week, They have added a new FreePBX Interface to Telephone Reminders for those who rely on FreePBX to manage your Asterisk PBX.
Telephone Reminders 3.0 tutorial is available on Best of Nerd Vittles site. And If you are using PBX in a Flash, you do not need to wrangle with scripts.
I think Ward's forum post will be a good place to start, or may be this blog post ;).
tag: Asterisk, FreePBX, FreePBX training, FreePBX 2.4.0,pbx in a flash, Nerd Vittles,
Posted by ravenII at 2/28/2008 11:52:00 AM
Om Malik at GigaOM said I'd be awed, no I am dumb founded!, I really thought Ebay-Skype was a big deal.
Sprint Nextel reported a $29.5 billion loss, scratched its dividend and lost 683,000 customers. The company wrote down $29.7 billion of the $36 billion it paid for Nextel in 2005.
tag: Telecom News, Ebay-Skype, Sprint Nextel, Sprint's Nextel loss
Voice over broadband continues to be the top application driving service providers to invest in VoIP and IMS networking equipment, according to Infonetics Research's latest "Service Provider VoIP and IMS Equipment and Subscribers" report.
Sales of service provider VoIP and IMS equipment grew 19% between 2006 and 2007, to over $3.9 billion, the report shows, with all segments in the market posting growth. Some segments, such as IMS core equipment, nearly doubled year-over-year.
"The overall next gen voice equipment market is maturing, as evidenced by single-digit annual sales growth in traditional equipment such as media gateways and softswitches, but other types of equipment, namely session border controllers, media servers, and voice application servers, will generally grow faster for the next five years. As we are starting the fourth year of a global telecom investment cycle and leaning toward a plateau, the overall next gen voice equipment will likely weather a slowdown starting in 2010 or 2011, buoyed by strong IMS core equipment sales," said Stéphane Téral, principal analyst for VoIP and IMS at Infonetics Research.The report provides analysis and rankings for Acme Packet, Alcatel-Lucent, AudioCodes, BroadSoft, Cantata, Ditech, Cisco, Comverse (NetCentrex), Ericsson, GENBAND, Huawei, IP Unity Glenayre, Italtel, MetaSwitch, Netrake, NexTone, Nokia Siemens Networks, Nortel, RadiSys (Convedia), Sonus, Tekelec, Thomson Cirpack, UTStarcom, Veraz, Verso, Xener, and many others.
Download report highlights by logging in to Infonetics' access portal from www.infonetics.com (see Service Provider VoIP and IMS). For sales, contact Larry Howard, vice president, at firstname.lastname@example.org or +1 (408) 583-3335.
News Via MarketWire
tag: voip news, VoIP Market Research, VoIP, IMS, softswitches, media gateways,
Wednesday, February 27, 2008
ARS is covering the e911, real 911 services for VoIP, Voice over IP services. Although FCC has been pushing for VoIP solution providers to provide proper 911 services known as e911, too much political and legal wrangling has lead to uninformed general public meeting with trouble while trying to use 911 services, like when they had PSTN services.
Finally we might be getting somewhere now that the bill submitted by Senator Ted Stevens (R-AK) has been approved by senate.
"All VoIP customers are one step closer to having "real" 911 services accessible to them, thanks to the Senate. The body passed the IP-Enabled Voice Communications and Public Safety Act last night, a bill that had been proposed by Senator Ted Stevens (R-AK). The legislation was approved by the Senate Commerce Committee last year, and will now go on to the House of Representatives for further consideration."
tag: E-911, e911, FCC, "real" 911 services, IP-Enabled Voice Communications and Public Safety Act
Techcrunch is telling us that Apple has sent out an email to the media announcing a March 6th event for laying out an "iPhone Software Roadmap" that will focus on the SDK that I mentioned this morning, as well as "new enterprise features". This event is invitation only and you guessed right, I do not have an invitation ;). I will be looking out for news from those lucky ones, unless I have the email on my Apple account which I have not checked yet.
tag: iPhone, IPhone Dev Kit, iPhone development kit,
Posted by ravenII at 2/27/2008 05:20:00 PM
I am very happy to see fellow blogger Alec Saunders' iotum going global with it's FREE Conference Calls application on facebook. iotum is supported by a international group of companies. Namely they are, Truphone in the U.K., Abbeynet in Italy and MOI Telecom in France. Following is the Press release by iotum and I suggest you also read, VoIP Watch article by Andy Abramson, one of the people responsible (I hope that gentle push came from you Andy, not MOI, Forgive me for guessing.) for this marvelous marriage (Only place I agree on Polygamy is market place). Andy, I think I am in need to find some purple minutes.
OTTAWA & February 27, 2008-- iotum today announced global availability of its popular FREE Conference Calls application on Facebook under agreements with Internet communications partners Truphone in the U.K., Abbeynet in Italy and MOI Telecom in France.
Iotum CEO Alec Saunders said the global expansion of FREE Conference Calls is in response to demand from users of the application launched on Facebook last September. "Now, thanks to our partnership agreements, anyone can join a FREE Conference Calls from anywhere in the world."
"FREE Conference Calls has many standard features not available from other providers," he said. "We give our users the ability to provide agendas for their calls; moderators can mute or unmute the calls; record a podcast; schedule calls in advance, or provide reminders. Hosts can also queue participants to speak, and we even provide a live "writing wall" for interactive text chat."
Since launch, Iotum's FREE Conference Calls service on Facebook has attracted more than 90,000 users. Conference calls have been created to hold public teleseminars, private meetings, family events, New Year's Eve countdowns, political discussions and to record multiperson podcasts.
Starting today, users can join an iotum FREE Conference Call using one of three methods:
- Direct dial from home, office, or mobile. In the United States, Canada and France, users can dial the conference server directly using an ordinary call and be joined to the FREE Conference Call. Iotum intends to roll out more directdial countries in the future by recruiting additional partners to service other parts of the world.
- VoIP. From anywhere in the world, users can call using their PCs. All that's required is an active Internet connection, and clicking on the FREE Conference Calls application icon in Facebook.
- Call back. A user can be called back simply by entering into the FREE Conference Calls application on Facebook the phone number where his or her call is to be received.
"Participants don't have to be on Facebook, but it's more valuable for them if they are," Saunders said. "And it's easy. Facebook users simply install the iotum FREE Conference Calls application and invite friends for a group chat."
