Sunday, December 31, 2006

Net Neutral AT&T (little Ma Bell) get go ahead to gulp up Bellsouth

The 1974 breakup of AT&T is becoming a joke now for those who remember. I had to dig up Wikipedia to get the information after reading a post by Russel Show. I was too small at the time to pay attention to AT&T. But if you check carefully, it looks like they are rebuilding "Ma Bell". What is next? World Domination?. I like what Russel had to say!
"My thinking? Maybe AT&T is ready to make nice, but here is a company with more than 100 years of fine print sophistry. They need to be watched like a hawk, and I'm not sure this FCC is quite up to the task."
Ha not much different from the ideas of people who know AT&T, that I spoke to!. Read more at Russel's Blog. You get to vote on how you feel about the deal and has links for the information on the deal.

Russel Show's article on "Ma Bell"

Alec Saunders makes Ultimate VoIP Blogger List.

He is definitely wrong! I am not there ;), even though I have 1.8 Blog Juice and there are about 15 Bloggers are below that level, he must be using some other info as well. May be because I have only two Blackberry related posts. May be it is my English!!
But the list is impressive. I can see most of the blogs I read on the top and some that I did not know but got to know because of the list. Thanks Alex.
May be next year, I will get somewhere on the list! Until then you can get to know the OTHER top Voip Bloggers. Follow the link(s);

Alex Saunders VoIP Blogger ranking

Blog Juice calculator

My Blog Juice

Trolltech releases GreenPhone, The "open" Linux phone platform

The has a report about Greenphone, The "open" Linux phone platform, released by Trolltech. Very good Idea with bad accountants around. It will be hard to attract Open Source developers with $700.00 device. But I applaud the effort and hope when accountants are not looking techies will reduce the price!. Then I will get one. Untill then I will stick with OpenMoko.

Links; Greenphone article
Openmoko Development

Saturday, December 30, 2006

OpenPBX RC3 is ready for you

The developers and project participants have released the 1.2 release candidate 3, The last developer release before the final OpenPBX 1.2. So if you want to get familiar with this Open Source PBX, which was derived from Asterisk, it is time to get your hands dirty.
This release includes the highly anticipated and robust new conference bridge application called NConference. The NConference application has many exciting features such as conference dial-out and runs without dependency on any hardware timing devices.
This release drops Editline support and introduces GNU Readline as a replacement. app_conference was removed and NConference, a new feature-rich application, replaces it. A new version of generic timers providing better performance is now inthe core to support FreeBSD 6.2, NetBSD, OSX and other OS. CLI Issues on OSX are hopefully fixed now. This release adds initial support to ucLinux. Also included are numerous other small bug fixes through out the entire code.

The new Nconference conference bridge is shaping up to be one of the new crown jewels of The new application has a wealth of features, but most importantly it works accross many different codecs and audio sample sizes.
The new conferencing application will work on all platforms supported by and will not rely on any hardware-based timers. Nconference also contains many new and innovative features such as "consultant". Consultant allows a moderator to have a helper that can only be heard by the moderator. A dial-out feature will allow moderators to call out of the conference to join new participants.

OpenPBX wiki, download and build notes.

Thursday, December 28, 2006

VOIP Service providers ranked

ISP planets Alex Goldman has released a ranked list of VOIP providers based of subscribers for 3rd quarter 2006. From the list, It seems mostly Cable companies are beating the rest of the providers, except Vonage, Skype, and Sunrocket. This is from a list of 12 providers whose data is publicly available. The report also mentions that many providers are left out due to unavailability of data.
Here are the first five from the list; follow the link below to get the rest and read the report.

1 Vonage 2,000 [as of September 30, 2006]
2 Skype 1,800 (paid VoIP only, worldwide, date of this number unknown)
3 Time Warner Digital Phone* 1,649 [November 1, 2006]
4 Comcast Digital Phone* 1,348 [October 26, 2006]
5 CableVision 1,101 [November 8, 2006]

ISP Planet have done this ranking since 2004. The history of ranking is also available.

VOIP Service providers ranked
VOIP ranking History

Tuesday, December 26, 2006

China Voice Holding Corp secures second VOIP Contract with Chinese Government

China Voice Holding Corp secures second VOIP Contract with Chinese Government.

China Voice Holding Corp., a provider of Voice over Internet Protocol (VoIP), Office Automation and Wireless solutions for government, businesses and consumers in the People’s Republic of China and the United States, announced today that it was in the final stages of completing the initial contract phase to provide its integrated Internet Telephony and patented groupware and Office Automation solution to the Chinese Leading Group Office of the Poverty Alleviation and Development Agency. The integrated voice and data solution is provided by one of the CHVC’s Chinese subsidiaries, Candidsoft Technologies Company Ltd. of Beijing, Inc. This agreement combined with the Company’s previously announced agreement with the Navigation Affair Administration Bureau of the GuangXi Autonomous Region calls for the Company to install a minimum of 85,000 seats. Upon installation, the Company projects that the revenue from these two contracts will exceed over thirty million dollars annually.

Mr. Chun Lin Xing, Founder and CEO of Candidsoft Technologies Company Ltd. of Beijing, Inc., said, “Now that we have been successful during our initial testing and evaluation phases we are aggressively moving forward to deploy our unique integrated solution to these government agencies. During the last few months we have been working with and evaluating various telecommunications companies to subcontract specific labor and capital intensive elements of our Chinese Government contracts. We now have all of the necessary elements in place to deploy, manage and support these government contracts.”

CHVC’s President and CEO Bill Burbank said, “We are very pleased to have passed the testing phase and at the same time identified and chosen the strategic partners to support these contracts. This proven model will be utilized to secure the additional government business that we have been offered.”

Information on China’s Poverty Alleviation and Development Agency may be found at

SOURCE: China Voice and Jogjaponsel

China Voice Holding Corp
China Voice

FreePBX 2.2RC3 is ready after fixing the RC2 bug.

After disappearing for a while (Did you have a holiday?), Rob Thomas is back and even looks like skipping X'MAS to give us a bug free version of FrrePBX 2.2RC3.
Rob released the Version 2.2RC2 just in time for X'MAS but had hold it back to fix a major bug that did not save the inbound route destination. So if you have downloaded the FreePBX 2.2RC2, you better head over to FreePBX site or Sourceforge to get the FreePBX 2.2.RC3.

FreePBX release announcement

Jeff Pulver's blog-tag takes many a turns, here is mine

A blog Game started by Jeff Pulver a few weeks ago, running around the blogsphere. Many a people have made their own and I like what Dave Gale at hitting the wire has done. I will play both!. If you you want you can follow the Jeff's version or Dave's version, links are below. SOlOSEO keeps tag on these, tracks the Daves version and invites you to upkeep Jeff's tag tracking.

Five things about me;
1. My first love is Physics, not VoIP
2. I play solo Orchestra, in my studio, even though I skipped my Piano classes
3. In 2001, I bought and sold about 15000 minutes of Voip a week, until a bad man cheated me.
4. I have been in all the continents of which about 20 countries more than I spent more than 3 months per visit.
5. I speak and write in about 7 languages, non of them properly!

I tag these hero's of VoIP and technology;
Jeff Pulver, Andy Abramson, Jo Ito, Ken Camp, David Gale You are it!

