Wednesday, February 28, 2007

San Francisco wants more with free wireless.

VoIP IP Telephony @
I have reported this before under the following post;
VOIP IP Telephony: San Francisco's Google Free Wireless finally on the go and now I hear that Board of Supervisors are putting holt to Mayor Newsom's plans to give free wireless internet access to the masses in San francisco.
In April, San Francisco chose its plan from among six. Atlanta-based EarthLink would create a wireless network and charge customers $21.95 a month. Mountain View-based Google planned to rent space on the network and offer a slower, ad-supported version for free.
Now after more than two years later, with Newsom having signed a contract with the Internet providers in January, the Board of Supervisors last week declined to even consider the deal. The board, whose approval is required, decided instead to investigate turning the project into a city-owned public utility.
Elsewhere in the press reported that;
"Google is considering targeting ads by location, so, for example, someone in Union Square searching the Web for a shoe store might see offers for nearby shops first.

But the American Civil Liberties Union, the Electronic Frontier Foundation and other advocates raised concerns about EarthLink's privacy policy.

They also complained that Goggle's ability to track the whereabouts of network users could prove irresistible to law enforcement (Google said people worried about such things could sign up using false names).

Additionally, some citizens of this high-tech city aren't willing to settle for just any wireless connection, even if it's free. EarthLink's paid service is about three times faster than Goggle's free one.

The speed of the free service "is so 1997, said Ralf Muehlen, a software developer who operates a small free wireless network here and wants the city to push EarthLink for better technology. "I'm a techie. It's too slow for me." "

Gizmo Client 3.0 talks to most IM clients,

VoIP IP Telephony @
One of my favorite VoIP clients, Gizmo Project has released a new client, version 3.0.
With the new client, Gizmo project brings you;
Drum roll.................
# It’s FREE!
# FREE call's to any GizmoProject, Yahoo! Messenger, Google Talk, or Windows Live user
# Free Voicemail and Conference calling.
# Super clear call quality
# Inexpensive add-ons that let you make and receive calls from any mobile phone or landline
So if you are wondering where my Skype Credits went (It is connection fee you **&%%*)
it is time you at least try this out. In addition Gizmo client, one also could use Gizmocall;
VOIP IP Telephony: Use Gizmocall to call me or anyone free
to complete a call when the client is not handy at your hands.
Gizmo is available for Windows, MAC and Linux as well as for Nokia N800, N80 and Nokia 770. Follow the links for downloads.

Gizmo Project

Metaswitch Conference Server UC9000 Streamlines Conference Call Set Up, Management and Control

VoIP IP Telephony @

According to a news release by businesswire, New SIP-based Application Server Enhances Robust UC9000 Unified Communications Portfolio, Creates Incremental Revenue Stream For Service Providers.

COMPTEL Plus Spring 2007
Booth 719

LAS VEGAS--(BUSINESS WIRE)--MetaSwitch, a leading vendor of next-generation carrier switching and application solutions, announced today the availability of a new component in its industry-leading UC9000 Unified Communications application suite. The UC9000 Conference Server combines standards-based SIP conferencing control with a reservation-less “meet-me” conference application and an easy-to-use web-based management interface. Capable of supporting conference calls of up to 500 participants, the MetaSwitch UC9000 Conference Server offers the industry’s most flexible, real-time conference capabilities for service providers and their enterprise customers.

UC9000 Conference Server delivers a full-featured, SIP-based conference solution that enables moderators to easily set up conferences via an intuitive web interface, and control participant interaction and call flow via the phone interface or through the web. Advanced features include:

* Lock/roll call/number of participant controls
* Dynamically updated participant list in real time
* Visible indication of current speaker and audio levels
* Ability to mute, gain control and disconnect any participant
* Call recording.

As part of the UC9000 Unified Communications application suite, the new conferencing capability integrates seamlessly both with MetaSwitch’s softswitch platform and other UC9000 enhanced services including unified messaging, auto attendant, music on hold, and privacy defender. Together, these features allow service providers to create value-added packages, generate significant incremental revenue streams, and enhance customer loyalty.

“MetaSwitch continues to advance the art of Unified Communications with dynamic products, features and functionality that address the market demand for value-added integrated services,” said Andrew Randall, vice president of marketing at MetaSwitch. “The UC9000 Conference Server will be used by our customers to deliver a robust managed service to the carriers’ communications arsenals as they compete for business in an increasingly competitive market.”

Please visit MetaSwitch at CompTel Spring 2007 at booth 719

BusinessWire News release
Comptel Spring 2007

What does JOOST mean, the upcoming IPTV service?

The meaning of Joost as mentioned on WeSeePeople. Funny!

WeSeePeople: What does JOOST mean, the upcoming IPTV service?

Tags: ,

Tuesday, February 27, 2007

Skype goes Pro in Europe

VoIP IP Telephony @
Skype has announced that it is ready to go pro in Europe. The initial release will not cover the all of EU but will extend to other countries later in 2007. This must be a good move since Europeans have started to be online more than before. Here is an excerpt from the news release. (For the complete press release follow the link provided at the end.
From Skype news release;

LUXEMBOURG, February 20th, 2007 – Skype™, the global Internet communications company, today announced the launch of Skype Pro. Skype Pro is a new Internet communications package offering zero cents per minute calls to domestic landlines along with a series of premium Skype features and discounts on Skype Certified™ hardware.

Skype’s popular features such as video calls from one Skype user to another, sending instant messages, transferring files, conference calls for up to 10 participants or joining in Skypecasts (live moderated conversations with up to 100 people) remain free to all Skype users across the world. Skype users can also use Skype to make free calls from one Skype account to another.
Skype Pro’s subscription package includes:

* Zero cents per minute calling to domestic landlines (previously €0.017* per minute)
* Free Skype Voicemail (normally €15* per year)
* €30 discount on SkypeIn™ numbers
* €5 Skype Credit included as part of the introductory offer (see below)
* A €30 discount on a Philips VoIP 841 cordless phone
* A €10 discount on an SMC WiFi phone
* Additional discounts on a series of Skype Extras are also available including desktop sharing, avatars, emoticons and ring tones

*15% VAT is added where applicable and all Skype Pro calls are subject to a small connection fee of up to € 0.039 per call.

For a limited time only, you can purchase Skype Pro on a five month basis for €10 and receive €5 Skype credit absolutely free. After this introductory period, customers can continue with their subscriptions for just €2* a month.

Skype Pro will initially be available across Europe in the following countries:

* Austria, Belgium, Denmark, Estonia, Finland, France, Germany, Ireland, Italy, Netherlands, Sweden, Norway, Portugal, Spain and the UK

Complete Skype news release

Saturday, February 24, 2007

Off to the mountains

VoIP IP Telephony @
See you people in about a week, there are some good snow!