Sherman Hu, producer and host of the ShermanLive.com blog, wrote: "As a coach to thousands of business owners about the innovative combination and profitable deployment of blogs, podcasts, video and social media for business, I see the ability to conduct training calls and have anyone outside of North America dial in from anywhere in the world at no charge as a priceless benefit. This initiative will have farreaching impact for my global business."
At Truphone, Platform Director Dean Elwood said: "Truphone already has built a reputation for telephony innovation by providing free mobiletomobile calls via the Internet. This tieup with Iotum fits perfectly with our vision of providing new ways for people to call each other without heed to old world boundaries and pricing structures."
Posted by ravenII at 2/27/2008 01:29:00 PM
A few times I tried to use Skype HQ Video calls but failed and since then I have turned to other solutions. oovoo video call being one. But I was reminded of Skype HQ Video by a post on ZDNet and just wanted to check if they have changed. According to the post, it has not changed and I tried to see if my experience would be better, since the Internet has surprises for you based on where you are, your ISP, and sometimes your computer.
So I fired up the Skype and well I could not find any contacts at that time who met video requirements set by Skype. So it has to wait. Until then you can read the post by a1gjv on zdnet. If you are too frustrated, then it might be time and More reasons for Skypekiller?
Tags: Skype, skype video mod, skype 640X480 video
Posted by ravenII at 2/27/2008 12:51:00 PM
I am counting down to IPhone Dev Kit as promised by Steve Jobs to be released in February. Now it is almost the end of February and I have not seen any sign yet. Post by Tom Krazit has a good overview and some comments!.
tag: iPhone, IPhone Dev Kit, iPhone development kit,
Faxing with Asterisk has been always a little trouble, the leading Open Source IP PBX, a VoIP platform solution. Some time ago I wrote about PIKA providing support for faxing on Asterisk with hardware front;
"PIKA Connect for Asterisk integrates PIKA high-density and low-density analog boards into the open-source Asterisk environment. Delivering reliable, analog connectivity for up to 24 channels per board, PIKA adds powerful capabilities in DSP-based fax and echo cancellation to the Asterisk development platform. Other features include dial plan integration, native switching, DSP based conferencing, and OpenVPOS support."
But today I read another article by Tom Keating, let me know more about the subject of Faxing with Asterisk on the software front as well. Enjoy the good article about a good faxing solution fro Asterisk.
" PIKA Technologies announced support for fax applications built on the open-source platform Asterisk. According to PIKA, "PIKA Fax software is now extended to Asterisk, allowing developers on the platform to easily build reliable fax capabilities into their applications.""
tag: Asterisk, Faxing on Asterisk, Pika, open-source VoIP, OpenVPOS, DSP-based fax
VoIP Your Life, a leading provider of Voice over Internet Protocol digital telephone services, announces the official launch of its new affiliate program.
Oklahoma City, OK (PRWEB) February 27, 2008 -- VoIP Your Life, a leading provider of Voice over Internet Protocol digital telephone services, announces the official launch of its new affiliate program.
"This program will play a substantial role in the phenomenal growth of VoIPYourLife," said Gary Holshouser, President of VoIP Your Life, adding that VoIP Your Life is actively seeking affiliate partners wishing to capitalize on VYL's extremely generous program. "Our commission structure is so liberal that we're expecting a robust reaction from new our affiliates.
Sources indicate that VoIP Your Life will be providing affiliates with a private forum to facilitate faster communications and greater responsiveness. Added Mr. Holshouser, "We're also going to provide our affiliate partners with access to real-time statistics for sales and traffic. Anytime they wish to, they can pull up detailed reporting and payout figures and, of course, they will get dedicated account management. The potential for this program has us very excited, to say the least"
To celebrate the roll out of the affiliate program, the company is also offering a few 'Master Agent' opportunities for those affiliates who desire to earn additional residual income. These Master Agents receive a bounty for new sign-ups, plus a significant percentage of recurring revenue.
VoIP Your Life provides high quality telecommunication services, including access points, local phone numbers and a rich variety of features that work just like a regular phone service. According to the company's website, http://www.voipyourlife.com, VoIP Your Life is consistently rated as one of the top 5 VoIP companies from multiple independent review sources. The company is also offering for a limited time free hardware, free activation and free shipping with no term commitments and no termination fee.
tag: VoIP Business, VoIP News, VoIP Your Life, VoIP, VoIPYourLife, free VoIP hardware, free VoIP activation
Tuesday, February 26, 2008
I hope you know what oovoo is. If not read the press release below. For some time it has been in beta operation providing Video services like video calls and video meetings and video call recordings. But now they have extended the service for free telephone calls.
These services include;
* FREE calls from ooVoo to a landline or mobile phone in the USA* or Canada.
* No credit card required.
* Add phone participants to your ooVoo video calls.
* Introductory period includes up to 120 minutes of FREE phone calls.
* FREE phone calls to the USA and Canada available through March 1, 2008.
NEW YORK—February 6, 2008—ooVoo®, an innovator in the way people communicate with real-time video, today announced the release of the newest version of its real-time face-to-face video chat service, ooVoo 1.5. Now offering more ways for ooVoo users to communicate and share conversations. ooVoo 1.5 combines the highest quality video chat available online (adjustable to up to 30 frames per second), video conversation recording and calling to landline and mobile phones, beginning with worldwide calling to the continental U.S. and Canada.
“ooVoo users are constantly looking to communicate and share their lives and experiences with friends, families and co-workers,” said Philippe Schwartz, CEO of ooVoo. “In each new version of ooVoo we look to improve the quality of our service and offer features that will help our users to communicate in a more meaningful way. ooVoo 1.5’s new features build on our core promise of making communicating online more meaningful with the highest video quality available—real-time or recorded—which presents our users with unsurpassed, face-to-face communication quality.”
Providing the best communication experience found online today, ooVoo 1.5 offers new ways to create and share content including recording a video conversation for friends or colleagues or for posting online. More than 1 million video conversations take place on ooVoo every month, and now users can record a face-to-face “reunion” with friends and family members from anywhere in the world on ooVoo. Business colleagues can record conversations to create video “notes” of an online ooVoo meeting. Journalists or bloggers can create an interview series to be uploaded on his/her web site. The length of a recorded conversation is limited only by the storage capacity available on the hard drive of the user’s computer.