Jeffs game of Blog-Tag
Dave's version of Blog-tag
SoloSeo Blog-Tag Track

Jeff Pulver's blig-tag takes many a turns, here is mine

A blog Game started by Jeff Pulver a few weeks ago, running around the blogsphere. Many a people have made their own and I like what Dave Gale at hitting the wire has done. I will play both!. If you you want you can follow the Jeff's version or Dave's version, links are below. SOlOSEO keeps tag on these, tracks the Daves version and invites you to upkeep Jeff's tag tracking.

Five things about me;
1. My first love is Physics, not VoIP
2. I play solo Orchestra, in my studio, even though I skipped my Piano classes
3. In 2001, I bought and sold about 15000 minutes of Voip a week, until a bad man cheated me.
4. I have been in all the continents of which about 20 countries more than I spent more than 3 months per visit.
5. I speak and write in about 7 languages, non of them properly!

I tag these hero's of VoIP and technology;
Jeff Pulver, Andy Abramson, Jo Ito, Ken Camp, David Gale You are it!

Jeffs game of Blog-Tag
Dave's version of Blog-tag
SoloSeo Blog-Tag Track

VON 2007 is coming to San Jose, as it always does!

In the world of IP communications, getting the right information, to the right people, at the right time is the ultimate measure of success – and so it is the ultimate measure of VON!
It is the message from PulverMedia. And it is music to my ears!
The VON 2007 will be held at Jan Jose Convention Center from March 19 to 22. The registration is open now and please follow the links given below.

There are many a things that you will have to cover at the Spring VON 2007, but here are some high lights Don't mis these;

* VoIP
Has the movement toward portal solutions given birth to a new ecosystem of services from the fixed line VoIP service providers?
* Fixed Mobile Convergence
What is the impact on the organizational structure of a service provider when jurisdictions converge as well as services? Who is the billing party?
Can the promise of new services be fulfilled at the edge or by its nature is IMS going to enable new services that belong in the core of the network?
Can IPTV deliver a significantly different experience than existing broadcast, cable and satellite, or is it a me-too offering?
* Wireless
How does broadband move to broadband wireless?

VON 2007 registration
Spring VON 2007 Site

Sunday, December 24, 2006

Best Of the Holiday Cheer to all of you

I hope Santa will come your way and Your holiday will be filled with laughter and Joy. If you pass by an unfortunate person, please try to make him/her smile. Don't forget that all the Kids are the same!

Best Regards!

Tags: , ,

Thursday, December 21, 2006

Asterisk 1.4 gives you frame caching

Asterisk Blog tells up about new feature in Asterisk 1.4 that is called "frame caching". Ii seems that this feature enhances the performance of Asterisk.

"One of the performance improvements for Asterisk 1.4 that I worked on is something that I call "frame caching". The idea here is to optimize the situations where Asterisk has to dynamically allocate and free memory to hold media frames as they pass through the system. This is something that can happen hundreds of times per second for a single call! Doing that many calls to malloc() and free() per second is expensive. So, after implementing a thread local storage API, I then created a thread-local list of unused frame structures. This means that the same allocation can be used over and over.

Here is the results of some of my testing of this code.


Scale: When this value is non-zero, the number of processor ticks will be divided by this number. This is useful when profiling code that takes more significant amounts of time to run.

Events: This is the number of times the counter was started and stopped.

Value: This is the total amount of processor ticks accumulated while this counter has been running.

Average: This is average number of ticks per event.

Name: This is a name used to describe each profile set.

Asterisk article gives you the results of two examples. Follow the link to read about examples and more.

Asterisk blog frame caching

VOIP and IM in hacker cross hairs in 2007

Just a few posts ago, I wrote about VOIP IP Telephony: Skype worm on the loose and now I see an article at Vnunet about their security predictions for year 2007.
Among web based spam and worms and malware attacking users together with image based spam, Vnunet also predicts that "Instant messaging and VoIP will become mainstream targets in 2007. Consumers and businesses will become victims of 'ransomware' which locks up data until they pay to have it released."
May be we will have to hire Mel Gibson, remember Ransom. I followed Andy's footsteps to find this article. So Skypebay, what are your plans?

Vnunet's IT Security predictions for 2007
Andy's footstep Voip and IM

Dark Side of VOIP, Bankruptcies come to play

GigaOM directed me to a press release by One IP Voice announcing that it has filed for Chapter 11 of the U.S. Bankruptcy Code on Wednesday, December 13, 2006 in Hartford and that, on December 19, 2006, a preliminary order to use cash collateral through December 29, 2006 was entered by the Court. One IP Voice has retained the firm Development Specialists, Inc. of Chicago, IL to assist it in pursuing strategic alternatives.

So this is the scene of a crowded market place. I hope One IP Voice can reorganize as Om Malik say because SMB market and Small voice providers all over the world rely on providers like One IP Voice. Bigger players has enough where withal to absorb losses and continue, like Vonage and Skype do.
Om Malik also guided to me to another eye opening article "85 VOIP service Providers shutdown in 15 months". Some of the companies mentioned were known to me and some were not. But that is 85 companies going out of business and 23 of them were from USA.
I think this is also a good forward looking information because as the market shakes down weaker players and the stronger which is left behind will provide us with better competitive VOIP Service.

GigaOM (Om Malik's) article
One IP Voice press release, PR Newswire article
85 VOIP providers bite dust

Tuesday, December 19, 2006 releases OpenPBX RC2 has announced OpenPBX RC2 is now generally available as a tarball and can be downloaded. The download link in provided below in the links.
This release includes the new multi platform compatibility that includes the ability to run on several BSDs as well as MacOS X. Visit OpenPBX wiki for more information.

OpenPBX is a fork from the Asterisk software PBX. The software PBX builds on the solid foundation created by the developers of Asterisk.
The OpenPBX community aims to develop a robust offshoot from Asterisk building on its strengths, flexibility and user community. Some of the planned features include modular architecture, native support for Sangoma TDM cards, integrated faxing and eventually integrated messaging.

OpenPBX will be community driven and released under the GPL.

OpenPBX is a free software PBX now running on several platforms.

Download OpenPBX RC2
OpenPBX wiki

One last beta before the final Asterisk 1.4 release

Asterisk has released information about a new release of Zaptel 1.2.12 and Asterisk 1.2.14 together with Zaptel 1.4.0-beta3 and Asterisk 1.4.0-beta4.
According to the Asterisk Development Team, This will very likely be the last beta release of Asterisk 1.4 before the final release, which is targeted for next Friday.

Also please check Asterisknow for Asterisknow versions of the above. Links at the end.

Here is the complete news release;

Zaptel 1.2.12 Released

The Asterisk Development Team is pleased to announce the release of Zaptel 1.2.12.

This release contains a number of updates:

* compatibility with Linux kernel 2.6.19
* bug fixes to the Xorcom Astribank driver (XPP)
* various other bug fixes

Thanks for supporting Asterisk and Zaptel!

Asterisk 1.2.14 Released

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.14.