Friday, February 23, 2007

Deltathree gets go2call

VoIP IP Telephony @

deltathree, Inc., a leading provider of SIP-based Voice over Internet Protocol (VoIP)
solutions for service providers and consumers worldwide, today announced that it acquired the service provider and consumer businesses of, Inc., a privately-held U.S.-based VoIP solutions provider. deltathree signed a definitive agreement to purchase these businesses from Go2Call on February 17, 2007. The transaction closed on February 19, 2007.
Under the terms of the agreement, deltathree acquired Go2Call's consumer and service provider businesses as well as key components of their regional infrastructure worldwide. With the addition of Go2Call's customer accounts, deltathree strengthens its service provider and consumer channels while expanding its presence to new regions of the world.

Get the full press release, follow the links below;
PR Newswire press release

Free VoIP Personal Incoming SIP Number(s) from Tpad

VoIP IP Telephony @

According to another Blogger, I did not check carefully enough. He has commented on this post. Please follow the link to get more information. You can also check the links to this post. In case you can find neither, click here ;)

I have a new Tpad number, a SIP number now, that could be used from anywhere in the world. I will not be releasing the SIP number itself but I will tell you how t get one for yourself. Follow the links at the end of the post.
While most other VoIP companies charge yearly for a personal SIP number, Tpad is offering this service for free and will provide as many as the customer wants at no extra charge.
Steven Johns, Marketing Manager for Tpad, said: "We believe that it is unfair to charge people just for a number to receive calls -- that's why we give all our customers a free number.
"We understand that each member of a household may want their own unique number so we offer as many numbers as the customer wants." is establishing itself as one of the market leaders because of its innovative service and unique offers that make using the internet for phone calls fairer and cheaper for its users.

Once you signup you will get the following too;
More Benefits - Free Voicemail / Call Forwarding / Instant Call History / No Annual Contracts / Per Second Billing / Free Incoming Tpad Number / Free Softphone / Earn Credit Feature /Pay as you Go Credit

Coming Soon - Free Video Calls / SMS Text Messaging / Enhanced Instant Messaging compatible with MSN -Yahoo - AOL and Tpad Networks

Tpad home
Press release Free VoIP Personal Incoming SIP Number.

Thursday, February 22, 2007

Skype 3.1 Beta and SkypeFind! Skype wants to be a search engine

VoIP IP Telephony @

Being done with your VoIP solution, Skype now reaches into other nooks and corners in your world. Skype has done a lot with the Skype 3.1 beta. SkypeFind is one of them.
SkypeFind is one of the most interesting features that skype ’ve done in quite a while now according to a post. Skype calls it “Local businesses you like”, and that’s what it is - a collection of businesses, with reviews and comments, built by everyone using Skype.
Well there are more to skypefind and here some of them according to the same blog post;

ADD and EDIT listings, OK, that should be a part of it without being able to add some thing to the list.

Review and comment on listings, good you need that, democracy wins

Find a Business! any business. OK we get the drift, Skype wants to be a search engine (Google?)when it grows up! Await the RSS feeds, mapping and such features soon. Follow the links to get the complete story with screen captures etc.

Skype 3.1 beta for windows
Hey we did the Skype directory first (A claim on one of the comments)

Cisco and Apple are iPhone buddies

VoIP IP Telephony @

“Cisco and Apple today announced that they have resolved their dispute involving the iPhone trademark. Under the agreement, both companies are free to use the iPhone trademark on their products throughout the world. Both companies acknowledge the trademark ownership rights that have been granted, and each side will dismiss any pending actions regarding the trademark. In addition, Cisco and Apple will explore opportunities for interoperability in the areas of security, and consumer and enterprise communications. Other terms of the agreement are confidential.”

–Joint press statement February 21, 2007
Which leads me to think of future conversations in the market place and else where.
So can your iPhone do wi-fi?
How come your iPhone is so ugly? Hello,
I want an iPhone. OK, Apple or Cisco? Apple of course you dumdum!

Then again Apple might have another law suit over iPhone as Quantum Technologies is examining the touch sensor technology used in iPhone. But I want my iPhone and the only thing that might hold me back is cingular, that already does not exist!
iPhone sensor technology

Wednesday, February 21, 2007

Verizon, Vonage Patent suite, chimes trouble for VoIP

VoIP IP Telephony @
In U.S. District Court in Virginia, Verizon will face off against Vonage in a patent infringement case by Verizon that could lead to ripple effects in the VoIP IP Telephony market.
The lawsuit charges Vonage with violation of seven Verizon patents, all related to Vonage’s VoIP based commercial service. If Verizon wins, Vonage could be forced to pay royalties or find workaround technologies.

The lawsuit charges that Vonage infringed patents related to completing calls between VoIP users and the PSTN or the public telephony network, authenticating VoIP callers, validating VoIP callers’ accounts, monitoring the usage of VoIP callers, protecting against fraud, providing enhanced features, and using Wi-Fi. I have not seen all the patents but might be a task for PUBPAT look into.

All the patents involve technologies at the intersection of the public network and VoIP applications, where all digital communications turns in to analog.

In July 2006, Vonage said it had acquired three VoIP-related patents from Digital Packet Licensing that address compression techniques related to the public network.

The patents were part of Vonage’s overall plan to find technologies that it could employ to work around Verizon’s contested patents.

But the Holmdel, New Jersey-based company continues to maintain that Verizon’s lawsuit is an attempt to snuff out VoIP companies that have been taking away its customers.

Asbury park press

Tuesday, February 20, 2007

Viacom in the Juice, Joost

VoIP IP Telephony @

I find from from multiple news sources that Joost has inked a deal with Viacom for video distribution.

Motley Fool says that Joost will protect Viacom's copyrights as well as give the media giant as much as two-thirds of the ad revenue. Even though YouTube has struck revenue-sharing deals with content providers in the past, the soft policing of infringing uploads has been frustrating to some entertainment companies.

May be Joost wil have enough money that they will buy unyielding Viacom some day! (That is my prediction, not motley fool's.)
Then the CNET news tells us that The deal is limited, at least at first: many of Viacom's most popular programs, such as Comedy Central's The Colbert Report and South Park, will not be available initially. Some of the featured offerings, however, will be MTV's My Super Sweet 16, Comedy Central's Freak Show, BET's American Gangster, as well as feature films from Paramount and its related brands.
OK, I am waiting to see those stuff on Joost.
I really like what I have seen during my testing of Joost. And looking forward to see more sooner.
Folks, Joost is getting better, as the time and software releases passes.

Motley Fool's view of Joost and Viacom
CNET's Joost news

Sunday, February 18, 2007

One of the threats to VoIP

VoIP IP Telephony @

I was at a site looking for information on sustainable development for one of my other blogs, Solarion, when I noticed an article about The threat of listening in on VoIP phone calls via ARP attacks.
What are ARP attacks?
ARP Guard - Protection from ARP Spoofing Attacks

I have no affiliation's to the company provided in the link

Saturday, February 17, 2007

New Joost applications (Mac and PC) out.

VoIP IP Telephony @

New Joost applications (Mac and PC) out.

If you managed to get your invitation to the Joost (I know some of you did get them) have new Joost applications now. So if you were waiting for the MAC version, now it is the time act. I am going to try it on my MAC Mini Intel. It has 2GB memeory, 160GB hd and the rest is stock. Of course it is connected to a 23" LCD and to a 1080P 40" TV to watch movies. Should be fun.
Following are the information of the application versions; Not all info, if you are a member, you can see it all at Joost site.