To further enhance the creative options available, ooVoo has added a suite of visual effects, such as backgrounds and facial overlays of characters and creatures that users can add to their video messages and live video conversations. In addition, the visual effects package allows users to share a desktop and files from their computer, on screen with participants in a conversation.
ooVoo 1.5 allows ooVoo calls—from anywhere in the world—to a mobile phone or landline in the continental U.S. and Canada. This functionality is a great enhancement to ooVoo’s high-quality, multi-party video conversations as it allows those who are unable to join via a computer meet with up to five others in an ooVoo conversation. It’s as easy to use as ooVoo’s video calling feature. Simply enter a phone number into the ooVoo dial pad and press the green call button. A phone icon will appear as the call connects and begins.
From February 4 to March 1, 2008, users who download ooVoo 1.5 will receive two hours of free outbound calls on ooVoo to any landline or mobile phone in the U.S. and Canada from anywhere in the world.
Additional features of ooVoo 1.5 include:
Import contacts from Yahoo!, Gmail, MSN Hotmail, Windows Live Messenger, AOL Mail, LinkedIn, Mac Mail, ICQMail, Mail.com and Lycos Mail ooVoo conversation ‘sidebar’ window allows use of the desktop during ooVoo video callsWith the adoption of ooVoo in more than 200 countries worldwide, ooVoo 1.5 supports 15 languages: Arabic, Bulgarian, Chinese, English, French, German, Hebrew, Italian, Japanese, Korean, Polish, Portuguese, Russian, Spanish and TurkishUsers can download ooVoo 1.5 at www.oovoo.com.
tag: Free Telephone Calls, oovoo, video calls, video meetings, video call recording, face-to-face video chat, ooVoo 1.5,
GrandCentral which I got invited even before Google acquired it is getting bigger. I mean really big. A post on Blogger Buzz letting us know that Blogger users know that they could sign up for GrandCentral accounts immediately. But the buzz is that it works as long as you have a google account. If you do not an account you can easily get one at Google.
I have been using GrandCentral for one number for all phones. Due to the nature of my work an hobby, I have too many cell phones and two office locations and home. I give people my GrandCentral number and internally I manage where I would receive the call. But there are othe uses as the blogger buzz explains.
If you get a webcall button, your blog readers readers could call you and leave voice mail messages for you without ever knowing your number.
Blogger Buzz: GrandCentral: receive calls and post voicemail with your blog
tag: GrandCentral, Blogger, Webcall, Blogger Buzz, webcall button,
According to following Arstechnica article, Google is getting into the job of making sure that you and I have enough bandwidth to handle all those applications that we are going to use. If you remember a few weeks ago we lost a few sub marine cables that carry data to and from middle east and south east Asia. Google making sure that the growth needs are met and this undersea cable that Google taking part in will run from USA to Japan.
So finally Google might be able to Tame the Submarine Cable Dragons
"Google announced today that it plans to join five other international companies in building an additional undersea cable between the US and Japan. The need for the cable itself is unsurprising, given that trans-Pacific bandwidth needs grew by 63.7 percent between 2002 and 2007, but the fact that Google is joining the consortium of Bharti Airtel, Global Transit, KDDI Corporation, Pacnet, and SingTel is unusual. Google is insisting, however, that this new move does not signal a change in focus for the company."
tag: submarine communication cables, Google, Bharti Airtel, Global Transit, KDDI Corporation, Pacnet, SingTel, undersea cable
Monday, February 25, 2008
Nokia Research Center and the University of Cambridge are Morphing into the future with a concept cell phone called Morph. The Morph is featured in an online display presented in conjunction with "Design and the Elastic Mind," (which itself is a new experience to watch over the net. Follow the link to watch the exhibition online and prepare to spend some time!) scheduled to run through May 12 at the Museum of Modern Art in New York City.
Morph, the joint nanotechnology concept, launches today alongside the new Design and the Elastic Mind exhibition at The Museum of Modern Art (MoMA) in New York City in which it is profiled. The exhibition will be on view from 24 February to 12 May 2008.
Morph is a concept that demonstrates how future mobile devices might be stretchable and flexible, allowing the user to transform their mobile device into radically different shapes. It demonstrates the ultimate functionality that nanotechnology might be capable of delivering: flexible materials, transparent electronics and self-cleaning surfaces.
Professor Mark Welland, Head of the Department of Engineering's Nanoscience Group at the University of Cambridge and University Director of Nokia-Cambridge collaboration, commented: Developing the Morph concept with Nokia has provided us with a focus that is both artistically inspirational but, more importantly, sets the technology agenda for our joint nanoscience research that will stimulate our future work together.
Dr. Tapani Ryhanen, Head of the NRC Cambridge UK laboratory, Nokia, added: We hope that this combination of art and science will showcase the potential of nanoscience to a wider audience. The techniques we are developing might one day mean new possibilities in terms of the design and function of mobile devices. The research we are carrying out is fundamental to this as we seek a safe and controlled way to develop and use new materials.
The partnership between the University of Cambridge and Nokia was announced in March, 2007 - an agreement to work together on an extensive and long term programme of joint research projects. NRC has established a research facility at the University's West Cambridge site and collaborates with several departments initially the Nanoscience Center and Electrical Division of the Engineering Department on projects that, to begin with, are centered on nanotechnology.
The Nanoscience Centre provides open access to over 300 researchers from a variety of University Departments to the nanofabrication and characterisation facilities housed in a combination of Clean Rooms and low noise laboratories. Research is aimed especially at multidisciplinary projects where engineering, biology, physics, chemistry and materials science meet.
The Electrical Engineering Division of the Department of Engineering builds on Cambridge's history of world-leading research in Photonics and Electronics by significantly enhancing collaboration with industry and by providing a focus for multidisciplinary research involving over 200 engineers, as well as chemists, physicists, materials scientists and bioscientists. It includes the 'Centre for Advanced Photonics and Electronics' and the 'Cambridge Integrated Knowledge Centre for Advanced Manufacturing Technologies for Photonics and Electronics'.
A video is available to watch at Nokia.
For further information, please contact the University of Cambridge Office of Communications on 01223 332300
tag: Future Phone, Nokia, Nokia Research, University of Cambridge, The Morph, MoMA, nanotechnology, Design and the Elastic Mind,
Sunday, February 24, 2008
Gizmo has a few different ways that will allow you to make all your calls free. One of the special programs allows you to call certain US numbers for free.
In order to use this backdoor, as Gizmo calls it, you need to find out if the number you want to call qualifies, for free calling. I did for one of my numbers and that session is in the image above, and I qualified. It is a mobile number. You will find your backdoor information here.
How to do backdoor dialing:
Two-way calling feature:After calling a mobile or land line phone that's available for Backdoor Dialing using Gizmo5, the person you called can call you back directly on your computer anytime you're logged in to Gizmo5!