This release contains a number of updates:

* a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it
* various SIP interoperability improvements
* memory and dialog leaks in the SIP channel driver
* a fix to music-on-hold random mode that was not really random
* an improvement to app_voicemail to ensure that the message duration is properly included in email notifications when voicemail messages are forwarded
* corrected a segfault issue during reload of the PostgreSQL CDR driver
* a change to no longer include a header file that does not exist on Linux kernel 2.6.18 (and caused a problem on Fedora Core 6)
* many other bug fixes

Thanks for supporting Asterisk and Zaptel!

Zaptel 1.4.0-beta3 Released

The Asterisk Development Team is pleased to announce the release of Zaptel 1.4.0-beta3.

This release contains a number of updates:

* compatibility with Linux kernel 2.6.19
* bug fixes to the Xorcom Astribank driver (XPP)
* support for Digium's TE110P Rev C, VPMOCT064 and new revisions of the S110M and S400M FXS modules
* various other bug fixes

Thanks for supporting Asterisk and Zaptel!

Asterisk 1.4.0-beta4 Released

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.0-beta4.

This release contains a number of updates:

* a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it
* various SIP interoperability improvements
* memory and dialog leaks in the SIP channel driver
* a fix to music-on-hold random mode that was not really random
* an improvement to app_voicemail to ensure that the message duration is properly included in email notifications when voicemail messages are forwarded
* corrected a segfault issue during reload of the PostgreSQL CDR driver
* a change to no longer include a header file that does not exist on Linux kernel 2.6.18 (and caused a problem on Fedora Core 6)
* logging of dynamic queue member addition and removal in queue_log
* a minor redesign of many CLI commands to be more similar to previous Asterisk releases
* significant improvements to IMAP storage support for voicemail
* a change to the SIP channel to avoid offering formats (codecs) that cannot be transcoded due to lack of available transcoders (along with dynamic activation/deactivation of transcoders)
* support for G.722 16KHz (wideband) audio passthrough, recording and playback
* support for standard prompts in G.722 format
* many other bug fixes

Some of the changes in this release are behavior modifications from the last release; please review the UPGRADE.txt file.

This will very likely be the last beta release of Asterisk 1.4 before the final release, which is targeted for next Friday.

Thanks for supporting Asterisk and Zaptel!

Asterisk IP PBX And Zaptel


Skype worm on the loose

Symantec has reported that Skype worm is out in the wild. W32.Chatosky is a worm that spreads through Skype chat messages. It is also known that the “32.Chatosky” worm propagates itself through Skype’s file transfer feature rather than a chat message, the chat message asks user to download a file called “sp.exe” which, when executed, installs a password-stealing trojan and continues to spread through your buddy list. The infection began in the Asia-Pacific region, particularly Korea.

From Symantec Security Response site;
When W32.Chatosky or sp.exe is executed, it performs the following actions:

1. Searches the registry for the location of the Skype application.

2. Displays the following message and then exits if it cannot find the registry:

I could not find Skype !

3. Executes the Skype application and displays the following message if it finds the registry:

Allow this program in skype!

4. Queries Skype for random users every 3 minutes.

5. Starts the Skype application and sends the following message to the users:

Check this! [http://][REMOVED]

Note: At the time of writing, this URL was unavailable but it reportedly contains the worm body.

Symantec Security Response encourages all users and administrators to adhere to the following basic security "best practices":

* Turn off and remove unneeded services. By default, many operating systems install auxiliary services that are not critical, such as an FTP server, telnet, and a Web server. These services are avenues of attack. If they are removed, blended threats have less avenues of attack and you have fewer services to maintain through patch updates.
* If a blended threat exploits one or more network services, disable, or block access to, those services until a patch is applied.
* Always keep your patch levels up-to-date, especially on computers that host public services and are accessible through the firewall, such as HTTP, FTP, mail, and DNS services (for example, all Windows-based computers should have the current Service Pack installed.). Additionally, please apply any security updates that are mentioned in this writeup, in trusted Security Bulletins, or on vendor Web sites.
* Enforce a password policy. Complex passwords make it difficult to crack password files on compromised computers. This helps to prevent or limit damage when a computer is compromised.
* Configure your email server to block or remove email that contains file attachments that are commonly used to spread viruses, such as .vbs, .bat, .exe, .pif and .scr files.
* Isolate infected computers quickly to prevent further compromising your organization. Perform a forensic analysis and restore the computers using trusted media.
* Train employees not to open attachments unless they are expecting them. Also, do not execute software that is downloaded from the Internet unless it has been scanned for viruses. Simply visiting a compromised Web site can cause infection if certain browser vulnerabilities are not patched.

Symantec Security site

Upcoming force in VOIP space, Oracle Corp.

I wrote a few posts ago about Microsoft entering the VOIP space. But another corporation, Oracle corporation is gearing up. It totally remained invisible to me until I read a post by Eric Hernaez over at The Third Screen. He writes about Oracle's 26 acquisitions since January 2005, of which eight have been of companies directly involved in the next-gen telecom space. I will not list them here as I am going to direct you to Eric's post.
I am closely involved with Oracle in my day to day work. And after reading and digesting Eric's article, I can see where Oracle might be going with this. New products that are released from Oracle are more enterprise centric than database centric. Yes all are yet connected to Oracle's flagship, Oracle Database but fringe products are reaching into products that were playground of other companies.
Oracle is becoming more and more of a service delivery company and I do not see a problem with VOIP being a one of those services.
I specially liked something Eric, wrote;
"Eventually, it will be possible to replace the entire switched telecom infrastructure with software and IP devices operating over a broadband packet network. When that happens, it’s hard to see how the traditional network equipment providers (such as Lucent/Alcatel, Nortel and Seimens), all accustomed to generating their bulk revenue by selling proprietary big iron solutions, will be able to keep pace."

The Third Screen's Eric
Oracle acquisitions

Open source SIP stack released

Open source SIP stack OpenSIPStack has been rereleased under a triple licensing scheme to ensure that it can be used by the largest possible number of individuals and development communities. This tri-license aims to address the perceived incompatibilities between Mozilla Public License (MPL), GNU General public license (GPL) and GNU Lesser Public License (LGPL). The stack was previously distributed under MPL 1.0.

Open Source SIP project, openSBC is based on the OpenSIPStack, a fully compliant (RFC 3261) SIP stack designed for stability and scalability, and with a heritage of commercial usage. The project currently contains reference implementations of a session border controller (OpenSBC), Yeya and several components that are useful to developers wishing to use Solegy’s service deployment platform.

OpenSBC is a reference implementation of a hybrid SIP proxy and B2BUA (back to back user agent) created from the Open SIP Stack core. It is well suited for a number of VoIP implementations. Among other things, it can be used as a Registrar for SIP endpoints, as an entry/egress point for SIP trunking applications, or as a far-end NAT traversal solution.

OpenSBC has been designed for scalability and flexibility. Deployments can grow incrementally with traffic needs because a primary instance can be configured to load balnce sessions across other instances. Each instance may be run on separate servers, or multiple instances may be run on a single server.

OpenSBC can perform the following functions:

Session Border Controller: Full back-to-back user agent (B2BUA) hides network topology with:
- Integrated web UI for basic configuration tasks
- far-end NAT traversal with RTP proxy
- Complete transparency for end-nodes with support for pass-thru of non-standard SDPs,
- Routing using static rules, ENUM or Solegy RTBE
- Comprehensive logs using syslog server
- Encryption of SIP and RTP packets with simple hash

Registrar: Fully standards-compliant with support for pass-through registrations (also referred to as upper registration) and integrated support for presence using SIP/SIMPLE or XMPP.