Windows version Beta 0.8.0

* Search has been given the sprite treatment, improvements include info panel stays open as the results are scrolled through.
* Channel Catalogue is improved and now includes all Channels targetted to the location of the viewer.
* Duration of each Show is now on the info panel for the show as well as the EPG.
* Trivia tracker introduced, delivers timed messages on supported shows.
* Improvements generally in delivering video, sprite performance, etc.

Mac OS X version

* First beta release

* Includes all 0.8.0 functionality.

What SER is and isn't

VoIP IP Telephony @

Sys Admins view of SER guts!
The following information is from SER site news report, which in turn plucked from a discussion on a Developers mailing list. It is so important so that I have entirely reproduced the post, for my own reference.
"Consider a more simple SIP proxy like repro. All you can do there is start the damn thing and give it the user data (what would be subscribers, aliases, and parts of the usr_preferences in SER 0.9). Sounds all nice and simple.

Now, as an VoIP operator, my world will be a little bit more complicated. I may have different services that run on separate proxy farms. I may have interesting add-on services like call forwarding, voicemail, IVRs, whatever else product management comes up with. Somewhere in a dark corner, I have some PSTN gateways or, instead, I have an agreement with some telco to do that for me.

If you draw this, you'll get at least half a dozen boxes with weird connections between. If this doesn't scare you, start sketching the call flows. You will suddenly find little funny quirks, that of course you can put into C code but if why? SER provides you with the opportunity to solve pretty much all of them in a very simple language.

Better yet: You write your script, you do a test call. If it doesn't work, you make a trace, you fix your script and try again. No compiling, no packaging, just a restart (BTW, something for the wish list: reloading the config on a SIGHUP). Another trace, another round.

Now we fast forward a bit. Your system is running just fine. But one of your PSTN interconnect partners updates their software and -- surprise -- all the calls to them fail. Sure, you could use another partner. But your friends in billing will tell yet that their prices for some destinations are just insane. We _really_ have to have that first partner.

Sure, you quickly figure out what the problem is. Sure, you call them and try to explain to the unfortunate fellow on the other end how SIP works and why their stuff isn't really SIP. Sure, after a while they give in and promise to fix it. But can they do that quickly? Nope. They have to go and talk to whoever delivers their software.

Half a year passes and nothing much happens.

Now, with SER all I need to do is find the route for the specific partner, do the magic with subst() and maybe some other horrible things and voila, it works. Everyone is happy. And should the partner actually ever get their stuff fixed, I can just remove those three lines I had to add.

With repro, things would have been quite different. I have to know enough C++ to actually grok their design or have to have someone doing this. Implementing the three line fix, testing it, producing it easily takes a man-day. With SER I did that in three minutes. Including
the test call.

What it comes down to is, that there is no universal thing. For NATi, there isn't six funny devices that you find a work around, report to the good folks at iptel, who then add another flag. NAT routers change with every software revision. Old things go away, new things pop up. It is your responsibility as a provider to stay close. That's what people pay you good money for.

Sure, SER is hard to get into as a beginner. If you want to stay a beginner and don't care about SIP, use repro. It'll probably work for you out of the box. If you expect to have to do more, invest the time, learn SIP, learn the ser.cfg. It will pay off later. Everything will be "SER gut" (Sorry, that just had to happen)."

SER Home

Friday, February 16, 2007

OpenSER v1.1.1 is out and Available in Binary

VoIP IP Telephony @

This release includes all fixes and corrections made to OpenSER since July 2006. The configuration file and database structure are fully compatible with v1.1.0, this release being a new packaging of CVS branch rel_1_1_0. It is recommended to update any v1.1.0 to v1.1.1.

The sources, binaries and packages for different OS distributions and architectures can be downloaded from:

The binary packages will be uploaded as soon as they are submitted by public contributors. You will find OpenSER packages in several Linux distributions packages for SUSE, Fedora and Mandriva and more are available in a RPM repository.
Click here for the link to the repository.
OpenSER 1.1.1 is now available in binary format for several embedded platforms, courtesy of Ovidiu Sas:
- Linksys NSLU
- Synology DS-101(g+)
- Iomega NAS 100d
- Freecom FSG-3
- Qnap TurboStation
And all routers supported by the oleg or dd-wrt distributions. Check out for yours.

Opera Mobile, Jajah beats MS to MS Mobile 6.0 platform for VoIP

VoIP IP Telephony @

Some of the first Windows Mobile 6.0 phones to hit the market will include Opera Software ASA's mobile browser, in addition to Microsoft's browser.

Motorola's Q as well as phones from High Tech Computer , Asus and Toshiba will include Opera Mobile, Jon von Tetzchner, co-founder and CEO of Opera, said at the 3GSM World Congress in Barcelona.

Microsoft launched Windows Mobile 6.0, the latest version of its mobile phone operating system, this week. Internet Explorer Mobile, Microsoft's browser, comes with the software. A number of handset makers announced this week plans to release phones running the Microsoft operating system. according to Reseller News.

In addition to Opera Mobile, the company also offers Opera Mini, a free browser that works on most Java-enabled phones. It communicates with a server hosted by Opera or a mobile operator that strips down web sites for quicker, less data intensive mobile browsing.
Jajah, a web based VoIP service provider, inked an agreement with Opera in 2006 April. So my guess is you may be trying Mobile Voip on your slick new cell phones soon.

Opera Mini
Jajah Voip mobile plugins

Adobe buys Antepo, a presence/Voip company

VoIP IP Telephony @
- -
Adbobe recently acquired Antepo, the maker of Rivoli Presence XMPP/SIP server software.
Rivoli features native support for the Session Initiation Protocol (SIP) and the Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE). The company is extending that support to include VoIP and presence integration with VoIP.
Antepo extended, in Rivoli its support for Extensible Messaging and Presence Protocol, (XMPP), which is supported by Jabber and clients such as GoogleTalk.
So now Adobe is going to have a presence for it's products, mainly acrobat reader. Here is the info you get if you try to go to

"Adobe is pleased to announce that it has acquired Antepo, Inc. Antepo is a technology company that developed the Antepo Open Presence Network (OPN) System — an award-winning platform for Enterprise Instant Messaging and Presence capabilities — enabling real-time communication and collaboration while meeting critical business requirements for control, security, integration, and compliance.

The Antepo technologies and expertise acquired will support the development of Adobe's products and solutions for knowledge workers. The addition of Antepo's Presence and Enterprise Instant Messaging solutions will further expand the capabilities of the Adobe® Acrobat® software family for enabling knowledge workers to communicate and collaborate with confidence."

Antepo's new home

Thursday, February 15, 2007

Secure multi-lateral VoIP peering software published to Sourceforge as Open Source

VoIP IP Telephony @

Atlanta, Georgia (USA) –– Feb 14, 2007. TransNexus, Inc. has made the OSP Toolkit and RAMS open source projects publicly available on SourceForge. The OSP Toolkit is a client side implementation of the OSP peering protocol. The OSP Toolkit, written in C, is a mature open source project begun in 1999 and has been integrated into numerous commercial and open source VoIP products. The RAMS OSP server is a java based OSP server developed for testing and as a reference implementation.

