Wednesday, January 31, 2007

Use Gizmocall to call me or anyone free


I learned about this at GigaOM site and tried to call him by his name! Did not work.
The Gizmo Project and the Sipphone Inc have brought out an web based (well Flash based) dialer that does not work on all computers. It will work with any browser that supports Adobe flash and a flash plugin from Gizmocall. Windows works, Linux not, Mac OS X did not try yet.
As I recall Similar application, Jajah did not have this problem. I could run it from my Linux box, which I started using more and more since Uncle Bill at Microsoft started to peek into what you are doing through his windows running on YOUR computer.
On Linux computer you get the following message;
Gizmo Call does not yet work on Linux computers. We like Linux and we use Linux and we’re going to make Gizmo Call work for Linux. Adobe just recently released Flash 9 for Linux which Gizmo Call requires which is why we have been delayed in releasing Linux support for Gizmo Call. You may want to try the Linux version of Gizmo Project available at http://gizmoproject.com which works on most Linux distributions. You can also try the Nokia n800 Linux tablet which runs Linux and Gizmo Project. Go Linux!!

Well I had Gizmo on the notebook already.

Anyway it seems a good project and allowing web, email and blog embedding the call to your phone is a goo idea. Bellow is sample.

Test Gizmocall;(it will take you to gizmo test call;
Call Me for FREE

So having partyline is also a good idea but having to download and install plugin is something that I do not like to do. It might be the case with most other people too.

Links;
GigaOM Making browser based calls
Gizmocall Freecalls to anywhere

Tuesday, January 30, 2007

Python in your SIP

PJSIP blog has an article about up coming (may be not! ;)) Python SIP User Agent, a softphone.
I like the reasons he gives about using Python and I like the WxPython a lot. I am sure the SIP UA will be great addition to SIP based soft phone we currently have. Visit PJSIP blog and read more about the Python SIP User Agent.

Links;
Python in your SIP
PJSIP Project

INTERNET TELEPHONY® Magazine's 2006 Product of the Year Awards

NTERNET TELEPHONY, a leading VoIP authority since 1998, is awarding the product of the year in many a fields related to VoIP IP Telephony during TMC’s IMS Expo is recognized as a distinctly important forum on next generation networks, and is being held at the Broward County Convention Center in Ft. Lauderdale, FL.

"INTERNET TELEPHONY is the only magazine that focuses on providing information in IP communications technologies. INTERNET TELEPHONY magazine provides readers with the best information necessary to learn about and purchase the equipment, software and services. INTERNET TELEPHONY offers rich content from solutions-focused editorial content to reviews on products and services from TMC Labs and Miercom. The only BPA-audited publication that's 100% dedicated to providing the highest quality content for the IP communications market, INTERNET TELEPHONY has 55,000 dedicated readers." And the Editor-in-Chief, Rich Tehrani, has been a leading figure in the communications field for a long time.
Internet Telephony Mag will have full list of awards on it's February issue. Many a companies and products have been recognized and awarded for their work in the field of VoIP IP Telephony.
I have been a reader of the magazine for almost a decade and it is one of the print magazines that I still read!
The 2006 award receivers include;
Dialogic, Teletech, Nominium, Patton, SIPdev.org, objectworld, Centillium, CTI Group to name a few.
The complete list will be available online at ITMAG site. Follow the links.

Internet Telephony Magazine
TMCnet

Monday, January 29, 2007

The History of the Telephone

I think there is more to The History of the Telephone than Alexander Grahm bell, the first name that jumps to our mind.
According to an article I found,
The invention of the telephone has a long intriguing and contested history. There exists great dispute over who deserves credit as the first inventor of the telephone.

According to dictionary.com the telephone is “An instrument that converts voice and other sound signals into a form that can be transmitted to remote locations and that receives and reconverts waves into sound signals.” The word telephone originates from a combination of the Greek words “tele” meaning “afar, far off,” and “phone” meaning ““sound, voice.”

Some historians suggest Francis Bacon predicted the telephone in 1627 in his book New Utopia, where he described a long speaking tube. This might have been foreshadowing since not enough was known about the transmission of electricity to make the concept a reality in that era. It was not until 1854 that a French investigator Bourseul suggested that transmitting speech electrically over distance could be possible.

Antonio Santi Giuseppe Meucci is the earliest endorsed claim to the invention of a voice communication apparatus. Meucci constructed a form of telephone in 1857 as a way to connect his second-floor bedroom to his basement laboratory. In Italy, Meucci is recognized as the inventor of the telephone. The Enciclopedia Italiana di Scienze, Lettere ed Arti recognizes Meucci as the original founder of the telephone in 1860.

European physicists credit German inventor, Philip Reis, as the first to transmit a sentence by telephone in 1860. Reis demonstrated his device 16 years before Bell took out a patent for a similar device. In 1872, Prof Vanderwyde demonstrated Reis's device in New York where it was supposedly seen by Thomas Edison and Alexander Graham Bell. On March 22, 1876, a New York Times editorial entitled "The Telephone," endorsed Philip Reis as the first inventor.

Bell evolved ideas from Reis's device in his subsequent development of the telephone. Bell enlisted Thomas Watson, an experienced machinist, to assist him in his research. In 1876, Alexander Graham Bell patented the electro-magnetic transmission of vocal sound by undulatory electric current.

Follow the link and read more on the history of telephone. I would like to know what someone would read in 100 years, under the same heading.

Links;
History of telephone according to telecost

SIP Magazine January 2007 issue is out!

TMCnet's SIP Magazine is now online. I have written about this before but still would like to let you know about it because I enjoy reading it.
This months issue brings you;

SIP Big in 2007
By Richard Tehrani, Publisher, Pubilsher's Outlook

• Session Border Control
SIP has gone through many changes since its humble beginnings. Still, SIP retains just enough of its simple former “endpoint-to-endpoint” characteristics to make it a bit difficult to pass across network borders and through firewalls.
• SIMPLE in the Enterprise
If and when IM (Instant Messaging) and presence eventually become an IETF standard, the SIMPLE protocol will undoubtedly be the principal component, thanks in part to its ability to integrate IM/presence with voice, video, data-sharing, and other elements of conferencing and real-time collaboration.
SIP Big in 2007
Both the above articles are by By Richard “Zippy” Grigonis.
There are many other information like regular columns;
Speaking SIP
• State of Emergency: VoIP and 911
By J.D. Rosenberg

Presence Enabled
• Present and Accounted For
By Joe Hildebrand

Special Focus
• SIP in America's Heartland - NNU Deploys the First Ever Inter-Tel 7000
By Greg Galitzine

I think it is worth a visit.

LINKS;
SIP Magazine



Sunday, January 28, 2007

A Standard for VoIP peering?

Anglero at Telecom's Tsunami tells us about OSP from VOIPUSER.ORG, OSP is as old as VoIP but has been used very little. All the occasions that I know, the OSP was used for billing purposes. When I ran a H323 gatekeeper, for servicing some of the big time providers requested me to provide call information via OSP. I had to get a SUN box just to run the OSP code, because it did not run on anything else at that time.

OSP is an Operational Support System (OSS) protocol well suited for managing inter-domain routing, access control and accounting of SIP transactions. OSP uses the communications protocols below to convey messages. The content of an OSP transaction is an HTTP message formatted according to the standard for MIME. Individual components in the message are XML documents and the message may be signed with an S/MIME digital signature.