Proxy: Fully standard-compliant with multi-protocol support (UDP, TCP, TLS*), processing and relaying signals from remote (SIP) and local endpoints.

Presence: Compliant with SIP/SIMPLE and XMPP standards with support for PUBLISH as well as SUBSCRIBE/NOTIFY events.

Event Packages: Support for message-summary information about waiting messages (voicemail) and presence

Solegy™ Offers Free VoIP Softphone for Microsoft Windows -- Customized Softphones Available to Service Providers with a Full Range of Calling Features and Back-Office Functionality in a Hosted Environment.

Open Source Sip website
OpenSIPStack web site
Solegy website

Monday, December 18, 2006

iPhone released, seven of them, by Linksys.

Remember the post I made on WIP300, VOIP IP Telephony: WIP300 Wireless-G IP Phone by Linksys Reviewed
Yes that plus a few other models have emerged as iPhone. But not as we expected, as you know WIP300 was by Linksys, a subsidiary of Cisco. Yes iPhone comes to you from Linksys/Cisco not from Apple. May be Apple was too late in registering the trade mark!
But Coolest-Gadgets has more information on the iPhone trademark! He also has an article on iPhone from a different angle. Follow the link below to catch the article. The article on CG states "For almost a year now people have been speculating about the possibility of Apple releasing a communication product called the iPhone. But apparently Linksys, a unit of networking equipment maker Cisco, already owns the “iPhone” trademark. According to, a company called Infogear registered the “iPhone” name in 1996. Cisco then absorbed the “iPhone” trademark when it acquired Infogear in the year 2000. Oops. Score -1 for the bloggers. We really should have picked up on that little tidbit months ago."
Indeed need to do better reserch
The phones themselves looks very similar to WIP300 and only two phones are new in the release. The others were in the market for a while. This is since 2004 and all the products has been wrapped under iPhone product line.

• iPhone Cordless Internet Telephony Kit - CIT200
• iPhone Dual-Mode Internet Telephony Kit - CIT300
• iPhone Dual-Mode Cordless Phone for Yahoo! Messenger with Voice - CIT310
• New! - iPhone Dual-Mode Internet Telephony Kit for Skype - CIT400
• iPhone Wireless-G IP Phone - WIP300
• iPhone Wireless-G IP Phone with Web Browser - WIP330
• New! - iPhone Wireless-G Phone for Skype - WIP320

iPhone Family Pricing and Global Availability:

• iPhone Cordless Internet Telephony Kit - CIT200 - Available globally through e-commerce retailers, retail stores, and VAR partners. Estimated Street Price: $79.99.

• iPhone Dual-Mode Internet Telephony Kit - CIT300 - Available globally through e-commerce retailers, and VAR partners. Estimated Street Price: $99.99

• iPhone Dual-Mode Cordless Phone for Yahoo! Messenger with Voice - CIT310 - Available in the U.S. through e-commerce retailers, and VAR partners. Estimated Street Price: $99.99

• New!! iPhone Dual-Mode Internet Telephony Kit for Skype - CIT400 - Available immediately in the U.S. through e-commerce retailers, and VAR partners. Global availability through e-commerce retailers, and VAR partners is expected for Q1 2007. Estimated Street Price: $179.99

• iPhone Wireless-G IP Phone - WIP300 - Immediately available in North America, Europe and Asia through distribution and authorized VAR partners. Estimated Street Price: $219.99

• New!! iPhone Wireless-G Phone for Skype - WIP320 - Immediately available in North America through distribution, online retailers and VAR partners. Europe, Asia and LATAM availability is planned for Q1 2007. Estimated Street Price: $199.99.

• iPhone Wireless-G IP Phone - WIP330 - Immediately available in North America, Europe and Asia through distribution and authorized VAR partners. Estimated Street Price: $369.99

Linksys iPhone
Coolest Gadgets article on iPhone

Friday, December 15, 2006

Philips VP6500, VP6000 Video Phone is ready for your VVOIP (WVOIP) Pleasure

Although first thing that came to my mind was "How is my ear going to see your ear?" quick research showed that it has speaker and also ear phone that is of similar length to iPod ear phone. So you can set the phone in front of you or hold it in front of you while using the said ear phone or speaker phone.
Just imagine people having video conferences while walking around!
If you get a VP6500, you can connect it to your TV, portable TV, via provided video cable or if possible upcoming iPOD TV for recording the conversation to be uploaded to Youtube later.
You will also need to be near WI-FI hotpot or WI-FI router connected to internet and of course an account with a VOIP provider. When Google Wi-Fi comes to San Francisco, peer 39 will be viewed from Tokyo, Japan, Live.

But if you are planing to have a long vvoip call, better to charge your batteries, the phone is powered by AA rechargeable batteries.

Here are some technical data from the documents provided by FCC;
• High quality 2,2" TFT display
• 65k colors
• QCIF+ (176 x 220 pixels)
• Progressive LCD backlight
• VGA resolution (640 x 480 pixels)
• Rotating mechanism (over 240 degrees) with automatic image adaptation
• CMOS sensor
• Digital zoom: 1,6x
• Brightness control
TV mode (Only VP6500)
• Connection to TV with cable (audio & video output)
• Remote video displayed: CIF, QCIF, SubQCIF
• Local video displayed: QCIF
General telephone features
• Voice-only calls mode
• Caller name & number identification
• Ring profiles
• 15 polyphonic ringer melodies
• Emergency call feature (when keypad locked)
• Customization of user interface (theme, colors, wallpaper)
• Earphones jack
Contacts list and Call history list
• 500 contacts with 4 numbers & 1 picture each
• Fotocall (show picture of caller)
• Call history list with 50 numbers max. (dialed, received, missed)
• WiFi certified IEEE802.11b/g
(up to 54 Mbps)
• IEEE802.11e
• IEEE802.11i
• Integrated antenna
• WiFi Signal strength indicator
• Security WEP-WAP-WAP2
• 2 x NiMh standard AA batteries
Weight and dimensions
• 168 grams
• 134 x 24 x 49 (H x D x W)
Power consumption
• The power consumption of the product during
a video call is 3.5 Watt.
• The lowest power consumption of the product
is reached when the battery is fully charged and
the handset is on the charger. The power
consumption in this case is 360 mW.
Temperature range
• Operation: 0°C to +45°C
• Storage: -25°C to +70°C
Relative humidity
• Operation: Up to 95% at 40°C
• Storage: Up to 95% at 40°C

Philips FCC filling at FCC.GOV

“VoIP Phone Business in a Box” gets new tools

Talkfree International VoIP reseller, has added two new tools to its centerpiece product, the “VoIP Phone Business in a Box.” The first tool, the “Inbound Telephone Numbers” program, arms TalkFree business customers and service resellers with a new service using Direct Inward Dial numbers (DID) from nearly 40 countries.
According to the press release by TalkFree, here are the benefits of the “Inbound Telephone Numbers” program include:

* Customers are given a “host” or “local” in-countryphone number
* A customer's clients and business associates in that country dial the “host number” at local or in-country rates.
* The call is then forwarded anywhere in the world using VoIP technology and bypasses international long distance rates.