The Open Settlement Protocol (OSP) is an IP Operations and Billing Support Systems (OSS/BSS) protocol defined by the European Telecommunications Standards Institute (ETSI TISPAN), OSP is officially known as ETSI Technical Specification 101321 for inter-domain pricing, authorization and usage exchange. OSP is unique because the way it uses PKI services to enable secure peer to peer communication between VoIP networks. “The OSP protocol was developed to enable direct multi-lateral peering among VoIP networks. OSP provides secure inter-domain access control and eliminates costly network bottlenecks”, stated Richard Brennan, Chairman of ETSI TISPAN Next Generation Network Architecture working group.

RAMS is a java based server useful for managing inter-domain VoIP routing, called number translation and Call Detail Record collection. RAMS supports the European Telecommunications Standard Institute's OSP Peering protocol (ETSI TS 101 321).

The OSP Toolkit and the RAMS OSP test server is available at links provided below. A free version of the TransNexus commercial OSP based peering server is available at

The OSP Toolkit
RAMS server

British Telecom give VoIP to Mobile users, with smartphones from HTC

VoIP IP Telephony @

Speaking on the announcement, Jerry Thompson, BT’s Chief of Applications and Devices said: “We are dedicated to providing our customers with a truly converged mobility experience. We are very excited about Windows Mobile 6 and the range of HTC devices as it will enable us to develop affordable converged services for customers over our wireless broadband and mobile networks. This is the beginning of an exciting journey for our customers as such services will allow people to stay on top of their jobs whilst away from the office.”

“We are bringing together our products and services to connect people simply and seamlessly and will look to bring the benefits of Mobile VoIP services to businesses. Our vision is that our customers are able to access all their applications and information on their choice of device, using the best network available with a simple and seamless user experience. It is also consistent with our strategy of providing customers with whatever they wish, either through our own expertise or through working in partnership with world class companies – in this case Microsoft and HTC – to deliver the best possible customer experience.”

BT has also pioneered the development and growth of wireless broadband connectivity across the UK and beyond to help realize the full potential of these new applications and devices; with almost 1 million Wireless “hubs” in homes and offices, access to over 2000 premium location BT Openzone hotspots in the UK and Ireland and over 30,000 hotspots globally. The company’s plan to build a first phase of 12 Wireless Cities by the end of March 2007 means that many of the UK’s major city centres will soon offer widespread Wi-Fi coverage.

BT Brings Mobile VoIP to UK Customers with HTC Smartphones

Wednesday, February 14, 2007

VoIP IP Telephony is the Blog of the day at REDORBIT

VoIP IP Telephony @

I was pleasently serprised to find that "VoIP IP Telephony" is the blog of the day at RedOrbit.
If you are wondering who the heck is RedOrbit, here is their description from the about us;
" is committed to providing stimulating, original content and presentation, with over 500,000 pages covering the vast ideological spectrums of space, science, health, and technology. The beautiful and engaging forum created at promotes a friendly and open environment, enhancing user loyalty and community, while advancing RedOrbit's goal of providing the world with a virtual Utopia for intelligent, curious minds. averages over 5 million unique visitors per month. With subject matter a bit more intellectually oriented than most, the average visitors tend to be well educated, between the ages of 25 - 55, with a median income significantly higher than that of Internet users as a whole.

RedOrbit has positioned itself perfectly to excel in the current Internet climate and well into the 21st Century. "

So I am pretty happy about it.
Follow the link to see it in it's own glory.

RedOrbit Blog of the day (If visited today, valentines day)
Any Other day

Skype gives a lame answer for bios dump saga and You can, remove skype, if needed

VoIP IP Telephony @

The much debated about Skype BIOS dump that many people has written about, including me;VOIP IP Telephony: Skype dumps you BIOS info and sends home!
Has brought out an answer from Skype. Sorry we did it for DRM! Bummer!! (it is still called intrusion) The users found out. I found out at VOIPSA blog (and VOIPSA is still more kind to Skype than me. So if you feel bad about what I write, read his article! a link to the article is below), that Skype has provided a answer and a solution. But it is still lame.
It would have been much cleaner had you mentioned that before users found out. Remeber SONY root kit saga. People never seem to learn.
Kurt Sauer also provided the simple solution - upgrade to the latest Skype 3.0 version,, where they now use a version of this framework that no longer reads the BIOS.
So Kurt, What DO you read now?

Anyway if anyone feels that they need to remove skype, I have given a link to one of my former articles below.
Also another method is to run it on a Virtual Machine like I do. So Kurt what did my reading give you?
May be running it as a portable application might confuse Skype, run it at least on two three computers. Follow the links.

VOIPSA's "How to avoid Skype 3.0 reading the BIOS of your system"

Remove Skype or Uninstall Skype

Skype's explanation about BIOS dump

Skype's promise that they seemed have forgotten

Time zone adaption for Asterisk

VoIP IP Telephony @

Matthew Gast over at Linux Journal (a magazine that I have read from the first issue, I still have the first issue!) has written an excellent article on time Zone adaption for Asterisk. (May be for other systems that are based on Asterisk).

The system Matthew built on top of Asterisk to handle time zone feature has two major parts. The key to the system is maintaining a time-zone offset from the time in London. (His code implements offsets only of whole hours, though it could be extended to use either half or quarter hours.) When a device first connects to Asterisk, its IP address is used to guess the location and, therefore, the time offset. After the offset is programmed into the system, incoming calls are then checked against the time at the remote location. Before the phone is allowed to ring, the time at the remote location is checked, and callers can be warned if they are trying to complete a call at an inconvenient time.
The currently published part one of the paper carries one through four sections;
Step 1: Estimating the Time Zone
Step 2: Confirming the Time-Zone Information
Step 3: Letting Callers Know the Time
Step 4: Using The Project
These steps includes a lot of information and the code that is used in each step is provided as a single tar file so one could get started right away.

Time Zone adaption for Asterisk, article and resources

Tuesday, February 13, 2007

WorxBox an independent TrixBox?

VoIP IP Telephony @

I was reading Sinologic article that lead me to this discovery. It seems that the TrixBox replacement have come to the open source arena. The AsteriskNow, TrixBox and the WorxBox have a lot in common.

See the following by the news releases at WorxBox project site which I have provided links below.
"Welcome to Worxbox®, The Asterisk Server with the Worx®! The goal of the Worxbox® project is to provide a production quality Open-Source Unified Communications Server that's easy to build, configure and manage. WorxBox® installs all the software needed to deploy a fully functional Asterisk® based PBX and creates a basic configuration which can then be easily optimized to meet the requirements of almost any business.

WorxBox® is a no-compromise, next-generation phone system that can scale to support several hundred users, but it doesn't stop there. Worxbox® also integrates Asterisk® with a comprehensive suite of supporting and complementary web, messaging, networking and business productivity modules. In fact, for many small businesses Worxbox® may be the only server they need!"