So VoIP peering among all the Providers and routing of SIP information among users will provide an stable standard among VoIP users. As Anglero says, OSP guarantees that when "Pamela.Anderson@voipuser.org" is calling you, it is actually Pamela Anderson.

Links;
Telecom's Tsunami: Setting a new standard in VoIP Peering -VoIPUser.org
VoIPUSER.ORG

Thursday, January 25, 2007

Mark Spencer, Asterisk Creator, Presents AsteriskNOW

Although many times I have gone through the same procedure, quite a few times, it is good to see and hear Mark. Hello people it is this easy, try it out. Is this the end of TrixBox? I don't think so! My TrixBox still got a few more tricks under it's sleeves. But I think I will follow Mark sooner or later.



Links;
AsteriskNow
Trixbox

Asterisk: The Future Of Telephony under under the Creative Commons license


O'Reilly Media has released the Asterisk: The Future Of Telephony under under the Creative Commons license. Kudos goes to O'Reilly Media and the three authors, Jim Van Meggelen, Jared Smith, and Leif Madsen. Download links and Authors links are under the links at the bottom of the post.
Once I know the rules, I will post it in IPTELEPHONY in Google Groups.
Asterisk: FOT has received wide attention in the VoIP and IPPBX realm. It is a very well put together book that appeals to new comer to Asterisk as well as to the IPPBX pro. For information, I have listed the contents of the book below.

Contents of the Asterisk: The future of Telephony
Foreword

Preface

1. A Telephony Revolution
VoIP: Bridging the Gap Between Traditional Telephony and Network Telephony
Massive Change Requires Flexible Technology
Asterisk: The Hacker's PBX
Asterisk: The Professional's PBX
The Asterisk Community
The Business Case
This Book

2. Preparing a System for Asterisk
Server Hardware Selection
Environment
Telephony Hardware
Types of Phone
Linux Considerations
Conclusion

3. Installing Asterisk
What Packages Do I Need?
Obtaining the Source Code
Compiling Zaptel
Compiling libpri
Compiling Asterisk
Installing Additional Prompts
Updating Your Source Code
Common Compiling Issues
Loading Zaptel Modules
Loading libpri
Loading Asterisk
Directories Used by Asterisk
Conclusion

4. Initial Configuration of Asterisk
What Do I Really Need?
Working with Interface Configuration Files
FXO and FXS Channels
Configuring an FXO Channel
Configuring an FXS Channel
Configuring SIP
Configuring Inbound IAX Connections
Configuring Outbound IAX Connections
Debugging
Conclusion

5. Dialplan Basics
Dialplan Syntax
A Simple Dialplan
Adding Logic to the Dialplan
Conclusion

6. More Dialplan Concepts
Expressions and Variable Manipulation
Dialplan Functions
Conditional Branching
Voicemail
Macros
Using the Asterisk Database (AstDB)
Handy Asterisk Features
Conclusion

7. Understanding Telephony
Analog Telephony
Digital Telephony
The Digital Circuit-Switched Telephone Network
Packet-Switched Networks
Conclusion

8. Protocols for VoIP
The Need for VoIP Protocols
VoIP Protocols
Codecs
Quality of Service
Echo
Asterisk and VoIP
Conclusion

9. The Asterisk Gateway Interface (AGI)
Fundamentals of AGI Communication
Writing AGI Scripts in Perl
Creating AGI Scripts in PHP
Writing AGI Scripts in Python
Debugging in AGI
Conclusion

10. Asterisk for the Über-Geek
Festival
Call Detail Recording
Customizing System Prompts
Manager
Call Files
DUNDi
Conclusion

11. Asterisk: The Future of Telephony
The Problems with Traditional Telephony
Paradigm Shift
The Promise of Open Source Telephony
The Future of Asterisk

A. VoIP Channels

B. Application Reference

C. AGI Reference

D. Configuration Files

E. Asterisk Command-Line Interface Reference

Index

Links;
download book as a single entity,a PDF file.(4.5MB) USA1 USA2 UK NL
Download each chapter is a seperate PDF file (3.1MB) USA1 USA2 UK NL
O'Reilly Media
Jim Van Meggelen
Jared Smith
Leif Madsen

Asterisk IP PBX for intel Mac

Nerd Vittles has released the version three of the All in one, plug and play IPPBX for the Intel Mac.
I have tried it on my Mac Mini and I love the performance. It is somewhat equivalent to my Asterisk server running on an Intel PC. But that is a Pentium III with lots of RAM, 4GB, and CentOS 4.2 as the base. It has been running fine for a while now.
I used the VMWare Fusion, but I will also try on the Parallels version once the my Parallel arrives.
Folks, I am not running this for production. Just as a test. But I tested it with real world scenarios.
Mac system Info;
Mac Mini 1.83Ghz dual core
2GB ram (maxed)
160GB HD (Notebook SATA)
Mac OS X 11.4
VMware Fusion
NV build 3 of Asterisk IP PBX.
By the way root password for VM is password on my image.

Links;
Nerd Vittles Intel mac IPPBX

Wednesday, January 24, 2007

2007 a year of Hyperdisruption

I was at an IDC conference in San Francisco this morning. There were a few analysts presented visions of the future for software development and IT in general.
Service Oriented Architecture (SOA), Software as a Service (SaaS), the growth in the SMB IT marketplace, were the main topics that were covered or the ones that I paid attention.
IDC also released a free version of the paper that sums up the discussions and views.
What I see glowing in the darkness is the SMB market that needs to wakeup. Once it wake up it will need to communicate. No further than the word communication lies th modus of convergence.
So in this SMB convergence space, what does it mean for an IP Telephony or VoIP company. I think the free paper is good starting point and then it is up to you to go further.

Links;
IDC free paper, Top ten predictions for 2007


Tuesday, January 23, 2007

The Asterisk Appliance Developer Kit


The Asterisk Appliance is a standalone embedded PBX. Targeted for small to medium businesses (2-50 users), and remote branch offices of larger organizations (2-50 users per site), the Digium Asterisk Appliance will feature the commercially licensed Asterisk Business Edition software and the first Digium-developed Asterisk GUI framework.
The Asterisk Appliance appears to take advantage of the Blackfin DSP's microcode programmability by implementing echo cancellation, and possibly other telephony functions, in hardware.
The Appliance's I/O includes eight analog ports (FXS, FXO), a WAN port, four LAN ports, hardware echo cancellation, and a "craft port" for debugging. Expansion is available through a CompactFlash slot suitable for voicemail storage cards or wireless radio peripherals.

The Asterisk Appliance Developer Kit was designed to allow developers to begin working on solutions based on the Asterisk Appliance before its general release. By developing new business applications using the AADK, kit adopters will have an opportunity to become authorized Asterisk Appliance Partners, qualifying for special programs, pricing and priority availability on production products built on this platform.

Purchase of the AADK includes:

* (1) Asterisk Appliance
o Complete Asterisk server with Asterisk GUI framework
o Up to eight analog telephony ports, configurable via modules
o One 4- port FXS module & two 4-port FXO modules
o Slots for Compact Flash and MMC add-on cards
o 8 MB onboard flash
o 64 MB onboard RAM
o 5 Ethernet ports (4 LAN, WAN)
* Cables for all port types
* IP-430 Polycom phone
* CD with all software
* Documentation and specifications
* How-to manuals
* Digium support details
* Asterisk memorabilia
It is priced at around $4K
Get more info and purchase this at Digium, follow the links.