The second TalkFree tool being added is a completely new and improved “Call Shop Management” software solution. This tool provides call shops with the flexibility to establish rates and margins separate from the wholesale VoIP rate charge. In addition the program delivers to the call shop real-time visual tracking of every call in progress and, at the same time, allows customers to view displays monitoring their actual communication time. An additional but significant feature of the program is the option to bill for calls less than 40 seconds in duration. “Call Shop Management” also offers one other ultimate convenience for the call shop and for the caller: the receipt. Each receipt is printed immediately and displays only the call shop's independently set rate in the local currency.
Press release also mentions that VoIP TalkFree is focused entirely on the Emerging World and promotes social entrepreneurship through the creation of small telecommunication businesses. The company is one of the top international VoIP reseller enablers, and is composed of a diverse team of people with a profound knowledge of the countries they serve.

Talkfree VOIP in a BOX

Vonage E911 support is at 93% now

Vnunet's Clement James, reports that Vonage America, a subsidiary of Vonage Holdings, has achieved providing E911 service to 93% of its customers.
The E911 functionality incorporates a feature that automatically associates a physical address with the calling party's telephone number.
In the past it has been a problem to provide E911 services as existing telephony providers, that has 911 capabilities, were not so keen on extending that service to upcoming VOIP IP Telephony service providers like Vonage. Some providers have paid incumbent telephony carriers to get access to the services, while services like Skype, keeps on dodging the bullet.
When a supported VOIP customer calls 911, the customer's call is automatically routed to the appropriate 911 center, with the caller's registered street address and telephone number appearing on the dispatcher screen regardless of where or what exchange they are calling from.

Over the past two months, Vonage has equipped over 170 locally run emergency call centres across the US with E911, bringing the total number of calling centres with emergency 911 service to over 6,400.

Vnunet provided the initial news

Thursday, December 14, 2006

Sprint Gives your Cell phone a PBX, with Sprint Wireless Integration

BUSINESS WIRE News article reports that Sprint has today announced the launch of Sprint Wireless Integration, a product that extends customers’ premises-based PBX features and functionality to their mobile phones. The solution offers business customers additional value and new capabilities by integrating Avaya “Extension to Cellular” capabilities and new Sprint network advancements.

Sprint Wireless Integration features include providing users with one phone number that simultaneously rings both the desk phone and mobile phone, along with one converged enterprise voicemail inbox. It also extends PBX features like conferencing and call forwarding to the mobile phone so users can get all the functionality of their desk phone even while away from the office. For example, mobile users can make intra-company calls by simply dialing the four-digit extension of the person they want to reach, just as they would from the office desk phone – with no access numbers to dial or codes to enter first.

Built within Sprint's IP Multimedia Subsystem (IMS) architecture, Sprint Wireless Integration is the industry's first "hosted mobility" solution. “By converging wireline and wireless functionality, Sprint Wireless Integration provides a better overall service – one that is more functional and also makes communication more simple and effective,” said Tony Krueck, vice president of product management and development, Sprint. “This solution is a great example of the promise of Fixed/Mobile Convergence.”

Sprint Wireless Integration provides:


* One phone number with simultaneous ring to both the desk phone and mobile phone (using the existing desk phone number)
* One voicemail inbox using the enterprise voicemail platform
* Abbreviated (e.g., four-digit) intra-company dialing from the mobile phone
* Class-of-service extended to mobile calls for better control
* Mobile call tracking/logging by the telecom manager using the PBX


* Outbound mobile calls routed through the enterprise PBX are “on-net” and included in the monthly service fee. (Inbound calls to the mobile phone do incur minutes.)
* Mobile-to-international calls are billed as if from the enterprise PBX or VPN
* Desk phones can be eliminated if desired
* Billed as an add-on feature ($20/month) to an existing Sprint CDMA Wireless Plan


* Premise-based Avaya Communications Manager (IP or TDM)
* Sprint CDMA mobile phone with data capability
* Sprint Dedicated IP or Global MPLS VPN connection

More details on Sprint Wireless Integration are available at

Businesswire your daily news source
Sprint wireless integration

Get your VOIP Gear at 2007 International CES® in Vegas January 8-11

Geemodo has a lengthy report about 2007 CES®, Consumer Electronics show in Las Vegas. Looks like this time the show is going to be the largest in it's 40 year history. According to the article, there will be a lot of exhibitors, representing more than 30 consumer technology categories, including electronic accessories, broadband, digital imaging, electronic gaming, high-performance audio, home theater, mobile electronics, robotics, VoIP and wireless communications.
Read more and find relevant links at Geemodo.

Geemodo: 2007 International CES® in Vegas January 8-11

Wednesday, December 13, 2006

Paris to get the largest fiber network in Europe

According to a Business wire News announcement by Cisco, Free, a division of Iliad Group and the leading triple-play over broadband operator in Europe, is to roll out the first and largest optical fiber network in France and Europe, based on the Cisco Internet Protocol Next Generation Network (IP NGN) architecture and using Cisco Ethernet fiber-to-the-home (E-FTTH) technology.

Free announced that residents will have access to broadband speeds that could exceed 50 megabits per second(1) for 29.99 euros per month, and benefit from advanced services such as high-definition IPTV, video on demand, multimedia communication (VOIP, VVOIP) and Web 2.0 services.

Michael Boukobza, chief executive officer of Free, said: "We are building a network of the future for our users and by working with Cisco, we can bring the future closer. With the ongoing debate about fiber access platforms, we have made a clear choice and decided on Ethernet point-to-point FTTH architecture because it is future proof and maximizes return on such an important infrastructure investment. Other options would not have set us so clearly apart from the competition. By taking fiber optic links directly to the home, we can be sure that France will continue to be at the forefront in technology and applications for the next few years, and even the next few decades!"

Business wire News network

Another disruption in North American VOIP space, Skype introduces new rate plans.

USATODAY.COM reports that Skype will offer a call rate plan that allows unlimited calls mobile and land line phones in USA/Canada for $29.95 per year. That is same as most other VOIP monthly plans! Talk about Disruption. May be Skype/Ebay is thinking we have already spent billions on this ship, lets go as fast as possible before hitting iceberg, may be the iceberg will break apart! I think they might do that.
The calls have to stay within, originate and terminate, within USA and Canada. Skype also went Gold with Skype 3.0 at the same time of this announcement.

But Andy Abramson at VOIP Watch tell us how to do that from other countries, using hotspotvpn, which will cost you about $9.00 a month. So do your Math, before going that route. He also points out that if you subscribe now, before January 1, 2007, you can get Skype deal for $14.95 and some goodies, they'll also get about 100 minutes of free international calls and $50 in coupons for Skype gear, such as a Motorola headset.
If you have Skype credit, you are also able to convert them.
But the Skype service in North America has been free for a while, since May of this year. So it is unlikely that you have any credit because if you did not spend it, Skype seems to have eaten it, according to Scott Wilson at Indigo Moon. Here is his own words;
"But suddenly, without notice or option, a few months ago Skype arbitrarily revoked the credit, saying it had been on their books too long. So, they removed my ability to spend the money I had given them, then kept the money without credit, and now want to charge me again. How stupid would I have to be to pay up front again for a service from a company that would do that?". Read his post by following the link provided.