The functionalities also shows some similarities;
WorxBox® Features:

* Complete CentOS Linux operating system
* MySQL Database
* Fully-featured Asterisk PBX
* SugarCRM
* Alfresco Document Manager
* Joomla! Web Content Management System
* Apache Webserver
* Zimbra Collaboration Suite
* DHCP, SMTP, FTP server
* Unified Messaging (E-Mail, Voice-Mail, Fax-Mail)
* Shorewall Router and Firewall (Requires 2 or more NIC)
* Webmin Server Managment
* Lots of Nifty Extras!

Easy Administration:

* Web-based Management of Server and PBX Functions
* Secure Remote Administration
* Centralized provisioning and management of IP telephones

So now you have three choices, WorxBox, AsteriskNow and TrixBox.

Links; article
WorXBox Project
Sourceforge WorxBox site

Vonage lets you say "Be my Valentine" in VoIP or PSTN way!

VoIP IP Telephony @

St. Valentine day is at your door step, and Vonage is offering the chance to let lovebirds greet their otherhalf with a Vonage-gram message and a poem that is personalized to the reciever. This includes a recorded message with some special content.
All you need to do is dial 1-700-Valentine and choose from a number of love poems and can also add a thirty second personalized message which will be delivered on Valentines Day. Your love wishes can be sent to any number in U.S or Canada or it can be even delivered to your e-mail account from where you can personally forward it to your sweetheart. The service starts on February 10th for both U.S and Canadian customers. Wish your loved ones this Valentines Vonage style but I would prefer calling personally to say I Love You in my own way!!

The original article appeared on the voip weblog

Native SIP in an iPAQ, iPAQ 51X

VoIP IP Telephony @

The HP iPAQ 500 series Voice Messenger features voice over Internet protocol (VoIP) capabilities, "push" email and the latest Windows® Mobile 6 operating system. The phone includes Wi-Fi, GSM/EDGE, and Bluetooth connectivity options.It's small, measuring just 4.7 by 2.8 x .7 inches and 5.8 ounces. The phone is topped by a 176 by 220 screen, which is lower resolution than the 320 by 240 screens on many other smart devices. It features six hours of continuous talk time on a fully charged battery, HP said.
The HP iPAQ Voice Messenger also leads the competition in battery life. As the first in HP's new smartphone lineup, the HP iPAQ 500 series helps highly mobile professionals stay connected wherever they are.

Another feature is that more than 20 voice commands available on the iPAQ let customers have hands-free operation. Using a powerful "voice reply" feature, people can reply to email by dictating and sending a voice response, without the need for any typing. Users also can listen to email and text messages, navigate through phone and calendar tasks and speak to start applications.

Additionally, the HP iPAQ Voice Messenger has built-in Wi-Fi to provide business customers a VoIP alternative to traditional office phone setups. By integrating the HP iPAQ Voice Messenger with office phone systems, businesses can eliminate the need for desk phones and benefit from streamlined communications and reduced IT management.

The VoIP is not a third-party client like Skype for Windows Mobile, but a SIP-based VOIP solution built right into Windows Mobile 6. It will work with any SIP-based corporate IP PBX or public service, HP announced.

HP news release

Monday, February 12, 2007

Trixbox 2.0 in CentOS VE, Virtual Environment

VoIP IP Telphony @

I wrote about OpenVZ on the gridtech blog; Link is below,
It is a virtual environment for Linux, like vmware or Xen. But today I noticed that their is an article on OpenVZ wiki, how to run Trixbox 2.0 on this environment.
The wiki article goes to expand on following topics;
1. Create VE from CentOS Template.
2. Add character device /dev/tty9 to VE
3. Install rpmforge & apt
4. Add DAG repository
5. Install yum
6. Install speex and libspeex-devel
7. Download Trixbox 2.0 tar.gz file (not ISO) and extract to /var/trixbox_load
8. Change to /var/trixbox_load and run script
So if you are looking to test trixbox in a virtual environment or want to test OpenVZ, follow the links below.

OpenVZ main site
Trixbox in a CentOS VE
gridtech: OpenVZ, another player in virtualization space

Hi-res Video hack for Skype (640X480)

As you all know, Skype communicates in 320X240 in video mode. Well it looks pretty small even in my notebook (1400X900). But it all changed today after reading Make Blog article about Jaanus Kase's experimental mod for Skype. Now I have 640X480 video in my Skype calls.
This works on both MAC and PC.
So how one needs and how one does this? well for actual instructions, follow the links below. But first thing is you need a web cam that supports 640X480 and at least 128K upload speed on your broadband connection. Once you have this, you can go into modify some files on your skype directory.

Make Blog article about Skype Video

Skype mod for video PC
Skype mod for video MAC

Asterisk 1.2.15 and Zaptel 1.2.13 released

VoIP IP Telephony @

If you are a Asterisk 1.2 track user, there is a new release out now. Asterisk 1.2.15 that brings a bunch of fixes and improvements. Some of the notables are;
* Support for Zaptel-based transcoder hardware, initially the Digium TC400B 92/96 channel transcoder.
* Handling of voicemail subdirectories when using ODBC storage has been improved, so that messages can be forwarded properly.
* A problem with forwarding voicemails from folders other than the user's INBOX has been fixed.
* The Zaptel channel driver can now support echo cancellers that provide 64ms or 128ms of echo cancellation per channel.

There is also complementary release of Zaptel 1.2.13. In addition to bug fixes, an important performance improvement for most Digium cards, and support for new Digium hardware and some significant improvements in the XPP driver for Xorcom's Astribank hardware is provided by this release.

* A modification was made to the drivers for all Digium PCI cards to improve their compatibility and performance when used in interrupt sharing environments.
* Support for the Digium TDM800P 8-port analog interface card was added.
* Support for the Digium TC400B 92/96-channel transcoder card was added.
* Support for the Digium High Performance Echo Canceller add-on software module was added.
* All drivers updated to Linux kernel 2.6.20 API changes.
* Performance improvements for multiple Astribank units.
* Astribank firmware protocol version is now 2.4.
* Astribank now supports Message Waiting light on analog telephone sets.
* Added a /proc interface to blink the leds on the Astribank to identify ports in large setups.
* fxotune is now supported by Astribank.

All users of Zaptel 1.2.x are encouraged to update to this release as soon as they can practically do so.

Asterisk 1.2.15 download
Zaptel 1.2.13 download

A new CEO for Digium and Mark is CTO now

VoIP IP Telephony @

Digium went under some changes this week and many a sources have reported about the new CEO for Digium, Danny Windham. Instead of going in to lengths about this I will just state what mark Spencer had to say;
"When Danny comes on board, I will be transitioning to the role of Chief Technical Officer (retaining my position of chairman of the board of directors), providing strategic vision for the company as well as being able to focus more extensively on the community, the customers and the technology.
My sincere hope is that this transition will not only directly benefit the Asterisk community and Digium customers, but will allow me to spend much more time with the community and with Asterisk, playing a more important technical role in our roadmap for both hardware and software.".
As far as I am concerned, it is a welcome change.