Gizmo Project also have a program called "All Calls Free" Plan. And this is how that works;
If both parties are logged into Gizmo Project, you should just make a Gizmo-to-Gizmo call, which has always been a free call. If the person you want to call is not currently online or logged into to Gizmo at the time you call, you can then dial the "home phone" or "mobile phone" numbers they have added into their Gizmo Project profile. That call will be free provided (a) you both are active Gizmo Project users, and (b) are calling a qualifying number in one of the 60 countries for which the plan is offered.
Gizmo would love to make this calling plan work for all numbers around the world. However, (at least for now) They are only able to allow free calls to landline phones in 60 countries, and mobile numbers in 17 of those countries (see chart below). If the person you want to call lives in a country that's not on the list, remind them to get Gizmo Project so you can call them on their PC for free, or purchase Gizmo Call Out credit to make calls at extremely low calling rates."
All Calls Free plan is currently is in effect for the following 60 countries and types of phone lines:
|Landlines & Mobiles|
US Virgin Islands
tag: Free calls, All Calls Free, Gizmo Project, Gizmo Call Out, landline phone, Gizmo-to-Gizmo call, Gizmo5, Gizmo5 Backdoor Dialing
VoIP News has a "feature article" on VoIP Security. There are many VoIP Solutions for making VoIP calls but security that people used to get with (did we really?) is certainly diminished. So go read the article to be savvy about VoIP Security Solutions.
tag: voip news, VoIP Security, VoIP Solutions
ATLANTA, GA – (February 20, 2008) – Verso Technologies, Inc. (NASDAQ: VRSO), a global provider of next generation network solutions, announced today that it has been selected to provide a North American Tier One telecommuniciations carrier with a Vclear Edge VoIP solution which will provide the carrier with the ability to offer a true VoIP Managed Service Offering.
“Carriers have been slow in the deployment of VoIP services to their enterprise customers due to the inability to reliably deliver on Service Level Agreements (SLAs) similar to what they offer today,” said Steven A. Odom, Chairman and Chief Executive Officer of Verso. “Our new Vclear Intelligent Demarc technology is paving the way for larger carriers to finally begin deploying VoIP reliably to their enterprise customers. Their rollout will provide them with a significant head start and strong competitive advantage as they transition away from circuit switching services.”
“We believe this may be the most significant initiative this customer has for the immediate future – a true VoIP Managed Service Offering,” said Jeff Donahue, Vice President – North American Sales for Verso. “It could have a place in every VoIP managed service offering that they deploy. The solution provides several benefits to the customer at both pre-deployment and post deployment stage with the end user. In pre-deployment, the carrier can assess network readiness in a non-intrusive manner, actively verify the networks capability, measure the networks performance and allow it to be tuned prior to deployment. In the post deployment stage, it provides real-time monitoring and reporting of jitter, delay, and packet-loss on a per call and site-to-site basis, detect and report current network problems, predict potential network problems in advance, and provide real-time performance troubleshooting so the carrier can allocate resources to the issue on a timely basis.”
“One of the first actions we took as part of our turnaround at Verso was to get closer to our key customers,” said Mark Dunaway, Verso President and Chief Operating Officer. “Being selected by this large North American carrier to provide a VoIP solution is a direct result of our working closer with them. Further, being selected by this customer in particular is further strong validation of our VoIP product and solution capabilities.”
For more information, contact Verso at www.verso.com or call 678.589.3500.tag: VoIP business, VoIP News, Verso Technologies, VoIP services, Tier One carrier,
The Talk Forever Home Phone plan, according to T-Mobile, is for those who want to keep their home phone but say goodbye to their expensive home phone bill. If you want to have this service, you will need the following:
• Existing high-speed Internet connection
• T-Mobile Wireless Router with Home Phone Connection
• A qualifying T-Mobile mobile voice rate plan and the Talk Forever Home Phone plan
• A compatible home phone: 5.8 GHz cordless phone(s) or traditional touchtone phone(s)
Yes that is T-Mobile VoIP service that is offered for $10 a month. T-Mobile customers can add a new T-Mobile home phone line that includes unlimited nationwide calling, true Caller ID, voice mail and many other included features.
HotSpot @HomeSM Talk Forever Home Phone plan is currently available for our customers in Seattle and Dallas only.
Learn more here, at T-Mobile.
tag: VoIP, T-Mobile, Cheap VoIP, Talk Forever Home Phone plan, HotSpot @Home, HotSpot at Home
BELLEVUE, Wash., and BERWYN, Pa. – Feb. 22, 2008 – T-Mobile USA, Inc., and SunCom Wireless Holdings, Inc. (NYSE: TPC) today announced the completion of T-Mobile’s acquisition of SunCom Wireless. This means more than 1.1 million customers in North Carolina, South Carolina, Tennessee, Georgia, Puerto Rico and the U.S. Virgin Islands will be able to benefit from T-Mobile’s award-winning customer service and the unique products and services offered by T-Mobile.
The acquisition enhances T-Mobile’s network coverage. The company now serves 98 of the top 100 markets nationally.
“We look forward to offering our new customers in the Southeastern U.S. and Puerto Rico T-Mobile’s industry-leading customer service, quality national network and unique personal communications products and services that help customers stay connected to those who matter most,” said Robert Dotson, president and chief executive officer of T-Mobile USA.
T-Mobile recently earned the highest ranking in the J.D. Power and Associates Wireless Customer Care Performance Study, the seventh consecutive period it has held the top spot.
T-Mobile and SunCom Wireless announced their merger agreement in September 2007. The deal received Federal Communications Commission approval February 8.
Through this acquisition, T-Mobile USA expects to significantly expand its national network to cover 259 million Americans, an increase from 244 million. T-Mobile USA also expects to realize synergies with a net present value (NPV) of approximately $1 billion through reduced roaming and operating expenses. Plus, the company anticipates further upside growth opportunities through the addition of the new markets.
tag: VoIP business, mobile Voip, SunCom Wireless Holdings, T-Mobile USA,
Friday, February 22, 2008
AsteriskNOW 1.0.1 my favorite Asterisk appliance (in software form) since Asterisk@home deviated from the path and began to spy on us, unsuspecting Trixbox users without declaring so. (I have replaced all my Asterisk@home and Trixbox installations, closer to 20 to AsteriskNOW, since that incident even though they have said that it was corrected.) I am also testing out PBX in a Flash by Nerd Vittles. I do not mind using or testing Trixbox on my experimental rigs but when it comes to people who trust me, my customers, I want to be able to say that my work is clean. And that I have given them the best possible solution.After the spying trouble, I had to go to all those people, apologize for giving them faulty solutions and install AsteriskNOW, which was in beta state then.