Andy's article tells you much more detail about the deal and impact on Skype, Earthlink and other VOIP providers.

Although you could have a phone number for everyone in the family, and call anywhere for almost free, remember, With Skype there is no E911 support. So you need to switch to your local voice provider to call 911, and how many people will remember that in an emergency? I think FCC or an Public Utility commission (PUC), will jump in once a firetruck or Police sent to a wrong place or Nowhere.
So if you are all about saving money and making Skype richer, time to subscribe.

USATODAY article
Andy's "Skype 3.0 Goes Live, New Cheaper Disruptive Calling Plans Announced for USA"
Indigo Moon worries about his money

M$ drops Live and adds VOIP to Communication server 2007

Just like with Netscape, where M$ got lucky, the software giant is again into the catch up or me too game. This time with VOIP. But this time it seems to have a plan.
Leveraging it's Office technology and the platform, and dropping Live from the previous release, M$ has entered private beta testing of Office Communications Server 2007, the software integrates voice over IP (VOIP) calling with traditional phone setups and PBXs.
M$ has also ensured a smooth ride by partnering with Nortel, Alcatel-Lucent, Avaya, Cisco, LG-Nortel, Mitel, NEC Philips Unified Solutions, Polycom, and Siemens to ensure Office Communicator 2007 supports existing desktop phones and TDM or IP-based PBXs. Customers can also use phone software on their PC to make calls without purchasing extra equipment, directly from Office Applications.

Gurdeep Singh Pall, Corporate Vice President of the Unified Communications Group at Microsoft said, “The convergence of telecom and data networks is happening rapidly. Software will integrate these two worlds, enabling IT managers to deliver new communications possibilities that include VoIP. With this open architecture and broad interoperability, Office Communications Server 2007 will give IT managers the flexibility to determine when and how and in what way they move their communications infrastructure forward.”
"Some of the capabilities available in the private beta of Office Communications Server 2007 are placing and receiving voice calls; advanced call routing; streamlined integration with the new unified messaging capabilities in Exchange Server 2007; multi party conferencing; call holding, forwarding and transferring; and compliance capabilities, all while working in concert with existing telephony infrastructure,"

VOIP Payphones, will they catch up too!

We have seen enough models and accompanying advertisements for VOIP Phones, Hybrid Phones, dual mode phones, Video phones, wi-fi phones etc.
So I set out to search VOIP Pay phones. Not many products out there yet. I did find a few far eastern companies producing them.
But I did find an Australian company that has product in deployment. pieNETWORKS has created VOIP PayPhone as Hotspot Webphone, which allows one to:

* Provide cheap phone calls through VOIP
* Provide broadband internet access
* Provide Hotspot wireless access
* Provide Digital Media to advertisers
* Remotely manage your network of devices
* Extract detailed statistics and reports of usage
So they must be finding a market in travelers and unconnected users. Although cell phones have reduced the use of pay phones everywhere, one can still find someone putting coins into one of those. So it might be a niche that might work. But very rarely I see people at airport kiosks that provides similar services here in USA. I have never used those.
May be they out to place one of these phones next to those kiosks.

Tongya VOIP payphone

Sunday, December 10, 2006

IPPBX market grows and may surpass traditional PBX and Key market

According to a report from Dell'Oro Group, the trusted source for market information about the networking and telecommunications industries, indicates that IP PBX market revenues grew 10 percent in the third quarter and will be the fastest growing market segment during 2007 when IP PBX revenues are projected to surpass those of Traditional PBX and Key switch systems.
“Implementing IP Telephony paves the way for enterprises to capture the benefits of advanced voice applications including emerging Unified Communication solutions,” commented Steve Raab, Director of IP Telephony Research at Dell’Oro Group. “In the third quarter, IP line shipments increased for all but one vendor, and we project strong growth in IP line adoption in 2007,” Raab added.
The report also reveals that enterprise PBX market revenues climbed 9 percent over the same quarter a year ago partly on strong sales in Europe. During the recent quarter, Avaya and Cisco both exceeded 17 percent line shipment growth in Europe as they continued to expand their presence in the region.

Dell'Oro group

Asterisk trademark and Give me all you got!

I think it is as important as explaining GPL v.2.0 or GPL 3.0, to explain to people that open source is a also a business model. In a business people expect to make money with the products/services they provide.
But when it came to Asteisk, which I have known since it's humble beginning. I met Mark Spenser, the creator of this project, once at a Linux world expo, a long time ago. I am sorry to say that not many people were around the booth but Mark was pretty enthusiastic and carried me through all the details. He had it all together with all the hardware and software running, allowing to make demo calls through a real life PBX.
One interesting thing about trademark was, There was a fork of Asterisk at that time, Asterisk-NG if I recall right. I asked about it from Mark, and his answer was a shake of his shoulders and said "yeah, I heard about it". I think it was appropriate response at that time, as he was paying more attention to the project/product than to business and copy rights or trademarks.
But now Asterisk is vastly different. It is a business and as with any business, you start hiring lawyers, and lawyers will always find things to do. And somethings are not pleasant but you got to do it.
Last week I read a few articles about asterisk@home. and one esteemed writer that I read regularly, referring to the matter as "dark side of Asterisk". Smith on VOIP got most of it right. But if he had referred to recent DIGG and DIGGAME saga, he would have understood the legal process.
I am not a lawyer (IANAL) and what I understood is that if you do not try to protect even attempt to protect your copyrights or trademarks, you might loose them, in a court, in the future. You have to show evidence that you did take steps to protect your trademark.
So Asterisk@home, a super project, that I use day to day, happened to be the target. It has a name too close to Asterisk, if asked from any one not familiar of the projects they might think Asterisk@home is a subproject of Asterisk.
I think it is right for Asterisk / Digium to go after Asterisk@home although not pleasant, the new name, Trixbox seem to be doing fine and the same following that Smith mentioned Asterisk@home, welcomed and enveloped the TrixBox.

I really do not see Smiths arguments but all I can say is open source projects are run by living people, who like money like Smith do. If his idea is only to write and inform people of VOIP, why Google ads and sponsored ads on his site? Because they make money and if Smith can earn some money while enjoying his writing all the better.
Asterisk / Digium did nothing much different, monetize their ideas. If Smith sees one of his articles copied word by word by a spam blog, I can only guess how he would feel. May be asterisk thought the same way.
Has Asterisk / Digium gone after OpenPBX? although it is based on Asterisk and does some aspects better than Asterisk, I don't see Asterisk going after them legally or otherwise.
I prefer the writing on the subject by Alec Saunders, he sees the same issue as Smith does, but Alec's article is more pleasant and to the point. Tom Keating also speaks of the same matter, and much closer to Smiths thinking.
Also remember that trixbox is under the arms of Fonality and Fonality is a direct competitor to Asterisk. Many analysts and writers have said that fonality might do better than Digium in the SMB IP PBX space, using Asterisk itself!

So you want the best answer! Ask yourself what would you do if you were Asterisk /Digium ? that is the best answer, whether it is right or wrong.
By the way people who feel sorry for Trixbox, write Trixbox (formally Asterisk@home), I do where ever it deems necessary it might help to let people know.