Links;
Asterisk Appliance Developer Kit


Masquerade your Asterisk Server with SIProxd or Firewalled Asterisk

Siproxd is an proxy/masquerading daemon specially designed for SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections possible via an masquerading firewall. It allows SIP clients (like kphone, linphone) to work behind an IP masquerading firewall or router. It could also be installed on the firewall itself. Installation is very simple as well.

SIP (Session Initiation Protocol, RFC3261) is used by Softphones and Hardphones (Voice over IP) to initiate a VoIP communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers.

STUN servers are used to help SIP clients to figure out its public visible IP address and use this one instead of th non routable IP address. As a drawback, usually on the firewall, a wide range of ports must be opened up for the incoming RTP traffic and the SIP client must also support STUN, which most of them do.

Siproxd provides another approach (application layer proxy) and places itself as outbound proxy in between the local SIP client and the remote SIP client or SIP registrar. It rewrites the SIP traffic on the fly and also includes a RTP proxy for incoming and outgoing RTP traffic (the actual audio potion of a SIP based VoIP call). The port range for receiving RTP data is configurable, so the firewall needs to allow /open only a small port range.

Now here is the Masquerading Asterisk Server;

The Asterisk server will register itself as a SIP UA (Client) to an external SIP registrar. In this example sipphone.com is used as the external SIP provider. As Asterisk does not allow to specify an SIP outbound proxy we need to use transparent proxying. The context values of the asterisk configuration needs to be adapted to fit your needs.


Various Configuration files;

siproxd.conf:

if_inbound = eth0
if_outbound = ppp0
hosts_allow_reg = 10.0.0.0/24
sip_listen_port = 5060
daemonize = 1
silence_log = 1
log_calls = 1
user = siproxd
registration_file = /var/lib/siproxd_registrations
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low = 7070
rtp_port_high = 7089
rtp_timeout = 300
default_expires = 600
debug_level = 0
debug_port = 0

Firewall configuration (iptables):

# redirect outgoing SIP traffic to siproxd (myself)
iptables -t nat -A PREROUTING -m udp -p udp -i eth0 \
--source 10.0.0.11 --destination-port 5060 -j REDIRECT
# allow incoming SIP and RTP traffic
iptables -A INPUT -m udp -p udp -i ppp0 --dport 5060 -j ACCEPT
iptables -A INPUT -m udp -p udp -i ppp0 --dport 7070:7080 -j ACCEPT

Asterisk configuration (SIP related part):

Note: Very important are the fromuser and fromdomain keywords in the client section. They are required to have Asterisk send the correct From headers in SIP dialogs.

sip.conf:

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
defaultexpirey = 900

; codecs
disallow=all
allow=gsm ; 13 Kbps
allow=ulaw ; 64 Kbps
allow=alaw ; 64 Kbps

; SIP Trunk to sipphone.com you can use you own outbound SIP trunk here
; the SIP number is taken randomly for this example
register=17476691234:@proxy01.sipphone.com

[17476691234]
type=user
nat=never
context=from-pstn
canreinvite=no

[sipphone1]
username=17476691234
type=peer
qualify=2000
host=proxy01.sipphone.com
fromuser=17476691234
fromdomain=proxy01.sipphone.com
context=from-pstn
canreinvite=no
secret=

; local SIP extensions
[200]
username=200
type=friend
secret=XXXXXX
qualify=500
port=5060
pickupgroup=
nat=never
mailbox=
host=dynamic
dtmfmode=rfc2833
disallow=
context=from-internal
canreinvite=no
callgroup=
callerid="Extension 200" <200>
allow=all

There you have it, a firewalled Asterisk server or Trixbox.

Links;
SIPROXD at Sourceforge.net

Sunday, January 21, 2007

MV-370, MV-372 GSM<->VOIP gateway


MV-372 is a 2 channel VoIP GSM Gateway for call termination (VoIP to GSM ) and origination (GSM to VoIP). It is SIP based and compatible with Asterisk. It can enable to make 2 calls simultaneously from IP phones to GSM networks and GSM network to IP phone.
This will enable one to save a lot on Mobile phone calls. The gateway could be found on EBay at the moment but I am sure it will begin to arrive in various VOIP gear shops and distributors.
The manual is available but it is bit hard to setup. But do not worry too much about it as the interface is web based and Good old VoIP-info.org has an article how to configure it in detail.
MV-370 supports single line and MV-372 supports 2 Lines in either path, VoIP to GSM or GSM to VoIP.

Links;
MV-37X GSM<->VOIP gateway
VoIP info org's article on MV-370

Skype Conference phone kit by Trendnet, TVP-SP4BK.


Trendnet has released a bluetooth enabled Conference phone for Skype, TRENDnet ClearSky Bluetooth VoIP Conference Phone Kit for Skype (TVP-SP4BK), shown above. It includes an bluetooth adapter for your computer that connects to the internet and Skype.In addition to looking very cool, The best part of the phone is that it allows hands-free VoIP conferencing via Bluetooth. The phone is available now for about $170.
The unit includes an integrated LCD screen and touch pad that lets users view Skype contacts, call history, status and SkypeOut credit balance. Also to be noted is that the phones battery is charged via USB cable.
Bluetooth has a range of about 300 feet according to the manufacturer and the fully charged battery has 4 hours of talk time and up to 72 hours of standby time.

You can also use the TVP-SP1BK as a bluetooth wireless extension to skype if you want some thing like a cell phone.

Links;
TVP-SP4BK,
TVP-SP1K

Saturday, January 20, 2007

Webinar, Building an Open Source IP-PBX with trixbox 2.0

Trixbox maker, Fonality announced the first of a series of free webinars to help users make the most of their trixbox installation. This webinar series will cover topics for beginners to advanced users. The first webinar entitled “Building an Open Source IP-PBX with trixbox 2.0” will cover the basic aspects and features of trixbox. This is a good introduction to the features, installation, and basic configuration of trixbox. The agenda for this webinar includes:

* System requirements
* Installation
* Hardware and ITSP support
* Skills required
* Basic configuration
* Basic troubleshooting
If you are Trixbox or Asterisk user, I am sure you will gain valuable lessons and techniques from this type of activities. Follow the link to get yourself registered.

Links;
Webinar, Building an Open Source IP-PBX

Thursday, January 18, 2007

Joost invite is here, and I am Joosting!


Update:
Thank you for all the requests. Joost has not given me any invites yet. When they do, depending on the number, some of you who requested it will be chosen. I don't know how yet. Thanks

Although I knew a few people who had Joost invites, I did not get any. But finally yesterday I got an invite from Joost itself. Thanks Joost. I really see a difference in Joost and other video services I have seen. I could chat with someone already while watching a video.
Before writing more about Joost, I need to verify the NDA again to make sure I am not breaking any rules or regulations.
All I can say is that experience is super. quality is great and I will be Joosting more for sure.
I think the other social attraction in the net is here. Go Joost!
(Watch out for tag "Joosting" started here.)
Since this is Bay area, Go E-40 and all hyphies of hyphy fame!!