Since the AsteriskNOW 1.0.1 was released, I have upgraded all my installations to newer version and all are happy campers now. But due to the needs of some of them and support needs, I will have to convert some of them to full blown Asterisk and hand them over to better capable people for support.
Most of the installations are used to VoIP enable legacy PBXs and as VoIP solutions for small ventures that spawn all over bay area. But these startups change for better and have upgraded with Asterisk Appliance from Digium and 3Com. Some have opted to use Asterisk Business Edition, the commercial counterpart of AsteriskNOW.
I have some of these installations also in some of the high end educational institutions and two in very far away lands that I have not physically seen since 2005.
What else could you use AsteriskNOW for totally hip voice mail system with IVR functions. May be for that conference system with developers and clients strewn all over the world.
Yes all those are very possible and have been made really ease with the AsteriskNOW 1.0.1 release.
Since the last beta release there were some addons and updates were added, although not an exhaustive list following stand out the best;
- Over 1000 updates and improvements to the GUI
- Updated to Asterisk release 1.4.17
- Updated to Zaptel release 1.4.8
- Updated Linux Kernel 2.6.22
- Polycom phone auto-provisioning
- Improved package management and update capabilities
- Open Settlement Protocol
- The phone provisioning module
tag: Asterisk, AsteriskNOW, AsteriskNOW 1.0.1, Trixbox, Asterisk@home, Asterisk appliance, Digium, IVR functions, voice mail, VoIP olutions, PBX, IP PBX,
Thursday, February 21, 2008
FreePBX 2.4.0 was released a few weeks ago and promise to bring about a whole bunch of changes and addons to FreePBX. Many of us have been testing for a while but the numbers keep on going up according to the FreePBX site post. The team is also carrying on with the FreePBX training program. The team is currently in discussions to see if we can bring you the next event in Las Vegas, NV the week of April 8th, 2008 and Philippe is asking your opinion in a form of a response to this post announcing the success of FreePBX 2.4.0.
The list of features, enhancements and bug fixes are too long to list here but some highlights include:
Simultaneous support for Asterisk version 1.2, 1.4 and 1.6 (best effort for 1.6 as it is still in beta)
New Voicemail Blast Group Module.
New Language Module.
Vast improvements and many feature enhancements to Paging and Intercom Module.
Internal improvements and several new features to Queues Module.
DUNDi™ Trunk support with automatic dialplan integration.
Vast improvements to handle Zap Channel Inbound routing and incorporating Zap Channel + CID routing abilities.
System wide extension and destination registry and automatic conflict and integrity detection.
New Custom Apps module to integrate custom applications and extensions into the above registries.
Vast improvements to Devices and Users mode especially around adhoc devices.
Call Confirmation support for hunt ring strategy in Ring Groups and Follow-Me.
tag: Asterisk, FreePBX, FreePBX training, FreePBX 2.4.0
Positron Public Safety Systems Inc. (PPSS), an IPC company, announced today that it has won an award from San Mateo County, California Public Safety Communications (SMCPSC) to upgrade their center to full digital services; a baseline system that is Next Generation 9-1-1 ready. As per the award, Positron will deploy and service VIPER™ and its Power suite of products, providing VoIP-based E911 services to 23 public safety agencies dispatched by SMCPSC.
SMCPSC was impressed with Positron’s proven reliability in supporting their E9-1-1 system, having used multiple Positron products for more than a decade. While applying a rigorous competitive procurement process to determine the best solution, SMCPSC considered numerous factors such as Positron’s reputation, ability to meet wireless Phase II requirements, ease of training and from a user’s perspective, seamless migration from an analog to a digital platform.
“The Positron products have served the County of San Mateo well over the last 8 years and we are very pleased with our choice to bring this new product line into our Dispatch Center, bringing us up to state-of-the-Art functionality”, stated Jaime Young, Director of the Office of Public Safety Communications.
“Leveraging on a 10+ year relationship with SMCPSC, we are honored to bring to our customer, the best VoIP technology available in the industry”, said Darrin J. Reilly, President of Positron Public Safety Systems. “This award as well as numerous other recent wins is a testament of our unwavering commitment to provide large and small customers, indispensable communications for their Next Generation E9-1-1 needs. Our VIPER platform has gained remarkable momentum across numerous states and we are pleased to be the vendor of choice for SMCPSC, a long standing Public Safety agency within the State of California.”
About San Mateo County Public Safety Communications (SMCPSC)
SMCPSC serves all branches of emergency first response including law enforcement, Fire and Paramedic services, in a consolidated operation. The center serves 23 agencies, fields over 270,000 calls per year and employs a staff of 58. The center employs state-of-the-Art technology and is committed to maintain a system availability rate of 99.9% per year. SMCPSC has received numerous performance awards, notably “Center of Excellence” from the National Academy of Emergency Dispatch. To find out more about SMCPSC, visit www.smc911dispatch.org.
tag: voip news, voip, San Mateo County, Public Safety Communications, smc911dispatch, SMCPSC, VIPER platform, Positron Public Safety Systems, VoIP technology, E911,
trixbox CE 2.6 Beta is the latest version of trixbox and is ready for you to download at Sourceforge. This release comes with CentOS 5.1 as the OS and Asterisk 1.4 as the engine, trixbox CE 2.4 is the best release of trixbox ever!.
18.104.22.168 Second beta of 2.6 Cleaner modules Embedded FreePBX Improved Asterisk Info module.
tag: trixbox, Asterisk, freepbx, IP PBX, VoIP
Wednesday, February 20, 2008
TMCnet has added a "Making Money with VoIP – Real World Examples" a free white Paper to it's white paper library. Just fill a simple form to get your copy.
tag: VoIP White Paper, VoIP business, Making Money With VoIP, Free VoIP White Paper
Tuesday, February 19, 2008
mindSHIFT Technologies, a leading provider of managed IT services and VoIP solutions, has earned the Cisco Advanced Unified Communications Specialization, which certifies expertise in the design, installation, configuration, operation, and maintenance of VoIP systems.“This important Cisco certification reflects mindSHIFT’s deep expertise in managing all aspects of the customer environment, both remotely and at customer premises,” said Bill Webb, VP of Product Engineering and Development at mindSHIFT. “The advanced VoIP certification marries our understanding of LANs and connectivity with expertise in Cisco VoIP solutions, enabling us to deliver end-to-end telecom and IT support for our clients.”Unified Communications, the technology mindSHIFT uses in its Dedicated VoIP solution, is Cisco's complete portfolio of Internet Protocol (IP) communications products and services.“Our law firm clients value the productivity advantages gained by the mobility and unified messaging provided by a Cisco solution,” said Michael Savino, General Manager of Professional Services at mindSHIFT.mindSHIFT, a Cisco Premier Certified Partner, has also earned the Express Foundation Specialization. This technology specialization is specifically designed to focus on the customer network infrastructure and network capabilities.For details on managed VoIP solutions from mindSHIFT, visit http://www.mindSHIFT.com/Products-and-Services_Managed-VoIP.aspx.
tag: VoIP Certification, VoIP business, CISCO, mindSHIFT, Advanced Unified Communications Specialization, VoIP systems, Cisco Express Foundation Specialization.