Another thing is selecting well known well used words as trademarks is not a good idea. Why do you think Microsoft settled with Lindows? food for your thoughts.

SmithonVOIP the darker side of digium

Alec Saunders Protecting tha Asterisk Brand
Tom Keating Who is redirecting trixbox web traffic

Saturday, December 09, 2006

Activa gives TAPI interface for Asterisk, dials out of outlook

Another Open Source application, Activa for Asterisk provides additional resources that adds additional services to Asterisk, Like call center implementations adding value in areas such as computer aided telephony, screen pop, click2dial, agent control, automatic dialing, outlook integration. TAPI enabled application interface to Asterisk is also provided via the ActivaTSP framework.

Activa package includes a basic framework that enables integration of Asterisk with your C++ applications. In addition it also includes an Asterisk TSP that is named Asterisk ActivaTSP.
ActivaTSP enables integration of TAPI based third party applications and Asterisk. The TSP itself has been developed using the basic ASTProvider framework.
As an example, one could initiate outbound calls through Asterisk using your Microsoft Outlook and other applications. Call screen-popup for incoming calls is also supported.
In addition to the above, advanced TAPI features such as ACD agent operation control and hands-free operation are being added.

The latest version, Activa TSP v1.2.2 Supports Fast Transfer (transfer without consulting) executing # + extension number.
All the functions mentioned will also work with TrixBox, what was once known as asterisk@home.

ActivaTSP home
ActivaTSP at Sourceforge

Vonage leads US VOIP Market 3rd Q 2006

According to another report by Telegeography, US VOIP subscriptions have gained 18% to 8.2 Million users in the third quarter of 2006. But it also reports that the rate of growth has slowed down from the previous years.
VoIP revenues for the second quarter were up about two and a half times - $732 million across the United States, compared with a year-ago level of $298 million.
TeleGeography predicted the market would grow by roughly 1.5 million subscribers in the fourth quarter to end the year with 9.7 million, or about 8.7 percent of the nation's households. Revenues are expected to approach $2.6 billion for the year, or more than 2.5 times the 2005 total of just over $1 billion.

Current Standings on VOIP Market;

Vonage Holdings Corp. 1.95 million subscribers.
Time Warner Inc.'s cable TV 1.64 million.
Comcast Corp. 1.35 million.
Cablevision Systems Corp. 1.10 million.

Certainly good news for VOIP suppliers, be it phones, services or minutes. VOIP is gaining the space, slowly but surely.


Friday, December 08, 2006

Does Skype hurt international Voice traffic?

Will services like Skype, basically computer based enhanced IM clients, eliminate the need for international carriers? According to data presented in the newly released TeleGeography Report and Database, Not anytime soon.

During 2005 we all spent 272 Billion minutes on telephone calls. this figure changed in 2006, it became 313 Billion. This is about 15%.
So how much of this is Skype? in 2005 it had 2.8% of the telecom pie. In 2006 it increased, to 4.4% of the 2006 pie. It is a hefty increase and may be Ebay is in a path to recover what it invested on Skype. Hurray for Skype.
So hoe did the Other VOIP providers do? Not so bad, they are slowly eating into the switched circuit service providers. The Other VOIP service providers increased from 16.6% in 2005 to 19.8% in 2006.
Both these VOIP services, reduced Switched circuit carriers, form 80.6% in 2005 to 75.8% in 2006.
So the telecom giants are still in control it seems. But IP based telephony will be all over us as time goes by.
'Someday, all calls will be routed over the Internet,' commented Stephan Beckert, Research Director at TeleGeography. 'But the numbers suggest that traditional international carriers aren’t going to disappear anytime soon.'

Updated continually since 1989, TeleGeography Report and Database has become the benchmark report for the international carrier industry. To learn more about the enhanced 2007 version, available now, please follow the links provided.

Telegeography Report and Database
Telegeography main site

Web Conferencing with DIMDIM

Since VOIP is all about communication medium, many a solutions like call center, VVOIP, Video conferencing are growing around the core VOIP solutions. A few such applications I already covered such as lightspeed, 1bizcom.

There is another aspect to this, web conferencing. There are many a solutions that aid and provide web conferencing. I like the all aspects of remote communications. Video, telephone and web has bought all of us lot closer than ever before. Other than the social aspects of communications NG (New generation) there is also a business aspects to all these. We do not have to run to Tokyo to have a meeting but combined Voice Video will bring us as close as possible.

The reason I spoke of web conferencing is that I would like to bring to you knowledge a web conferencing product, DIMDIM. Although I support all types of products that bring us a better communication mediums, I have a soft spot for Open Source Products. Some of them may as successful as Digium's Asterisk and some of them might be just emerging from enclaves that they thought bout and built. If not for OpenH323 and OpenGK, I would have never gotten in to VOIP, almost a decade ago.

So what is this DIMDIM thing? Of all things, it does not dim any thing. One thing is sure it will light up some lives and people and of course some business meetings.
So how does DIMDIM do it? Well look at it's feature and versions should give you a picture.

DIMDIM current Features;
* Presentation and Document sharing: Interactive real-time Collaboration over documents and presentations allows enhanced expression and exchange of ideas.
* Audio and video sharing: High-quality multi-party video and audio sharing can be used to personalize meetings with a face-to-face approach.
* Application sharing: Full screen as well as specific application sharing from a Presenter’s computer can be used to show and educate even a novice audience. This is in keeping with the "show, don’t tell" principle.
* White board and Annotations: Realistic interactive collaboration which involves a lot of annotations, corrections, group drawing (and doodles in the margin) is enabled through digital whiteboard and annotations feature of Dimdim. These features are possible on existing documents too leading to seamless distributed brainstorming.
* Chat: As in any real world meeting there will be sub-groups of people engaged in conversation and exchange of ideas (some serious ones others more like “kicks under the table”) within the larger assembly in a web conference. This is facilitated through the multi-user chat feature.
* Polls: Polls enable the presenter to gauge the mood of the participants and to take decisions considering the opinions of many.
* Question manager: Question manager which is like a moderated Q&A setup enables the presenter to better manage the interaction.
* Record and Archive: All the interaction is recorded and archived for sharing with non-attendees and to reach a broader audience.
* E-Learning: Teachers can conduct classes as web conferences so that physical presence of the teacher as well as students is not necessary. Also the archived sessions can be used by students for later reference as well by the teacher as a teaching tool.

There are more to the features and editions. I think in order to get an idea what DIMDIM is really like, follow the links below.