Links;
Joost, formerly Venice Project



Get your iPhony today

READ THE UPDATE BELOW;


Webtown gave me a link and info about an iPhone mock up that got Apple steaming. The Gerlog has the scoop and I even left a comment.
I think Apple has right to it's IP but to go after some imitations that does not come even close to that over priced, now defunct (what??? it is AT&T now) Cingular only iPhone. It looks like all the icons a too generic anyway.
Anyway looks like Apple shares did a little dive today, I wonder why, Apple put those brilliant engineers and designers in front and put those over zealous lawyers behind. Why don't you spend time on Cisco?

Talking about iPhone, Apple have you found a way to keep grease out of the screen yet?

UPDATE;
The links leading to the iPhnony have disappeared but clever people can still find it. May be you can find how to here;
weseepeople: VOIP IP Telephony: Get your iPhony today

Links;
iPhony at gearlog
Webtown Apple Iphone Hacked.

Skype lets you speak to your blog

Skype lets you speak to your blog, well Skype along with little bit of work by some hardworking people. It also does not have to be Skype, any telephony application or a device that could call and leave a voice message.
Skype Journal gets the credit for leading me to this news and The Speak-a-Blog Blog! by Simon Crowfoot. This is where you see an examples of speaking to a blog. The technology is provided by Spinvox.
Voice to text has come long way since I played with Voice recognition a decade ago.
I will post try to post on The Speak-a-Blog blog via Voice and see how it works.
You can post a test message via following methods;
0141 238 1075 (UK), 1 9737945439 (US), Skype; speakablogblog

Links;
Skype Journal talk to your blog
The Speak-a-Blog blog!
Spinvox

Wednesday, January 17, 2007

Avaya offers a tender to Acquire Ubiquity Software

Avaya news
Avaya Inc. (NYSE:AV) today announced a tender offer by its wholly-owned subsidiary to acquire Ubiquity Software Corporation plc (LSE:UBQ.L). Ubiquity's core software product is one of the leading software platforms for the development and delivery of SIP end-user applications.

The offer price is 37.3 pence in cash for each Ubiquity share. This values the entire issued and to be issued share capital of Ubiquity at approximately GBP 74.3 million (approximately $144 million)1 after adjusting for the assumed proceeds from the exercise of options over Ubiquity shares.

Ubiquity develops and markets SIP-based communications software for fixed and mobile communications service providers, systems integrators, independent software vendors and channel partners. Ubiquity's range of products has been developed to take advantage of the telecommunications industry's migration toward all-IP networks.

"We believe that the addition of Ubiquity's next generation software platform to Avaya's portfolio will help customers and developers enhance the integration of communications technologies and business processes," said Micky Tsui, vice president, global communications solutions, Avaya. "We believe that Ubiquity bridges a wide range of fragmented technologies that cost customers' money, time and speed-to-market."

Links;
Avaya news release
Ubiquity Software

Mobiboo aims to unseat Skype, in UK, with "all you can call" plan


Tech Digest's Andy tells me that Mobiboo has announced a low cost unlimited calls tariff for both home and business users, allowing unlimited free calls to UK geographical numbers, and low cost call rates to UK mobiles and overseas numbers.

For £39.99 per year (personal) or £89.99 per year (business), and the download of Mobiboo's Dashboard PC software, your computer becomes a screen phone, similar in nature to what Skype offers.
Read More on Andy's post, follow the link.

Links;
All you can eat VoIP on Tech Digest

Get your Mobiboo here

Broadsoft and BEA Systems paves new ways for VOIP

SAN JOSE, CA and GAITHERSBURG, MD, January 16, 2007 - BEA Systems, Inc., a world leader in enterprise and communications infrastructure software, and BroadSoft, Inc., a leading provider of VoIP application software, today announced a broad-ranging strategic alliance. This alliance includes joint development, sales, and marketing of solutions that will integrate the BroadSoft® BroadWorks® suite of VoIP applications with the BEA WebLogic® Communications Platform product family.


The BEA-BroadSoft partnership is a multi-year, multi-phased strategic alliance. Under the alliance, the two companies are conducting joint sales and marketing efforts, and BroadSoft has already begun to integrate and port the BroadWorks software platform with BEA WebLogic® SIP Server.


BEA and BroadSoft are teaming up to provide service providers the ability to offer industry-leading applications over an innovative, standards-based platform. The combination of BroadWorks software and BEA WebLogic SIP Server-a converged Java EE-SIP-IMS application server-will enable service providers to deploy revenue-generating services with a rich set of VoIP capabilities. It will also be designed to provide a flexible and extensible architecture that will work in an all-IP and IMS environment in the future.


"This alliance brings together two industry-leading telecom application platforms, which will allow service providers to deliver BroadWorks-based fixed and mobile VoIP services with IMS enablers and capabilities that can be hosted on BEA WebLogic SIP Server, " said Ken Rokoff, vice president of business development at BroadSoft. "Until today, no VoIP application platform in the market has integrated with a converged Java EE-SIP-IMS application server. This unique offering can help drive new service revenues for service providers."


"Service providers are demanding VoIP solutions that can shorten time-to-market for enhanced communication services over NGN and IMS architectures," said Mike McHugh, vice president and general manager, BEA WebLogic Communications Platform, BEA Systems. "By bringing the BroadSoft BroadWorks solution together with the BEA WebLogic Communications Platform product family, BEA and BroadSoft will provide a feature-rich set of VoIP service enablers designed to help operators implement new revenue-generating services more rapidly and more cost-effectively."


For more information about BEA WebLogic SIP Server, and the BEA WebLogic Communications Platform, please visit http://www.bea.com/sip and http://www.bea.com/wlcom, respectively.


The BEA WebLogic Communications Platform product family, including BEA WebLogic SIP Server, is a key component of the BEA SOA 360 platform. Announced in September at BEAWorld 2006 in San Francisco, the BEA SOA 360 platform is designed to deliver the industry's most unified SOA platform and spans all three of BEA's product families, AquaLogic®, WebLogic®, Tuxedo and the company's newly unveiled SOA collaborative tooling environment, BEA Workspace 360. (See Sept. 19, 2006, press release titled "BEA Announces SOA 360o; Industry's most unified SOA Platform to transform and optimize business.")


BroadSoft's IMS-compliant BroadWorks platform provides a comprehensive range of VoIP applications, including hosted PBX, IP Centrex, mobile PBX, business trunking and residential broadband services fully integrated into a single VoIP application platform. BroadWorks provides these applications with the reliability, redundancy, scalability and regulatory capabilities required to deliver carrier-class service.


News Source BroadSoft

Links;
BroadSoft
BEA Systems

Monday, January 15, 2007

Joost gives a boost to Skype Founders, again...

Doug Berger over at Gadgetel tells me that Venice Project has changed the names and moved new hosting site/s in the name of Joost. In europe, this will sound like "Yoost" and I like the sound and the idea behind it.
If you are looking for invites to Venice Project... OOps to Joost, go over to Gadgetel and leave a comment. They might pick your comment to give the free invite. I did!

Links;
Gadgetel Joost invites!
The Venice Project aka Joost

oPhone Please....

Doc Searls sends out request for a replacement for all closed up iPhone by Apple.
"It is time for an equipment maker to not only make an open phone that is open to all kinds of development, but to turn their carriers into "dumb pipes" for their own good.".
I would love an oPhone..

Links;
Doc Searls oPhone call out

Tags: ,

Who used iPhone before 1996? Vocaltech did it seems.