Friday, February 09, 2007

Skype gives you 24 min to call China, Hong Kong, and Taiwan for new year!

VoIP IP Telephony @

Skype is ringing up Chinese new year with a free gift to families and friends enjoying the dawn of the year of the pig.
In celebration of the Chinese New Year, Skype announced that its users in the United States and Canada can make free calls to China, Hong Kong and Taiwan, through March 8.

Launching today, the 8-8-8 promotion allows U.S. and Canada-based Skype users to redeem twenty-four minutes of SkypeOut credit to call landlines and mobile phones in China, Hong Kong and Taiwan, and wish friends and family abroad a Happy New Year. Such calls normally cost $.021 a minute plus a $.039 connection fee. Skype to Skype calls anywhere in the world remain free.

Skype Chinese new year offer

Simens C455 IP released with invitation to join

VoIP IP Telephony @

The Gigaset C455 IP makes it easier to join VoIP or Internet Telephony according to Siemens. This new phone is configured to use new VoIP services, such as an on-line directory or, a service by Siemens. Any Gigaset Phone buyer could join the Gigaset community free. The phone calls between Gigaset users are free through this service. This hybrid telephone, C455 IP, with integrated answering set has both, POTs and VOIP connections. One can connect a phone line and a LAN connection to the phone.

A PC is not needed for the operation of the phone. One POTs connection and two VoIP connections could be made at the same time. It also has the capabiity to support up to six VoIP service providers. The integrated directory stores 100 contacts.
The answering machine has 30 minute capacity and could be protected by PIN and accessed from anywhere, like in a traditional answering machine.

In adition to, in Germany this hybrid telephone supports also the access to the klickTel on-line telephone, a industry book also over 30 million entries.
The Gigaset C455 IP is available starting from March 2007 in back color at the price of 109.99 Euro

Siemens so far has the information in German and I was not able to find the English information on the phone.

Simens Gigaset C455 IP

Thursday, February 08, 2007

Skype dumps you BIOS info and sends home!

VoIP IP Telephony,

According to a post appearing on pagetable describes how skype program called locatd in your computers;
“\??\C:\Documents and Settings\Myria\Local Settings\Temp\12\”;
might be sending your BIOS information to server somewhere in Skype. Here is a snippet from the original authors publication, to which I have provided information below under links.

"An unreadable executable file coming from Skype sounds interesting, so I look at it. It’s 46 bytes long. For copyright reasons I can’t post the file or a complete disassembly. However, I can describe the program in terms of 16-bit DOS C:

int main(void)
fwrite((const void far*) 0xF0000000, 1, 0xFFFF, stdout);
fwrite((const void far*) 0xF000FFFF, 1, 1, stdout);
return 0;

It’s dumping your system BIOS, which usually includes your motherboard’s serial number, and pipes it to the Skype application. I have no idea what they’re using it for, or whether they send anything to their servers, but I bet whatever they’re doing is no good given their track record."

Well be it Skype, Microsoft or anyone sending back my information without my knowledge is not a good thing. So may be people might have to visit "remove Skype".

Pagetable "Skype reads your BIOS"

Remove skype from my own article;
VOIP IP Telephony: Remove skype, stop skype or detect skype with skypekiller

Talkswitch gives a hybrid PBX for SMB's, TS-244vs

VoIP IP Telephony @

Talkswitch has released a new addition to its pbx, IPPBX line. The new hybrid 244vs. I have deployed one of their systems (not 244vs) with Polycomm phones and I and the users are very happy with it.
If you need to provide VoIP service for SMB, Small Business, I would suggest that you have a look at these systems. It is not the perfect solution but it will get you there without much hassle. Unless you have resources to manage an IPPBX like Asterisk or Trixbox.

I learned about this at VoIP News.
The TalkSwitch 244vs is a full-featured hybrid VoIP/traditional PBX telephone system that includes everything you need to sound big, control your communication costs and stay connected everywhere.
2 telephone lines (POTs lines)
4 VoIP trunks
4 local extensions
10 remote extensions
VoIP and traditional: TalkSwitch gives you both as the 244vs is a hybrid system, with capacity for VoIP and traditional telephone connections. You can save with VoIP services and VoIP inter-branch connections without sacrificing your connection to the traditional telephone network. It's the best of both worlds.

Remote extensions keep you connected, if your work doesn't stop at the office, and neither does TalkSwitch. Any phone, anywhere, can be added as a remote extension of your system. Cell phones, home phones and phones in other offices can be reached with simple 3-digit dialing, and calls can be seamlessly transferred and screened. With remote extensions, your customers, employees and partners can always stay connected.

Unlike most PBX systems, TalkSwitch doesn't tie you down to proprietary telephones. TalkSwitch systems work with standard analog phones and selected IP phones, so if you already have telephones, chances are they'll work with TalkSwitch. If you do need phones, check out our line of analog telephones tailor-made for TalkSwitch.

Talkswitch Press release
VoIP News article

Wednesday, February 07, 2007

POTS/VoIP Line Card for Multi Service Access Platform, from ZyXel

From ZyXel News
VoIP IP Telephony
LAKE BUENA VISTA, Fla.February 6, 2007
Efficiency and cost savings are two of the main drivers for the transition to VoIP. As service providers are evaluating the timing and benefits of such a move, ZyXEL Communications today at the NTCA Expo 2007, announced the VOP1248G, a POTS/VoIP line card. Designed to ease the transition to VoIP networks, the new card features a media gateway that converts analog voice to VoIP, thus eliminating the need to install special VoIP phones or an analog telephone adapter (ATA) on customer premises. Now, service providers can offer their customers next generation voice services while saving on costs.

ZyXEL’s new POTS/VoIP line card works in conjunction with ZyXEL’s IES 5000 and IES 5005 chassis-based Multiple Service Access Platforms (MSAP) and is targeted to telcos looking for an easier way to offer the benefits of a VoIP architecture to its customers. The VOP1248G is a 48-port module card that enables wireline carriers to provide lifeline telephone service to their residential and business subscribers. ZyXEL’s IES 5000 and IES 5005 MSAPs provide support for ADSL 2+, G.SHDSL.BIS and even VDSL 2 line cards, and are ideal for telcos who would like to offer multiple services over the same box.

“ZyXEL is the fourth largest IP DSLAM vendor in the world,” said Munira Brooks, senior vice president of sales, marketing and business development. "This VoIP line card solution allows service providers to offer feature rich telephone services to their customers and do it economically by smoothly transitioning to an IP based network.”

The new POTS/VoIP line card supports all network signaling protocols, digital signal processing and an IP packetization mechanism based on open standards that include SIP and H.248/Megaco for broad range interoperability with leading softswitch manufacturers. Additionally, the up-to-date QoS function guarantees the service quality of voice packets. The VOP1248G works in conjunction with ZyXEL’s IP MSAP’s, the IES-5000 and 5005, and co-exists with any broadband DSL line card in the main chassis. The IES-5000 and 5005 support strong IP QoS functions based on IEEE802.1q to guarantee and prioritize different types of traffic including voice, data or IPTV.