DIMDIM feature Matrix
DIMDIM at Sourceforge

Thursday, December 07, 2006

Call Center Application (WEB 2.0) for Asterisk

Another open source Asterisk based has come to bloom at 1bizcom. The open source software based 1bizCom is next generation web-based, multi-tenant, distributed, mulit-lingual, inbound, outbound Video enabled VoIP & VVoIP call/ contact center solution for Asterisk with Built-in phone, IVR, CRM, Predictive dialer, ACD, Chat, Mail, Fax, Video and other features.
The system runs on Windows IIs web server and the MySQL data base could reside on a Linux server if preferred. The system is in beta and the current version seems to be and the application package is known as 1bC.
According to the 1bizcom site;

1BizCom has all the salient features which goes beyond with new creations like efficient call handling, Automatic Call Routing (ACD), Voice Logging/Recording, Authentication, Conferencing, Data base screen pops, Answering machine detection with detailed statistics and reporting for reduced response time, improved customer experience and quality management of the operations. 1BizCom is a highly scalable Call Center Solution supporting three-dimensional scalability, which will enable client to scale up and optimize the operations as per changing requirements with vendor independence. 1Bizcom helps users to take care of the entire value chain and continuously work on processes & help to generate value at peak places.
Exclusive Features

Automatic Call Distribution (ACD)
Call handling features
Call connection notification on the screen
Call transfer, Call Conferencing, 3rd party verifications
Voice Logging, Monitoring
Call Detail Record (CDR) and Reporting
Web based solution
Maintaining call history

I have not tested the application, due to the fact that builds were pulled off by the developer. The developer left the following message on the Sourceforge Forum;
"Thanks for outstanding support that has been landed to 1bizcom since its lauch few days back...Unfortunately, due to some critical issues, I have to pull the builds...But, please note that we are highly motivated by the initial community support landed and we are firm to provide you a very strong collaborative open source platform.
Please, give us some time and we will get you an improved version of the software very soon. "

But if you are handy with subversion, it is possible to grab the application and the code. See the links bellow. But I will watch out as the developer might have had a good reason to pull out the build.

I have gone through the demo provided on the side and I feel that his Asterisk add on has clear long path ahead.

1bizcom site
Sourceforge project
1bizcom SVN at Sourceforge

SightSpeed set sights on you and video Conferencing

Sightspeed has released an enhanced version of their Sightspeed 5.0. Sightspeed 5.0 already had a good following already. With the new version, they really have done some enhancements.
If you did not know what sightspeed is; Through SightSpeed, users can place free video calls, leave video messages, or post videos to their blogs or personal Web pages. SightSpeed also offers unlimited PC-to-PC voice-only calling (works great for Macs, too), and provides calling to and from traditional and cell phones. And users can try SightSpeedTV, now in beta, for personal viewing of TV on a computer equipped with a TV capture-card—anywhere in the world, complete with real channel-surfing.

Here are the enhancements mentioned in the press release;
Enhanced SightSpeed 5.0:

* SightSpeed Pro plan subscribers (i.e., small-business and power users) will now be able to record three-minute video messages for e-mail or video blogs (vlogs)—a 50-percent increase in recording time, and six times more recording time than available to free users.
* All users who want to share their SightSpeed-produced video messages (in particular, bloggers who wish to vlog) can now easily copy and paste the relevant HTML snippet for the video player directly into their Web site or blog for a significantly more compelling overall user experience.
* SightSpeed “click-to-call” video buttons will now have greater visibility and usability, and can easily be pasted into community or social-networking profiles (such as MySpace, Facebook and others) or other personal or business Web pages. The benefit is that subscribers can give others the ability to easily click-to-call the SightSpeed user through a Web interface, without the need to download any software. This is significant for individuals and business users alike.
* The notification that is automatically sent to video-message recipients has been redesigned with a cleaner, more compelling look and feel.
* Checking out the latest SightSpeed beta service, SightSpeedTV will now be easier because the link is no longer hard to find. Users who want to experience this truly live video place-shifting functionality will now find SightSpeedTV front and center within the SightSpeed interface, thereby giving them a uniquely compelling personal television experience anywhere in the world.
* International users who want to upgrade to the Pro plan or make voice calls to regular and cell phones will now have an easier way to pay for those services and become power SightSpeed users. SightSpeed has now significantly improved its billing and transaction-processing operations to make it easier for users outside the U.S. to subscribe to fee-based SightSpeed services and pay securely by credit card.
* With the holiday season upon us, there’s no better way for far-flung friends and families to stay connected than with a gift of SightSpeed. An enhanced online ordering system now makes it much easier to give SightSpeed starter kits and have them shipped anywhere in the U.S.
* And finally, thanks to an agreement with Commission Junction, online referrals from independent Web site owners will now be eligible for commission when the referrals result in a sale of SightSpeed services.

Links;, in my backyard! literally
Known Sightspeed blogger

Genius Mouse for VOIP and all IMs

Geniousnet has released Genius Navigator 380, which is 1200 dpi optical mouse as well as a tethered VOIP / IM phone.
According to Genius, Genius Navigator 380 can be carried along with your notebook anytime, anywhere. As long as you have internet connection, be it in the office, home or your favorite hangout, and you are writing that paper, over due home work or browsing the Internet, and the LED on the Navigator 380 blinks to tell you that someone just called you through Instant Messenger or your VOIP application like Skype. And if the PC speaker is on, the ringing will also alert you for the incoming phone call. So one just needs to pick up the mouse, and open it up like a clamshell mobile phone to Navigator 380.
Genius also tells that, the Genius VOIP Mouse software can support up to six Instant Messengers (Skype, MSN, Yahoo, GoogleTalk, QQ and AIM) and combine these IMs into one window. Seven buttons in Navigator 380 are for the phone functions --- Phone on/off, List up/down, Volume up/down and OK for call up the IM window.

Genius net

Wednesday, December 06, 2006

Meraki wireless mesh may come to San francisco

Meraki is an extension to the MIT Roofnet project, with the hopes of bringing free or low-cost Internet access to people around the world. I have been testing using and messing around with MIT's Roofnet for about two years, along with many other open source Wi-Fi projects. What draws me to MIT Roofnet is the Mesh feature. Allowing to create a network mesh of wireless devices and extend the reach of wireless network.
According to Katie at GigaOM, It looks like Google has also seen the value of MIT Roofnet, Meraki and it has together with some VCs has funded the Meraki.

Meraki’s co-founder Sanjit Biswas has told GigaOM that the company completed a bridge round of funding last week, which included Google and “a few Silicon Valley angels. We’d bootstrapped the company so far, so this cash is really just for growth/acceleration . . .and for the development of some products we plan to launch next year,”
The company currently sells a $49 wireless 802.11b/g router, which is according to the Meraki that is in beta, that allows users to build a wireless mesh network or extend the range of a community network.
Google showed a Meraki router previously at a San Francisco WiFi community meeting/ townhall meeting, a good, inexpensive way for residents to extend San Francisco’s planned city-wide WiFi network indoors. What goes on outdoors?
Unlike MIT's Roofnet project, Meraki is not completely Open Source. Some part of the software / router is closed source as Meraki tries to commercialize the Roofnet project. Also Management of the mesh networks are handled by Meraki and I did not see the source for that either. But visit to the sites open source code, with some knowledge of Roofnet project, openWRT, it is possible yet to run your own open source wireless mesh, if you do not want to pay.
But I think it is worth to get a Meraki router and play if you have any interests in community wireless networks. I do, in San Francisco.
Meraki also invites you to develop and extend firmware, build new applications that run on the Meraki firmware. Examples to date have included integration with embedded devices like video cameras and detailed log collection for network research.

For Wireless meshes this is company to watch. Visit to Roofnet says that all Roofnetters are taking sabbatical at Meraki. That is a lot of Brain Power!

GigaOM article
Geemodo: San Francisco Google Free Wireless gets a Blow, almost literally
Meraki Networks
MIT Roofnet

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