Flat Planet and a phone lead me to a Jeff Pulver's article about iPhone. It seems that before Cisco's iPhone was registered after the fact that Vocaltech's and Jeff use of the name iPhone.
As the articles mention, both the authors have had connection to this iPhone name. I also did some search but could not come up with an instant where iPhone was used for a cell phone. Does Cisco's trademark discription include cell phone or only internet phones?
I think it is going to be a interesting case unless Apple and Cisco settle out of the court.

Links;
Flat Planet and Phone..

Jeff Pulver + iPhone



Asterisk experts, Help out to finish Asterisk Cook Book

Bruce Stewart at O'Reilly Emerging Telephony, ETEL Community, informs us of a project targeting Asterisk Cookbook.
"O’Reilly Media is in the process of building the Asterisk Cookbook, and we’d like to invite the Asterisk community to contribute. We’re looking for two kinds of contributions. First,
we’re looking for problems you’d like to see solved in the book. If you need to make Asterisk do something and just can’t figure out how, let us know. We’ll try to solve the problem for you. Second, we’re looking for more advanced Asterisk users to contribute solutions to problems that they’ve faced."
I think this is cool idea. There are many a places that people go to get help on Asterisk. I do frequently comb through many a forums to find answers my questions/problems. Now there is another place to go or read once the book is published. Sometimes it helps to remember if you read it in book. I hope O'Reilly will keep the given wiki site live after the book is published. Follow the links below for invitation and the Wiki site.
You will need to create an Account in order to seek solutions or provide solutions.

Links;
Asterisk cook book invitation.
Asterisk Cookbook Wiki site



Gas Pump with WiFi dispenses MP3s, could I also make a VOIP call?

Geemodo reports that a new Gas Pump demonstrated will have WiFi access and MP3 dispenser. If the gas pump is networked, obviously, then users with VoIP phones, WiFi, will be able to make calls near Gas Stations, Like good old Pay phones! that adorn every Gas Station here in USA.

Links;
Geemodo: Gas Pump with WiFi dispenses MP3s

Tags: , ,

Saturday, January 13, 2007

FreePBX reaches 2.2.0, FreePBX 2.2.0

Rob has managed to release FreePBX 2.2.0 when I was not looking. FreePBX 2.2.0 was released on January 5th and you all may already have it running. I will start on coming Tuesday (it is a Holiday for us on Monday, honoring Dr. Martin Luther King)
First of all if you have already installed and and having problems with Enum lookups, Rob directs us to a correction in the extensions.conf file.
edit line 428 of /etc/asterisk/extensions.conf and put a “!” before the = sign. The line should read (when fixed):

exten => s,1,GotoIf($["${ARG3}" != “”]?PASSWD:NOPASSWD); arg3 is pattern password

Now that is aside, There is a bunch of important changes that happens in FreePBX 2.2.0 and instead if me putting up snippets of Rob's text here I let you go over to FreePBX site and read the complete article with all the comments etc. Also if you have not downloaded it, it is the time to do so.

Links;
FreePBX release notes on FreePBX 2.2.0

The "virgin aura" from Skype is gone... But it is skype users who are nailed.

Please replace "nailed" with appropriate word.
I was revealed to the fact that Skype has become Skype Bell. Even though Skype told us earlier that it will not become another telephone company, it has just started doing so! Thomas over at Telecom's Tsunami lead me to the facts stated below. You will also find the link to his original post at links.
".. this evening Skype announced the one of the oldest tricks in the 150 year old Telco "how-to suck more money from your existing subscriber base" manual by announcing for every SkypeOut call made, they will charge a $0.039 per call "connection fee"."

If 5% of Skype's calls are Skypeout calls, and assuming 1.5 Billion calls per year is made through Skype, this old Telecomm schools trick of charging a connection will add about $3 millions to Skype pockets.
But this might work well with investors. And it is of course not so good for the skype users.

Links;
Telecom's Tsunami: Skype 2007: The "virgin aura" is gone...

Friday, January 12, 2007

iPhone get hammered, on battery, GSM and pirce

I myself got tagged into this iPhone PR for Steve Jobs and can't get away. I think this will be my last iPhone post for a while.
After those ooh and aaghs, now comes the hammering. First I got the dose from Scobleizer's post about The "iPhone reality distortion field" which lead me to read first five reasons by Paul Kedrosky.
I think either Apple and the gang will fix these issues, specially the battery life, the Cingular only, and the price.
I can do everything that iPhone says it does and more with $199.00 phone I got from Sprint. (I got button to press!) Imagine an iPhone if someone used it while eating a good burger! and before long there will be iPhone rub off marks on suits and shirts! Any way I hope the first batch of users will be the guinea pigs so we can get better and cheaper iPhones later, in 2008 may be. Good luck iPhone!

Links;
Paul Kedrosky 5 reasons not to like/get iPhone
Scobleizer's post completing the list


Technorati 100,000 barrier passed

After checking in almost for two years, VoIP IP Telephony fell below 100,000 rank today. It might not mean anythining to you or you may just not care, but it is very improtant to me. I will post again when I drop below 50,000! It stands at 96,155 but all familiar with Technorati know that this figure is dynamic, means it could change easily. If you have a blog, and still haven't registered at Technorati, please do. You will be happy among 55 million other bloggers.

Links;
Voip IP Telephony Technorati Links


iPhone, Demo, Google maps, and GPS

Blog de alepod has published a iPhone presentation demo by Steve Jobs. If you have not seen the demo you can go there or find it on youtube.
The question is Google maps are great but will we have GPS. The rumors about Apple Macs getting GPS enabled have been known for a while.Also known is that iPhoto code has links to Google Maps. New Macs will have a GPS chip incorporated and the Mac OS X.. will be aware of it.
The Apple Blog is happy to blog about Apple GPS rather than iPhone, so I added two together.
Since iPhone runs Mac OS X.., the GPS features are builtin to iPhone already or will ve have to wait for the next version of iPhone.
I am sure if it was enabled, Steve Jobs would not have let it pass without showing it. So GPS enabled iPhoto on your iPhone might be for those who wait. Unless one has many a $599 to spend.

Links;
Blog de alepods iphone presentation
The Apple Blog peels Apple GPS

Thursday, January 11, 2007

BlackBerry or (i)Phone, iPhone, what is your pick?

Tom Van Aardt from MyADSL lead me to a poll held by ZDNet's Russel Show;
if you could have one for free: Apple iPhone or BlackBerry Pearl?
I voted for ???? and it has 7% of the vote. (no it is not a BB). Go Vote yourself and see who the winner is.

Links;
MyADSL post on iPhone Poll
Russel Show's Poll, iPhone or BlackBerry?

SMC enhances VoIP IP Telephony Portfolio

Like all other network gear manufacturers, SMC is not letting the opportunity in the VoIP IP Telephony pass by. SMC’s growing portfolio of VoIP products includes a host of network-building switches, routers and adapters, as well as many specialized devices that are available now. Those specialized devices that are available now include the SMC7908VoWBRA, a Wireless ADSL router with built-in Voice over IP gateway; the SMCWTVG Wireless Travel Voice Gateway, which integrates the function of a wireless gateway and a VoIP ATA (Analog Telephone Adapter) into a compact, take-along device; and the recently-introduced WSKP100 Wireless Internet Phone for easy, portable, computer-independent Skype calling. In Q4, 2006, SMC will add to its line of VoIP products by shipping a desktop, SIP-based IP phone and a full-featured IP PBX. The IP PBX integrates analog trunk and extension call management with Standard Internet Protocol (SIP) trunk and extension management in an all-in-one PBX unit that is scalable to multiple sites and extensions. And, its ability to support wireless or wired devices makes the reality of a mobile office seamless as users move between mobile and desk-based phones. The Desktop IP phone will feature an intuitive LCD display, programming via the keypad or built-in Web interface, and two 10/100 Ethernet ports for connecting the phone to the local LAN and to a PC for added convenience.