Enterasys Networks Inc., the Secure Networks Company™, announced today it is the first networking company to complete certification for ShoreTel’s Technology Partner Program. The ShoreTel Technology Partner Program provides market and innovation leaders the opportunity to expand their reach by connecting directly with ShoreTel's growing community of channel partners and customers.
“ShoreTel and Enterasys have proven themselves in our operations,” said Bob Glaze, Chief Technology Officer for the City of Oakland, California. “We have found them to be one of the best and simplest choices to use for unified communications.”
Security is among the top concerns of enterprises deploying voice over IP (VoIP) systems today. Enterprises need to detect unauthorized use of VoIP systems, prevent service disruptions and eavesdropping, and monitor for new threats against the voice infrastructure—all while ensuring reliability and quality of service. Rather than approaching VoIP security in a haphazard fashion, Enterasys recommends an intelligent, automated and integrated approach to enable and secure wired and wireless VoIP communications.
“As we celebrate our 25th anniversary in the networking industry, we’re proud to be a ShoreTel Certified Technology Partner,” said Mike Fabiaschi, Enterasys President & CEO. “Enterasys has developed a comprehensive approach to discover, classify, prioritize and secure the industry-leading unified communications solutions from ShoreTel.”
“With Enterasys, our customers will now have a tightly integrated security solution,” said Mark Arman, ShoreTel VP of Business Development. “Networking and security are clearly converging to proactively protect unified voice, video and data communications.”
Enterprises worldwide want to ensure the same reliability, quality, manageability, mobility and security of the traditional PBX with VoIP and unified communications solutions. The Enterasys Secure Open Convergence solution delivers a way to sense and automatically respond to security threats against the IP telephony infrastructure, enforce network access control policies, and comply with regulations for monitoring and safety such as CALEA and E911 in the United States.
Enterasys offers the ability to centrally manage a set of communication policy rules which can restrict the type and amount of network traffic directed at specific voice-related servers. Also, policy rules can be enforced to ensure that critical services are not “spoofed” causing service disruption or theft of critical communications. In addition to proactive security of VoIP resources, events occurring on the network that can compromise the voice service can be detected, and ultimately mitigated in order to preserve voice availability. Advanced Enterasys Dragon® security applications have the ability to detect specific threats against the voice infrastructure by using a strong set of VoIP attack signatures that include decoders for H.323, H.245, Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP). Once threats against voice services are detected, NetSight® automated security management can be used to locate the exact source of the threat and take appropriate action to resolve the issue.
The Enterasys Matrix® and SecureStack™ switches are network switching and routing devices which can automatically discover, classify and prioritize ShoreTel unified communications systems and traffic. Embedded security, priority and bandwidth control is provided for every user, device and application without sacrificing performance. The Enterasys standards-based open architecture interoperates with the existing infrastructure so enterprises can leverage the advantages of ShoreTel solutions while assuring reliability, service quality, manageability, mobility and security.
ShoreTel enables companies of any size to seamlessly integrate all communications – voice, data, messaging – with their business processes. Independent of device or location, ShoreTel’s distributed software architecture eliminates the traditional costs, complexity and reliability issues typically associated with other solutions.
tag: voip news, VoIP business, Enterasys, ShoreTel, City of Oakland, VoIP, CALEA, E911, Secure Open Convergence, H.323, H.245, SIP, MGCP, NetSight, SecureStack,
Monday, February 18, 2008
GIPS announced recently that its award winning communications platform, REX is available to all of us. The GIPS REX Solution consists a set of communication tools for PCs (REX PC) and mobile devices (REX Mobile), as well as an SDK for application developers.
REX provides following features such as:
- Advanced Presence
- PC to PC calling
- PC to outside number calling (via PBX or gateway)
- Instant Messaging
- Patented Firewall/NAT traversal technology
The REX Solution includes REX PC and REX Mobile for enterprises and service providers, and the REX SDK for application developers.
“We are proud to recognize the greatest achievements in the advancement of VoIP and IP telecommunications technologies,” said Rich Tehrani, TMC President and Editor-in-Chief of INTERNET TELEPHONY magazine. “GIPS continues to deliver cutting edge VoIP technology and the REX softphone has the high-quality VoIP processing that companies have come to expect from GIPS – that’s why we awarded it the annual product of the year award.”
“Enterprises and developers of next generation services are always looking for new and innovative ways to keep people connected while providing productivity tools for the mobile workforce of the 21st Century,” said
REX PC turns any computer into a corporate PBX extension, keeping the functionality of a desktop phone in place, with additional features for enhanced communications. Enterprise users can call or IM colleagues from an internal contact list, or dial outside numbers. Remote workers can place and receive high-quality calls while traveling outside of the office, without the need to establish a VPN connection.
The REX Mobile softphone allows service providers to customize and deploy a branded design of their choice. With the same features as REX PC, users are able to establish calls in any area with a wireless Internet connection. GIPS’ voice processing technology overcomes quality challenges within wireless networks and gives users the reliability they expect from voice calls.
Along with REX PC and REX Mobile, GIPS is offering an SDK for application developers to build a voice or chat application. The REX SDK is a flexible solution that delivers the key building blocks developers need to create a softphone or add voice, IM or PBX functionality into an existing application.More information on REX can be found at: http://www.gipscorp.com/rex/rex.html.
tag: VoIP, softphone, GIPS, REX Solution, REX PC, REX Mobile, PC to PC calling, Instant Messaging, Voicemail, Firewall/NAT traversal
Speakeasy, a bestbuy company and where I usually go to test my broadband speed, has launched its Integrated Voice offering, a service that is cheaper than replacing traditional business phone lines and also an alternative to a rip-and-replace upgrade that many other companies are offering.