Voice Services Supported by the VOP1248G include:

* Emergency / 911
* Do Not Disturb
* Selective/Anonymous call rejection
* Call waiting
* Call hold
* Music on hold
* Call transfer (blind and attended transfer)
* Call return and call back on busy
* Off hook warning tone
* Three way conference


The VOP1248G POTS/VoIP line card is now available to service providers for field tests and includes a 45-day free trial. For more information on the VOP1248G or the 45-day free trial, please call 1-800-255-4101.

ZyXel News release

Load Balancer for Asterisk and VoIP, from VocalScape

VoIP IP Telephony,

VocalScape Networks announced today that they have released a load balancer for VoIP IP Telephony usage.
The Vocalscape Load Balancer began as an open source project which was adopted and improved upon by Vocalscape. It was made compliant with Asterisk, a popular open source PBX, and the algorithm was revised to more evenly distribute calls. Previously, the Load Balancer would send calls to a primary server and only when the primary server was overburdened would calls be sent to additional servers. The new algorithm balances the load by evenly distributing the calls between the servers. As an additional benefit, the Load Balancer provides failover capabilities. If a server is not responding, the Load Balancer will route all calls to servers that are functional.

"Vocalscape has developed the Load Balancer to meet our customers' needs," commented Ron McIntyre, President of Vocalscape. "As our customers grow their user base, they will need to add additional servers to handle the higher volume of calls. The Vocalscape Load Balancer will allow them to evenly share the load among multiple servers."

The earliest known (to me) SIP load balancer was at May be this is the one they improved. Vovida Org was one of the pioneering VoIP opensource sites that became less functional. Follow the links to get an idea of how it was like, to develop and write VoiP applications in those days (Ha! it was only 5-6 years ago).

SIP Load Balancer at
Yahoo news

Tuesday, February 06, 2007

JaJah flake at Pageflake, New widget at Pageflakes

Voip IP Telephony,,

JaJah, the web based internet telephony service which I have written about before;
VOIP IP Telephony: Disruption in VOIP IP Telephony Field By Jajah! Or is it?
is turning a new page. A new page Flake to be exact. By collaborating with Pageflakes there is now a Jajah Falke that you could add to your numerous flakes at Pageflakes.
Basically it is a good idea to add it to you web site or blog if you want people to call you directly, if you know how to add flakes to your site!
Also there is a Mac OS X Dashboard widget created by a Jajah fan Greg Smithies. The widget is called Jajah Dialer that allows you to make calls from a Mac Desktop. Both are good tools if your phone rings through Jajah

JaJah flake at Pageflakes
Jajah telephony service
Mac OS X jajah widget

FreePBX back online with a new Server
A few days ago I asked to help out FreePBX group because they were having trouble with their server. They have got a new server and the system is fast according to Rob.
But they are about 1/3rd short of cost of a years worth of hosting. Please visit FreePBX and donate if you are a user of FreePBX. If you are an Asterisk user, I am sure FreePBX will help you out a lot. Have a Look, Follow the links.

FreePBX is, to quote FreePBX's About page, a Standardized Implementation of Asterisk that gives you a GUI to manage your system. If you’ve looked into Asterisk, you’d know that it doesn’t come with any ‘built in’ programming. You can’t plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about. FreePBX simplifies this by giving you a pre-written set of dialplans that allow you to have a fully functional PBX pretty much straight away."

New server anouncement
FreePBX Donations page
FreePBX Bounties Page
Blog article

GEM for Networks, Skype Manager to be released

VoIP IP Telephony
FaceTime Communications has announced software that will make Skype easier to manage. The company will release GEM (Greynet Enterprise Manager) 3.5 next month.
FaceTime Enterprise Edition creates a framework for authorized access to IM and other greynet applications outside the perimeter, coupled with a mechanism for detecting and preventing rogue or unauthorized access. The result is comprehensive protection without compromise on cost, performance or operational considerations. With the recent addition of Greynet Enterprise Manager (GEM), enterprises can now manage security policies and aggregate reporting for IM, P2P, web conferencing and VoIP chat threads across distributed enterprise environments. GEM extends enterprise anti-spyware to include targeted remediation and repair of infected endpoints, without the need to deploy client software.
Here are some of the functions of GEM;
Comprehensive Coverage

* Widest support of IM applications – public and enterprise IM networks, and professional community networks
* Extensive controls over the productive use of P2P and VoIP applications such as Skype, as well as WebEx web conferencing
* Proactive protection against a wide range of spyware and other inbound malware threats


* FaceTime Security Labs is the world's largest greynets research operation
* Global team of researchers plus a two-million-strong user community and a worldwide honeypot network
* Sponsored by industry leaders, including Symantec, McAfee, Sophos, Microsoft, Yahoo, VeriSign, and Webroot

Identity-based Management & Compliance

* User/group level policy based access control and management
* File transfer archival support for EIM networks
* TrueCompliance™ supports the strictest interpretation of compliance regulations and e-Discovery requirements
* Tamper-proof archival and integration with enterprise messaging archival compliance solutions
* Seamless integration with all major corporate directories, email and WORM storage systems

Read more about GEM here

VoIP/SIP enabled GSM base station announced

VoIP IP Telephony, (

TECORE Wireless Systems announced the launch of its pre-IMS VoIP/SIP enabled GSM base station to be showcased at this year's 3GSM World Congress. The BTS-4000RM delivers up to 64 high powered GSM/GPRS/EDGE voice and data channels occupying as little as 5U of a standard 19" telecommunications rack.
The BTS-4000RM base station eliminates the need for traditional BSC, TRAU and PCU elements, embedding these functions within the base station platform. The BTS-4000RM may be integrated directly with TECORE's SoftMSC® as well as other mobile core network solutions over IP, TDM, or IP over TDM. Moreover, the BTS-4000RM is enabled to support TECORE's exclusive c® Backhaul Free(TM) base stations, allowing operators to deploy 'plug and play' coverage extensions without the need for wired backhaul links. Due to its small size and flexible interface options, the BTS-4000RM is also ideally suited for remote towns and villages that may require satellite backhaul as well as other applications where macro RF coverage and performance is needed.

With the launch of the BTS-4000RM, TECORE also introduces network interface enhancements to its GSM base station product line. This includes the internal translation from a traditional GSM RAN to an all IP SIP-enabled network interface. Thus, the BTS-4000RM supports direct connection to a pre-IMS network. Added capabilities include transcoder-free operation to minimize network bandwidth, and direct routing of local IP call traffic to optimize overall backhaul utilization. Specifically targeting applications requiring compact radio access solutions, the BTS-4000RM offers flexible network integration and deployment capabilities while providing macro RF coverage, capacity and performance.

“The operators to whom we’ve delivered and demonstrated the BTS-4000RM are very excited about the flexibility this solution offers,” said Terry Williams, Chief Technology Officer for base station products. “Large capacity, flexible interfaces, and more functionality in a smaller, easy to deploy platform are key to many specialized mission critical applications, and this platform delivers.”