“As a global company, we know first-hand the huge benefits of VoIP,” said Tony Stramandinoli, SMC Networks ’ Vice President of Global Marketing. “We’ve been our own best beta test site, using IP telephony to make co-workers in Taiwan , Spain and the UK just an extension away. The high-quality and ease of use, combined with cost savings, make colleagues across the globe as accessible as colleagues down the hall.”

For more information about SMC Networks, SMC’s broad line of products for VoIP, or others in its full complement of networking products, visit www.smc.com, or call 800-SMC-4YOU (800-762-4968).


Cisco go after "iPhone" name and sues Apple, but iphone.com lives on.

It is likely that even if Cisco wins the lawsuit and forces Apple to change the name of the iPhone to something else, it will not make much of a difference, at least according Michael Gartenberg, an analyst at Jupiter Research. "If they have to name it something else, it won't sell any less than if it was called iPhone," he said.
People will end up calling it iPhone, any way. Cisco you are set to loose even if you win! So be good!
Cisco said it wanted to keep Apple from "infringing upon and deliberately copying and using" the trademark.

Linksys, a division of Cisco, has been selling wireless products with the iPhone name since early last year, with new products added to the line in December.

Analysts have called Apple's move into the rough-and-tumble cell phone market its most significant yet into the consumer electronics industry.

Apple spokesman Steve Dowling said: "We think this is silly. There are already several companies using the name iPhone for voice over Internet Protocol products.

"We're the first company to use the name iPhone for a cellular phone and if Cisco wants to challenge us on that we're very confident we'll prevail."
News Factor has a lengthier article. Follow the link below.

The funny part is the trademark iPhone registration. Earlier it was 1999, now according to news factor it 1996.

Mike Kovatch of Santa Rosa has owned the domain name www.iphone.com for 13 years, using it for his Internet Phone Co.

On Tuesday, when Jobs unveiled the iPhone, Kovatch temporarily shut his site after getting overwhelming traffic. He declined to discuss the matter in detail, saying, "We've had lots of people contacting us. We're staying in stealth mode until we know what's happening."

Links;
iphone.com
Newsfactor article

Wednesday, January 10, 2007

Apple's iPhone iTV (AppleTV), finally comes.

After announcing dropping of "Computer" from Apple computer Inc, which is now only called Apple Inc (Will there be another Beetles lawsuit?), Steve Jobs unveiled Apples new phone and the AppleTV.
The Apple phone, which I liked instantly, is slightly larger than your average phone, but comes with a 3.5" touchscreen that covers the majority of the front plate of the device. At 11.6 mm, it is not the thinnest phone out there, but it is thin enough to be able to escape the definition of being bulky. Standard equipment include 4 GB or 8 GB of flash memory storage, Bluetooth and Wi-Fi connectivity, a web browser, push email (with Yahoo email). Instead of relying on other software companies, Apple decided to use Mac OS X as operating system for the iPhone, which will be available in a quad-band GSM/EDGE version from Cingular Wireless.

Pricing for the phone is set a $500 for the 4 GB version and $600 for the 8 GB model. It will be available in the U.S. beginning in June of this year.

But while the touch-screen device looks truly gorgeous, sporting just two buttons and a 3.5-inch screen, it does seem awfully ambitious -- even for Apple. Jobs hopes to sell 10 million iPhones, which would give Apple 1% of the global mobile phone market share, in 2008.
The iTV, now called the AppleTV, connects directly to iTunes and can stream content to up to five different Macs or PCs. It comes with what we would consider a smallish 40 GB hard drive and includes 802.11 b/g/draft-n wireless capability. It delivers video content in 720p - which means that storing content on that tiny hard drive may turn out to be a challenge. But then, the AppleTV is reasonably priced at $300.

Links;
Apple's new gadgets

Monday, January 08, 2007

Voice of Telephony, Allison Smith, Interviewed

If you played around with Asterisk, then you know her voice. Even if you have not come near an Asterisk server, you may still have heard her voice, I have, So if you are curious about this voice behind many interactive applications, head over to Ronald Lewis' Pod Cast. You will be able to hear her.
Thank you Allison, thank you Lewis.

Links
Ronald Lewis podcast of Allison Smith interview
Interviews Podcast: Interview with Allison Smith, North America's Leading Voice Over Artist

Nokia's N800 connects you to Skype, Wirelessly


At the CES conference in Las Vegas today, Nokia introduced its next generation widescreen Nokia Nseries multimedia computer, the Nokia N800 Internet Tablet.

Nokia said its new N800 Internet Tablet will permit wireless phone connections through eBay Inc.'s Skype service. The model is available immediately in the United States and selected European countries, and Nokia said the Skype features will be available for download by June.
"Working with the leading mobile handset manufacturer puts us in a unique position to get Skype to the mobile masses," said Eric Lagier, head of Skype's business development in hardware and mobile operations.
Voice minutes over Wi-Fi networks are far cheaper than minutes on cellular networks because they use free radio spectrum and the Internet and do not require large cell towers. Skype has a variety of calling plans, including a $30 annual subscription to make unlimited calls to any regular or mobile phone number within the United States and Canada.
Like its predecessor, the Nokia N800 Internet Tablet is based on Nokia’s desktop Linux based Operating System. The Maemo development platform was launched in 2005 to provide Open Source developers with the tools and opportunities to create innovative applications for use on Nokia’s Internet Tablets. Users of the Nokia N800 will be able to benefit from a wide range of third party applications.

Nokia, meanwhile, unveiled the N76, a light, clamshell model that includes a 2-megapixel camera and 2 gigabytes of expandable memory. But, at a half-inch thick, it is only slightly thinner than Motorola Inc.'s popular Razr model and is nearly twice as thick as Samsung Electronics Co.'s X820, which the South Korean company claims to be the thinnest on the market.
The Finnish company, which makes one in three phones sold globally, has suffered from a lack of thin models in the last two years as consumers sought slimmer phones following the success of Motorola's RAZR and Samsung Electronics Co.'s X820


Links;
More about nokiaN800

Sunday, January 07, 2007

Asterisk drives cars, slot cars now.

The Open source IPPBX, PBX application Asterisk is known to have revolutionized the VoIP, IP Telephony arena. But some innovative souls putting Asterisk now to drive cars, two slot cars on a track.
The project takes advantage of Asterisk, the open source PBX, to take in the phone data and spit that out to a Java based soft phone. This soft phone takes in a SIP stream which I then take amplitude data from. I use that data, scale it a bit an send it serially to an Arduino board.

The circuit itself is quite simple. It uses a transistor to send voltage to the track (12v DC), and I use the incoming serial data to PWM..this allowing for a difference in speed.
The link will give you more info and photos with an up coming video.

Links;
Voice over IP controlled Slot Cars

Saturday, January 06, 2007

San Francisco's Google Free Wireless finally on the go

After 8 months of intense negotiations, finally has reached the Board of supervisors. If approved, the city will have wireless access within a year.