SEATTLE, February 5, 2008—Speakeasy (www.speakeasy.net), a Best Buy company, (NYSE:BBY) and one of the nation's leading providers of voice, data and IT solutions, is expanding its product offerings with Integrated Voice - a more cost-effective replacement for traditional business phone lines. Speakeasy's new Integrated Voice business phone lines are ideal for companies that have between two and 12 telephone lines and want to keep their existing voice equipment while reducing the cost of their phone services. Pricing begins at just $19.95 per line, with long distance charges of just $0.029 per minute, saving many businesses up to 35% on their phone lines.
This phone line replacement combines voice and data services over a T-1 or high-speed DSL Internet connection, where bandwidth is dynamically allocated between voice and data for best voice quality. Integrated Voice is a convenient, cost-effective solution that allows customers to work with their existing PBX or key system and broadband connection. Customers can choose Speakeasy broadband service with voice quality technology, or use their own broadband connection. This service is available nationwide, and provides substantial savings over competing Integrated Access solutions. Speakeasy developed this solution in response to the needs of small businesses for flexibility, choice and a simplified solution for their business communications.
"We have a long history of listening to our customers and innovating around their needs," said Bruce Chatterley, Speakeasy's president and CEO. "With the addition of Integrated Voice, we can now offer customers a choice between a service that works with existing phone equipment, or a full-featured next-generation hosted voice service."
Integrated Voice is available now nationwide. To learn more, visit: www.speakeasy.net or call 1-800-556-5829.
tag: VoIP business, SMB, speakeasy, bestbuy, PBX, IPPBX, high-speed DSL, Integrated Voice
According to Reuters article Research In Motion is sueing Motorola for non payment of royalties on patents held by RIM. RIM also alleges that Motorola is charging exorbitant royalties on patents owned by Motorola.
"The civil action, filed on Friday in U.S. District Court for the Northern District of Texas, alleges that Motorola infringed on a number of patents held by RIM.
In addition, RIM alleges that Motorola "is demanding exorbitant royalties...for patents that Motorola claims are essential to various standards for mobile wireless telecommunications and wireless computing that RIM practices."
This includes technology that allows mobile telephone handset users to use Wi-Fi, RIM said.
At the same time, Motorola is refusing to acknowledge or pay royalties for certain patents held by RIM, the BlackBerry-maker said."
tag: VoIP Patents, WiFi, RIM, Motorola, WI-FI, patent lawsuit
NY Times has a nice article on iPnone in China. Specially the edge enjoyed by China Mobile has with it's 400000 plus iPhone users. These iPhones are unlocked and loaded with Chinese iPhone software.
On the other hand this underground operation that might account for missing iPhones in apples accounting might also blow a hole in Apple accounting for as much as $1 billion over the next three years, according to some analysts.
Perhaps it is not too late to change the marketing plans.
tag: iPhone, iPhone China, China Mobile, missing iPhones, unlocked iPhone, Apple
Thursday, February 14, 2008
Android, the open-source mobile platform from Google, was shining at this week's World Mobile Congress in Barcelona, Spain, where some prototypes were displayed to attendees. Also a new Software Development Kit (SDK) for Android was released on Wednesday.
Google Developer Advocate Jason Chen, wrote on the Android Developers Blog, that the newest SDK, called M5-rc14, provides updates to the application programming interfaces (APIs) and developer tools.
"There are a couple of changes in m5-rc14 I'd like to highlight:
- New user interface - As I mentioned when we introduced the m3 version of the Android SDK, we're continuing to refine the UI that's available for Android. m5-rc14 replaces the previous placeholder with a new UI, but as before, work on it is still in-progress.
- Layout animations - Developers can now create layout animations for their applications using the capabilities introduced in the android.view.animation package. Check out the LayoutAnimation*.java files in the APIDemos sample code for examples of how this works.
- Geo-coding - android.location.Geocoder enables developers to forward and reverse geo-code (i.e. translate an address into a coordinate and vice-versa), and also search for businesses.
- New media codecs - The MediaPlayer class has added support for the OGG Vorbis, MIDI, XMF, iMelody, RTTL/RTX, and OTA audio file formats.
- Updated Eclipse plug-in - A new version of ADT is available and provides improvements to the Android developer experience. In particular, check out the new Android Manifest editor."
Go get your Android
On-Demand IP Telephony Solution helps the Salvation Army Spread its Forces
Sydney, 12 Feb 2008
Integ Communications, a leading Australian integrator of communications technologies and a UXC company, today announced it has won the contract to deploy its ‘Telephony as a Service’ solution (iTaaS) for The Salvation Army Australia Eastern Territory (TSAAET).
Launched in September this year, iTaaS is an on-demand, IP telephony solution. It uses a hosted delivery model to give organisations access to the most up-to-date IP Telephony offerings at a fixed monthly rate and is customisable to suit an organisation’s specific business requirements.
The six-year contract was awarded to Integ via a competitive tender process under Group Communications Tender (GCT). GCT brings together a consortium of partners who supply telecommunications goods and services at competitive rates, to organisations across Australia.
Integ and TSAAET will roll out iTaaS across its 50 major sites and 800 terminal sites throughout New South Wales, Queensland and the ACT, replacing the organisation’s legacy PABX systems.
Under the iTaaS model, the hardware and software components of the TSAEET telephony solution remain the property of Integ and are housed in a secure data centre. The IP handsets, remote media gateways and passive communications servers will be located at TSAAET's territorial headquarters in Sydney and divisional headquarters in Brisbane as well as across its 50 major sites (average of 18 users per site) throughout NSW, QLD and the ACT.
“The telecommunications network that links our divisional headquarters to offices in our regional and metropolitan centres is critical to our operations, enabling us to provide the superior levels of care and assistance Australians in need, deserve,” said Wayne Bajema, IT Manager, TSAAET.
“When it came to upgrading our existing telephony infrastructure, we knew an IP-based solution would provide the collaboration capabilities, stability and flexibility we were looking for. The challenge lay in finding the internal skills required to manage this technology as well as in the upfront cost required for capital purchase and implementation.
“Integ’s iTaaS solution offers us the best of both worlds. We have access to the most up-to-date telephony solutions immediately without making a significant upfront investment. We also have a team of experts from Integ that manage those components of the solution that we can’t handle on an ongoing basis.
“iTaaS also gives us the flexibility to incorporate other IP-based applications such as unified communications quickly, easily and cost effectively as the organisation expands” Bajema explains.