“We are very pleased with customer response to this new product development and technology innovation,” said Doss McComas, Vice President, Business Development at TECORE. “This product builds upon our company’s ongoing commitment to advanced, smaller, more powerful base station solutions for the market.”

The BTS-4000RM will be on display at the 3GSM World Congress 2007 in Barcelona, Spain, February 12-15, 2007, at the TECORE Wireless Systems pavilion located in Hall 8, stand 8C78.

News source TECORE Wireless

Monday, February 05, 2007

Asterisk Boot Camp Madrid

(If you see this post on an automated SPLOG, original post is is at SNAPVOIP)

From March 26th to 30th, in yet to be confirmed location and facility, Avanzada and Digium will conduct next Boot camp in Madrid. Associated with this boot camp is a chance for Asterisk and VoIP technical personnel to acquire Digium Official dCAP (digium Certified Asterisk Pofesional) Certificate.

Avanzada 7 Boot camp is an intensive course on Asterisk Technology conducted in in Spanish where one will learn to handle most important details and aspects about open source Asterisk technology.

Avanzada 7 & Digium launch Asterisk official training program for Spain, Portugal and South America.
Follow the links for appropriate information.

Avanzada 7 Boot camp
Avanzada 7 Boot camp in English
Digium dCAP
The original post at Sinologic where I found the information.

Saturday, February 03, 2007

Splogs and Feed test for feed spamming

Feed test for feed spamming.
Splogs such as voipniche steals posts via rss feeds. Now I have set my feeds to be short. All Bloggers who has splog problems could test this. In addition to short, one also could disable feeds for a while.

Splog problem is described somewhat at wikipedia and is a good starting point.

If the splog is at Blogger, one could follow this link to get information
Blogwerx sentinel allows you to monitor who steals your has an article on sentinel.
Fight splog has a lot of information as well. From fight splog site;
From JenSense:

If you see a site violating the AdSense terms, you can now file an anonymous spam report that will get to the quality team for checking. To file a report, you simply go to the page that is showing AdSense ads and click on the "Ads by Google" (or "Ads by Goooooogle") link. In the form on the next page, include the term "spamreport" and put in a short reason about why you feel the site is violating the AdSense terms or policies. You can also enter your own email address, if you wish, then click submit.

ebiquity has a very good article on the subject, "Pings, Spings, Splogs and the Splogosphere: 2007 Updates" I say it is a must read. The article is rich with links, data (graphs).

Friday, February 02, 2007

VoIPniche is still at it stealing posts

As this post mentioned;
voipniche steals for adsense
still continue to steal posts. Looks like the guy do not care. In case it is needed, I have captured what is needed.


Voipniche steals for Adsense has been stealing posts from all over the place. I was notified of this fact by a comment left over by a reader.
But I got a laugh today reading eurotelcoblog. He posted an article referring to stealing posts. And the post appeared on the site unedited! Ha. May be Adsense money is too attractive to this person.
In case post get deleted I have captured the post. If you notice, he times your post earlier than the copied post, making it appear that you copied the article.

Update 2:
I have taken down the links to voip_niche, you need use cut and paste URL's if you was to see his articles.

Well this post appears on before eurotelcblog's post. This is what I meant by timing posts to be earlier than the original post.
This post appears at this link voipniche times your posts differently mine are about 5 hours early. He even tracks back your links.
Hello Rob, at FreePBX, remove VoIP_niche track back link, the original article is here on this blog!

Eurotelcoblog post on voipniche Same article on voipniche


Skype gets Wireless Voip and Mobile VoIP help from Hellosoft

San Jose, CA — January 31, 2007 – HelloSoft Inc., the world’s leading provider of VoIP products, announced today that Skype has licensed HelloSoft’s Internet-Telephony technology which will be incorporated into Skype’s communication software. HelloSoft’s technology will allow consumers to enjoy free, high-quality voice calls anywhere their smartphones and Pocket PCs can connect to the Internet.
“HelloSoft products being a low-footprint design are extremely cost-effective and power-efficient,“ said Krishna Yarlagadda, HelloSoft’s President and CEO. “Our technology will support today’s mobile and wireless generation as they use Skype™ on the go. We are excited to be working with Skype to enhance the performance of Skype for Windows Mobile.” “We are always looking for ways to enhance the voice quality for Skype users,” said Eric Lagier,
Skype’s Director of Business Development, Hardware and Mobile. “Working with HelloSoft will not only help provide millions of Skype users around the world the voice quality they expect from Skype, "voip niche com violates other people's posts" but helps deliver the same clarity on their mobile devices that they get on their computers.”
HelloSoft's award winning VoIP technology is the industry’s most optimized and comprehensive software solution including SIP, media subsystems, vocoders, and IMS/VCC. HelloSoft has a range of VoIP products to meet the price and performance needs of multiple high-volume markets, including next generation multi-mode mobile handsets, PDAs, IP phones, and ATAs.
About HelloSoft
HelloSoft’s vision is to empower, enable and expedite the converged communications eco-system globally, by providing highly optimized, power-efficient, cost-effective, mass market, communications technologies and solutions. A pioneer in feature-rich customizable VoIP platform, combined with its best-of-breed Cellular and WLAN Intellectual Property (IP) portfolio, HelloSoft’s offerings enable costeffective,
low-power and high performance mass deployment of multi-mode mobile and portable
communication devices for the converged market place. HelloSoft is the inventor of RISC-only implementation, which has been adopted by 30+ major semiconductor and OEM/ODM companies worldwide. HelloSoft strategic partners include Texas Instruments, Intel, ARM, MIPS, Symbian and LongBoard.

News Source

Thursday, February 01, 2007

FreePBX needs your help.

If you have visited recently only to find that the site is not available, That is because Rob and the group at FreePBX needs a new computer and better hosting. Please follow the link to the post by Rob and you will see why he needs our help. The best that happened to me since I got to know Asterisk a few years ago. I have been a heavy user of FreePBX.
FreePBX is, to quote FreePBX's About page, a Standardized Implementation of Asterisk that gives you a GUI to manage your system. If you’ve looked into Asterisk, you’d know that it doesn’t come with any ‘built in’ programming. You can’t plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about. FreePBX simplifies this by giving you a pre-written set of dialplans that allow you to have a fully functional PBX pretty much straight away. Some of the features that FreePBX supports out of the box are:

* Unlimited number of Voicemail boxes
* ‘Follow Me’ functionality
* Ring Groups with calls confirmation (so if, eg, a cellphone is out of range and diverts to voicemail, all the other phones keep ringing)
* Unlimitied number of Conferences (limited by available CPU power - about 300 simultaneous users in conferences on a P4 3ghz - 600 with a dual core!)
* Paging and Intercom functionality for SIP and IAX phones that support it (Eg, Snom, Aastra, Grandstream)
* Music on Hold (via MP3s, or streamed off the internet)
* Call Queues
* And many other features
So now you know what FreePBX is go find out more..

Keep FreePBX alive and healthy by supporting it

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