Once the system is operational, the Internet search giant Google will provide free service on the network at relatively low transmission speeds, while EarthLink and others will charge for faster service. The monthly fee will be $21.95, but 3,200 low-income residents will be charged $12.95.
“You can’t continue to rhetorically talk about the digital divide and not do anything about it,” Gavin Newsom, the city’s mayor, said in an interview.
The project has faced opposition from a variety of critics, including some who argued that the city should own the broadband network and some who raised fears of privacy violations. The new agreement establishes some basic privacy protections and requires EarthLink and Google to fully disclose their privacy policies.

Mr. Newsom said that as early as next week he would submit the plan to the County Board of Supervisors, which must give its final approval to the project. The network could begin operating within a year of approval.



Links;
Broadband news

Mark Spencer interviewed by OSDIR.COM

I met Mark a few years ago, at a Linux world Expo, in San Francisco. when I told about Asterisk to people asked me if it was shift+8. Many years have passed since then and now those same people tells me what Asterisk is, but I doubt they know much about Mark. So I direct them to any information about Mark is published.
Howard Wen of ODdir.com has done a good job of interviewing;
"From Analog to VoIP: Asterisk Brings Telephony Together Under One Open-Source Platform"
I like the reason for Asterisk to be. ""I had about $4,000 to start it out with, and I wasn't about to buy a phone system, so I figured I'd just make one," Spencer says."
So started the Asterisk. Follow the link to learn more about Asterisk creator, Mark Spencer.

Links;
Howard Wen's interview with Mark Spencer

Friday, January 05, 2007

Top 10 Software stocks 2006 and Number one is a VoIP IP Telephony company.

Fellow Blogger Burnham has a list of top ten software stocks for 2006. The post gives information about the company and a short comment for each company. The order of listing is based on stock price change and according to the post, To qualify for this list a company had to start 2006 with at least $50M in market cap and its main business had to be selling software as a license or a service.

Here is the first one, which is a Voip Ip Telephony Company.

1. Interactive Intelligence
Price Change: 340% Ticker: ININ
Comment: Pioneer in enterprise IP Telephony saw rapidly increasing sales as enterprises started to take VOIP seriously, especially call centers.

Links;
Burnham's Best: Top ten Software stocks

Internet Telephony Conference and Expo East, 2007


The Internet Telephony Magazines Internet Telephony Conference and Expo will be held in Fort Laurdale, Florida, Broward County Convention Center.
The dates are;
INTERNET TELEPHONY Conference & EXPO East
Conference Dates: January 23-26, 2007
Exhibit Dates: January 24-26, 2007

Rich Tehrani has an exclusive Show Blog that keep you updated as things happen. See links for the Show Blog.
Also Rich tells us that already people from 68 countries have registered for the show. From the past experiences, I have been very happy with Internet Telephony conferences and always try to be at every show due to the rich resources that are in the communication field gathered in one place. A visit to the Show's web site (Links will take you there) will show you the list of sponsors and exibitors. So mark your Calendar, and be there. There is still time to register and find a hotel!
The Conference and Expo will be co hosted with Call Center 2.0, The Premier Technology Event for Call Center Decision Makers and IMS EXPO, The Premier Global Event on IP Multimedia Subsystem.
Links
Rich's article about 68 countries
Rich Tehrani's show Blog
Internet Telephony Conference and Expo

Thursday, January 04, 2007

Web-Meetme, Scheduled conferencing for Asterisk

I came across this package a while back and have been playing a bit with it. But I really got it working after Dan Austin released the version 3.0 just a two days ago. Then again I had to switch to SVN version in order get it completeley working.
Managed conferencing solution using Asterisk, Web-MeetMe and a database-driven scheduler. From the projects readme.
Web-MeetMe for Asterisk
Web-MeetMe is a collection of PHP pages that leverage
the database for scheduling and the Asterisk Manager interface
to monitor and control active conferences.

The scheduler currently supports these features:
+ Conference number conflict avoidance
+ Recurring conferences
* Daily, weekly or bi-weekly
+ Enforces a user password if a admin password
is set
+ Seperate views for past, current and future
conferences
+ Editing future conferences
* Includes a series of conferences
+ Deleting past conferences
+ CDR-like view of past conferences
* Includes a link to conference recordings
+ One click features to-
* Extend an active conference (time)
* Kick out all participants
* Out-call to include participants
+ Optional early join features
* fuzzystart allows a caller to enter a
room early (in seconds, set in cbmysql.conf)
* earlyalert notifies a call that their
conference has not yet started, if they
enter a valid conference number and the
conference is scheduled to start soon.
(in seconds, set in cbmysql.conf)

App_cbmysql
When combined with a database, MySQL or Postgress,
CBMySQL can authenticate a caller, verify that the conference
they are attempting to join is scheduled and enforce the
maximum participent count.

CBMySQL can distinguish between a conference
administrator and a standard user based on pins. Different
options may be set in the configuration file to join the
caller to the conference based on their status. A standard
user might be joined in a listen only mode as an example.

CBEnd
CBEnd is a small PHP script that monitors for active
conferences. When it identifies a current conference, it checks
the scheduler database and enforces the conference duration.
The default behaviour is to announce that the conference is to
end five minutes before the scheduled end time. This allows the
conference owner or administrator to update the conference using
Web-MeetMe to extend the duration.

CBEnd uses the Asterisk Manager interface to issue commands
to the conferencing application. The current commands include
identifying active conferences and the 'Kick all' command.

This script is also responsible for maintaining the
CDR tables. The script is optional, but not using it limits
the overall system functionality.

** Installation
* Dependencies
- Asterisk 1.4 or later
- MySQL 3.23 or later.
!! Other databases may work, but testing
!! and development have been against MySQL
- PHP 4.3 or later
!! Reports depend on mod_gd and GD-2.0.28
!! or later
!! PEAR::DB (php-pear)

* Install Web-MeetMe
Download the latest Web-MeetMe tgz file from
http://www.sf.net/projects/web-meetme and extract it to your webservers
document root.
* Setup the database
Create a table called booking in your database. The
table is described in ./cbmysql/db-tables-v5.txt

* Compile and install CBMySQL
App_cbmysql is now included in the web-meetme package,
located in ./cbmysql. To install just run make; make install

Copy the sample cbmysql.conf to /etc/asterisk and create
a dialplan similar to the one in cb-extensions.conf.sample
Modify the settings to suit your system. The location of the
mysql.sock file is likely not correct, check /etc/my.conf for
the correct location.

Modify the ../web-meetme/lib/defines.php to match you
database settings.

Copy ../web-meetme/phpagi/phpagi.example.conf to
/etc/asterisk and modify it to match the settings in your
manager.conf

If you intend to use the authentication functions,
it is strongly recommended that you use SSL and force all
conenctions to the Web-MeetMe pages to use HTTPS.

* LDAP or Active Directory integration
Edit ./lib/defines and set the AUTH_TYPE to adLDAP.
Set the ADMIN_GROUP to appropriate list, the default is
"Domain Admins", but you might not want the network folks
controlling your telephony systems. Optionally you can
chanege the browser session timeout.

Edit ./lib/adLDAP and replace all instances of
"yourdomain" and "yourserver" with the values for your
network.

* Usage
If all of the steps have been followed, you should be
able to open your browser, connect to the Web-MeetMe page and
schedule a conference. A new conference start time defaults
to the time it is scheduled, so modify it if needed. Any
required fields that are left blank will be auto-generated.
Note the conference room number and password.