Services and applications delivered to TSAAET include:
· LAN switches
· Call servers
· IP Media Gateway; and
· Unified Communications
“Integ’s new hosted approach offers organisations that have IT resource and budget limitations, the opportunity to experience the significant cost and productivity benefits that IP telephony delivers,” said Ian Poole, CEO, Integ Group.
“Integ’s new delivery model is about offering increased choice and flexibility for organisations,” Poole said.
Monthly fees will be paid by TSAAET based on the number of users accessing the hosted solution. The per-user fee structure incorporates both the hosted and premise-based hardware and software elements, as well as management and administration of the system for the period of the contract.
“Integ’s iTaaS solution not only meets our needs in terms of providing the IP Telephony infrastructure we need when we need it, but the hosted model also ensures we know how much it’s going to cost us each month, helping us track and manage our costs,” Bajema said.
tag: voip news, Integ Communications, On-Demand IP Telephony, Salvation Army, Telephony as a Service, iTaaS,
Wednesday, February 13, 2008
T-Mobile has entered into a new strategic partnership with Yahoo to introduce a range of mobile services across Europe.
The agreement supplants Google as T-Mobile's mobile search services provider in Europe, and sets the stage for Yahoo oneSearch to become the exclusive mobile search service for T-Mobile customers from April.
Read more at vnunet.
tag: T-Mobile, Yahoo mobile, google,
Monday, February 11, 2008
The creator of Proof of concept tool for VoIP Security that I wrote about a while ago, Peter Cox, together with Stuart Morrice has launched a new company UM Labs, to provide effective security that he has been preaching for a while now. From the press release below it is evident that we will be seeing more of UM Labs and the products for long time to come. I for one, will seek a gateway that the company is bringing to the market.
London February, 11th 2008, a new company, UM Labs Ltd, has been created to address the growing need for effective security for Voice over IP (VoIP) and Unified Messaging (UM) Security. The founders of UM Labs are experienced Internet Security professionals, Peter Cox and Stuart Morrice. The adoption rate of VoIP and UM is growing; fueled by the promise of greater flexibility, integration with other applications and more cost effective service delivery. However, concerns over of the security of these applications are preventing many organizations from fully adopting the service and failing to realize the potential benefits. These concerns are founded partly on confusion over the scope and types of security threat that face the applications and partly over a lack of easy to use security products.
UM Labs was founded to address this problem by providing a range of easy to use, cost effective products that deliver effective security for VoIP and UM to users ranging from small business and branch offices through large enterprises to service providers and telco operators. Peter Cox CEO of UM Labs commented, “The security model applied to many VoIP networks is one of isolation, physically separating Voice and Data or using VLANS to keep them apart and preventing any external IP connections. Unfortunately separation sacrifices many of the benefits of VoIP and makes Unified Messaging and the integration of all communication applications impossible.” “Until this problem is addressed, VoIP networks will not deliver their full set of potential benefits. The first step to solving this problem is to recognize that VoIP is a specialist application requiring a specialist approach to security. Standard Firewall products do not do a good job at securing VoIP. At best they complicate the deployment of VoIP at worst they present so many barriers that it is virtually impossible to deliver a VoIP service without compromising the security of other applications.” “It should come as no surprise that the best way to secure applications such as VoIP and UM is to deploy a specialist security gateway. Both web and email have spawned their own security markets, in each market there are a number of products delivering security controls that standard Firewalls cannot deliver.”
The goal of UM Labs is to enable each VoIP network to delivery its full potential set of benefits by providing products that implement a more realistic security model. This security model recognises that VoIP networks need external connections to enable SIP trunk connections and to safely extend the service to home workers, roaming uses and branch offices.
To reach this goal, UM Labs will launch a range of security gateways through 2008. Each product will secure VoIP applications based on the Session Initiation Protocol (SIP) and other UM applications. The first release is designed to secure remote VoIP connections to home users and roaming workers and to provide security for SIP trunk services. Security controls include firewall grade IP level security coupled with application specific security controls designed to combat threats such as VoIP call hijacking, call flooding and unauthorized call monitoring. The latter is provided by VoIP encryption services using the industry standards TLS and SRTP with a choice of key management algorithms to support the widest possible range of hardware and software phones. Supported key management algorithms include Phil Zimmermann’s ZRTP and SDES as used by Snom and other phone manufacturers.
UM Labs Website
tag: VoIP Security, VoIP, Unified Messaging Security, Peter Cox, VoIP networks, VLANS, SRTP, SIP trunk, ZRTP, SDES, Snom, VoIP encryption, VoIP call hijacking,
Users in over 200 countries can now take advantage of EQO’s low-cost mobile calling
Barcelona, Mobile World Congress, 11th February, 2008, EQO Communications (www.eqo.com), today announced the expansion of its award-winning low-cost mobile VoIP service to over 200 countries. This now extends the availability of EQO’s calling services from its original 30 countries to over 200 countries including India, Russia, the Philippines and Brazil.
People from around the world have already been enjoying EQO’s free application that brings free international texting as well as free instant messaging to mobile phones. With this recent service expansion, all EQO users across the globe can now use EQO to make cheap calls. In addition, all EQO calling users will now have access to EQO’s presence feature, which displays in real time when their EQO friends are online and available to receive phone calls.
“In many countries, mobile calling rates are still ridiculously high,” said Bill Tam, EQO’s CEO. “We wanted to break through the cost barriers by providing an easy, cost-effective option for everyone to stay in touch with friends and family in more places around the world.”
EQO is free to download your mobile phone, and is a truly mobile application that does not require a computer, WI-FI connection or new contract. With EQO, users can chat on the go for free using MSN, Yahoo, Google Talk, QQ, AIM, ICQ and Jabber, or call or text anyone in the world at super-low rates. Users can also invite their friends to join EQO, and call them at 50% off EQO’s regular calling rates or message them for free*.
EQO works with a broad array of handsets from major manufacturers such as Nokia, Samsung, Motorola, Sony Ericsson, LG, Blackberry and HTC, bringing the benefits of VoIP, IM, chat and social networking to regular, everyday mobile phones. EQO’s application is available in multiple languages including English, Russian, Portuguese, Italian, German, Spanish, French, Mandarin and Cantonese.
* Please note that minutes and data from mobile plans are utilized when using EQOtag: mobile Voip, EQO, MSN, Yahoo, Google Talk, QQ, AIM, ICQ, Jabber, social networking, low-cost mobile calling