Dial the number you have assigned for conferencing, and
try the conference you have just added. If the time and password
are correct, you will be joined to the conference.

You can now use the Web-Meetme monitor function to manage
your conference.

Links;
Web-Meetme download
Asterisk IPPBX


Trixbox 2.0 and Trixbox 2.0 VMWARE image is released

If you are here looking for Trixbox info, Please remember that Fonality killed the Trxbox CE. But there are plenty of alternatives, Follow this link "Trixbox Alternatives", if you are interested.
 

Fonality gives you a new year present, Trixbox 2.0
Fonality has just released Trixbox 2.0. All the Asterisk lovers who were afraid of Linux and configurations of Asterisk, should be happy. Because the installation is even better and easier than previous releases.
Trixbox site ven boasts that “
Installs in less than 15 minutes, and features easy customizable configuration and ‘point-and-click’ updates “, and it is true, even though I only had time to play with the VMWare version.
In addition to the regular CD version, you can also download a VMWare image which you can use o test and play on a virtual machine before deploying to you SMB's or your home IPPBX, like I am doing now.
I promise you you will be happy and your VoIP experience will be a smooth ride.

Here is the press release;
LOS ANGELES – January 4, 2007 – Fonality®, a leader in open source, Asterisk®-based IP telephony systems, today released trixbox® 2.0, a free, easy to use, open source telephony and application platform. The new version, available for immediate download, can be installed in less than 15 minutes, supports multiple languages and provides increased reliability and stability, flexible user customization, and support for a wide-range of hardware vendors. The software also allows the community to upgrade individual deployment components versus having to reinstall from scratch with each upgrade. trixbox.org will also be hosting its first ever training Webinar entitled “Building An Open Source IP-PBX With trixbox 2.0” on January 30, 2007.

“The trixbox community said they wanted a super-reliable Asterisk deployment that removed all the headaches and that is what we strived to deliver in this release of trixbox 2.0.” said Andrew Gillis, founder of trixbox and director of community development at Fonality. “I always envisioned trixbox as a platform that would be both easy enough for people to quickly deploy and stable enough that they could stake their reputation on it. trixbox 2.0 is the realization of this vision.”


trixbox 2.0 comes with a new point-and-click package manager which lets installers, via simple clicks of their mouse, decide which applications they want to install with trixbox. The advantage to the package manager is two-fold: first, it let’s you choose how lean or rich of a deployment you need. Secondly, it informs the installer, over time, of any new updates to any packages within their trixbox installation as vendors release them.

trixbox 2.0 includes a host of packages, or applications, including Apache, an enhanced version of Asterisk, FreePBX, Flash Operator Panel, MySQL, phpMyAdmin and SugarCRM. In addition, the new release provides call detail reports, an endpoint manager, VoIP service provider wizards, deeper application integration with SugarCRM, drivers for Sangoma and Rhino voice cards and support for multiple languages including English, German, Portuguese and Spanish with more to come.

On January 30, 2007 Kerry Garrison, senior product manager for trixbox at Fonality and author of “TrixBox Made Easy,” a step-by-step guide to installing and running your home and office VoIP system, will host the first trixbox.org Webinar, designed to deliver important information to IT managers and integrators who want to deploy trixbox 2.0 installations. This 90-minute session will provide an overview of trixbox 2.0 and discuss what is included, system requirements, hardware and ITSP support, skills required for successful deployment, installation and configuration steps and additional resources available from trixbox.org.

“The new features in trixbox 2.0 make it the application standard for telephony-savvy businesses and integrators deploying open source IP telephony systems,” said Garrison. “The upcoming Webinar will provide the most up-to-date and useful information about how to successfully implement trixbox 2.0 installations.”

For more information, or to sign up for the trixbox Webinar or download trixbox 2.0 for free, visit www.trixbox.org.
About trixbox
trixbox, formerly Asterisk@Home, is the world’s largest community of users of open source, Asterisk-based voice over IP telephony platforms. The value of trixbox is that, in under 15 minutes , a non-technical user can download and install, not only Asterisk, but Linux, SugarCRM, MySQL, FreePBX and other applications. trixbox tightly integrates these open applications to work together on one physical server, providing companies with a PBX phone system and the surrounding applications they need to support their business. The trixbox community has the largest number of registered users and the most active forums for discussing and resolving open source telephony and Asterisk-based issues.

About Fonality
Fonality, www.fonality.com, is a leader in open source, Asterisk-based IP telephony systems. PBXtra™, Fonality’s award winning IP-PBX product line for small and medium businesses, is based on a modified version of the popular open source Asterisk code base. Fonality has modified Asterisk to add reliability, stability and enterprise-class features, and is the world’s largest commercial Asterisk-based deployment. PBXtra uses Fonality's patent pending architecture to deliver all the advanced features of an enterprise-class phone system and call center at 40 percent to 80 percent less than traditional offerings, and is deployed to more than 18,000 business users in the U.S. and other countries. Fonality’s trixbox, www.trixbox.org, the world’s largest community of users of Asterisk-based software, with an average of 50,000 downloads a month, provides a free, easy to install Asterisk-based VoIP phone system.
###

Fonality and trixbox are registered trademarks and PBXtra is a trademark of Fonality. Asterisk is a registered trademark of Digium, Inc. Fonality and PBXtra are not affiliated with, nor endorsed by, Digium Inc.

For Downloads, on the Sourceforge site link given below select the 2.0 branch and correct image you need, CD(iso) or VMWare image.
Links;
The trixbox 2.0
The trixbox 2.0 VMWare Image

GigaOM wants your home phone, for internet services

Allan Leinwand of GigaOM fame has an interesting post about more interesting idea. He wants to bring internet to your home phone, your wireless handset as you roam around your home.
Imagine checking price of medium pizza in your neighborhood and placing an order without having to touch that big old yellow pages. Well that is a part of his idea. You can read the full length article at the link given below.
I only worry about mixing up my mobile phone and wireless home phone. If the technology provides Allan's solution, both phones will be similar in appearance and will be doing similar work. I might switch the phones and have the home phone in my pocket when I go out.
Ok here is my idea, give me another Dual Phone, not the kind we know as dual phone so far. I want a mobile phone, that turns in to a wireless extension to my home phone when I come home!

Links;
Allan's article on GigaOM
Dual phone with yahoo inside


Wednesday, January 03, 2007

Vittles strikes again with U-Rang II


Within five days of releasing U-Rang, Nerd Vittles has released an update, upgrade to the product, U-Rang II.
After all those articles helping us with Asterisk and other numerous VoIP articles, (Not to mentions ideas for filling up our blogs! ;)) Nerd Vittles seem to be spewing out VoIP applications now.
When you use the application, when someone calls you, phone slip (Picture above) will pop up on your desktop with the time and date of the call as well as CallerIDname and number information of the caller. And, if you’re using the Nerd Vittles CallerID enhancements, then your Asterisk system will check the Google Phonebook, AnyWho, and AsteriDex for supplemental CallerID information in addition to what’s provided by your local phone company.
NV CllerID enhancements could be installed using PBX-in-a-Flash script, also provided by Nerd Vittles for Trixbox 1.2.3.

The U-Rang II could be downloaded from the link given below.

Links;
Nerd Vittles U-Rang II article
Download U-rang II


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