Here is our new location: http://www.asteriskvoipnews.com/

Over the next week I will be moving the archives over and adding a feature two. I hope you enjoy the new look and layout. If you have any questions or comments please use the "Email Us" link at the new url.

Thanks,
-Dal

Hello All,

I just wanted to drop a line to let everyone know that we will be moving the blog to a new platform that will add a bunch of features are give us more control of the design to help deliver all the news in a more developed way. The inital process is almost complete and once I have a few design issues hammered out I will post the new url for everyone so please update those BOOKMARKS(I check the logs :P) :) Here is a list of some of the new features we will be integrating:

-More Defined Categories for Asterisk Specific Topics (This will always be a sizable part of the core for this blog)
-More Categories covering: VoIP, Skype, VoIP Legislation, WiFi/WiMax Deployment and more.
-Archive Calendar Function and Robust Search to help people find archived info
-More Community Written Content and Help Articles
-Some cool plugins I haven't even found but I know is out there
-Maybe even a logo...(you never know?)

Well that's it for now. Once it's ready I will make a post to let everyone know and most likely move all the old archives to the new platform so people can search just one place. If you have any questions just hit the email link on the right and I'll try my best to answer them. Have a great day and I will be in touch soon. Now time for me to get back to editing my new CSS :)

-Dal

Click Here: Announcing Astmanproxy 1.20

Integrics is pleased to announce the release of ITSP version 1.6. This version has the following new features:

- Comes in 2 Editions:

* Carrier edition, for 250 to tens of thousands of users on hosted systems. Integrics sells this edition directly and through partners.

* Office edition, for 10 to 250 users. This edition is sold only through our partners, for them to sell as PBX systems at their customers' sites.

- Post-paid and external application billing, as well as the existing pre-paid billing.

- PDF invoices, including invoice management for both resellers and customers.

- Call shop interface.

- First version that can be installed by our partners rather than by Integrics staff. We'll be launching a formal partner and reseller programme in a few weeks; more details to follow.

- Many customer self sign up wizard improvements, such as credit card capture, click through terms and conditions which can be set by resellers, etc.

- Lots of smaller improvements based on customer feedback.

The demo system at:

Demo System

Running 1.6. Further information, including a feature list, is at:

http://integrics.com/products/itsp/

--
Alistair Cunningham,
Integrics Ltd,
+44 20 799 39 799
sip:acunningham@integrics.com

Click Here: Why VoIP Needs Crypto?

Voice over Internet Protocol (VoIP) telephony Relevant Products/Services from , once a tool used primarily by uber techies, has matured into a viable and less-expensive alternative to the PBX systems used by businesses of all sizes. With VoIP, companies have the opportunity to discard the prepackaged offerings of traditional telecommunications and instead opt for a phone system that is customizable and highly adaptable.

According to a report released in January from the research firm Yankee Group, the VoIP market is expected to reach $3.3 billion in service revenue by 2010. The report said that businesses are favoring VoIP because the technology offers measurable savings, an excellent converged platform for voice and data, and improved ways to manage communications within the enterprise.

"Virtually all major carriers, systems integrators, and equipment vendors now offer different varieties of business VoIP services," said Taher Bouzayen, a senior analyst for telecommunication strategies at the Yankee Group. "And those that are not already exploring ways to capitalize on this revenue opportunity need to start now."

Even the U.S. military is contemplating a move to VoIP. Avaya, a VoIP vendor, recently announced that the Air Force is testing the technology with an eye toward deploying it for military communications in the field. According to Avaya client executive Vic Galante, the Air Force realized that it would have to turn to commercial technologies to save money.

Click Here for the Full Article

Talk about mind-boggling changes. A new project will allow businesses to connect GoogleTalk users to their Asterisk, telephony servers, Mark Spencer told me yesterday. He should know. Spencer is the author of the Asterisk open source, IP PBX and CEO of Digium, the company packaging Asterisk as a business-grade solution

The project is currently in beta with availability set for some time around June, smack in between the May release of Digium's Business edition of Asterisk B.1 and the next rev of the public Asterisk 1.4 release in July.

I'm so incredibly excited about this project for lots and lots of reasons. Inter-company collaboration will become a lot easier now for Asterisk users. Extending out the Asterisk network is a cinch and think of all of the cool applications one could create for GoogleTalk. Heck, just real termination becomes easier.

But I get ahead of myself.

One of the things that's always puzzled me is why Google didn't just support SIP or even Mark's own IAX protocol used in Asterisk. Mark had the same question until he studied Google's SIP replacement, Jingle

Click Here for the Full Article

Click Here for Uplink Connects SIP & Skype

Click Here: iotum helps makes Asterisk relevant to more Enterprise Users

Since early 2006 movement has come into the VoIP industry. New VoIP providers are now launching all over the world with each one of them hoping and expecting a share of the ever growing popularity and income stream. At the last count the research company www.myvoipprovider.com had almost 650 VoIP phone providers listed.

This has had one distinct advantage for the consumer - VoIP costs internationally are dropping at an alarming rate. A few VoIP providers in Europe have taken their marketing activities to the extreme by offering free calls to a wide range of up to 50 international destinations. VoipBuster was the pioneer early 2005 and has since then launched a barrage of sister companies offering exactly the same type of service. Time will tell if this "Free VoIP" campaign has any long term merit.

Even in this highly competitive enviroment some companies still manage to stand out of the masses. In mid March 2006 two companies launched, in one case, relaunced their services. Lycos decided that it is time to join the race with the likes of Yahoo and possibly in the very near future Google and MSN. Using Globe7's technology Lycos launced an interesting softphone with a free US phone number, 100 free minutes and an integrated mp3 player and video.

Click Here for the Full Article

Hi Folks,

I have done a bunch of new work on AstManProxy, including 1) adding in the Action: Challenge authentication mechanism (basically done), 2) adding in support for SSL, 3) added patch from Steve Davies that will do basic user authentication.

The SSL support is based on code from John Todd at Tello (digium #6812). I do not have the SSL stuff functional yet but it is compiling and I am hooked into their underlying routines. I just need to figure out how to hook all the functions in now, which I'll be working on over the next few days.

Also, I have setup a proper branches/tags/trunk structure on svncommunity.digium.com (thanks to Kevin Fleming for his help). 1.13 is currently in tags & trunk, and branches contains the 1.20 development work.

Anyway, please take a look at the 1.20pre branch and let me know what you think:

Click Here for Download of 1.20pre

Also I have turned on moderation on the astmanproxy list, so we should not have any more spammers... sorry about that.

Best,
Dave

Many in the VoIP service industry have known for years that caller ID can be spoofed (that is, misrepresented) relatively easily. In fact, one need not be an expert at using Asterix's Linux PBX software or know the other tricks of the trade - he can simply pay a few dollars for an Internet telephone caller ID spoofing service. (We're not going to provide free advertising for these services here) While this may seem harmless, it opens up the door to a number of serious vulnerabilities.

More and more caller ID is being used to authenticate people's identity. Credit card companies have long been using caller ID in the card activation process. Financial institutions such as Citibank and American Express are now using it to authenticate identity of account holders who dial in to their telephone service. In business, caller ID is used to signal whether a caller is calling from inside or outside the firm. 911 call centers use it to determine who is calling and where to send emergency responders. Voicemail systems, particularly cell phone voicemail systems, automatically playback messages based on caller ID.

Click Here for the Full Article

The FreeSwitch project announces the immediate availability of a brand new Open Source Jingle XMPP signaling library as well as an endpoint module enabling a Jingle telephony gateway. The library dubbed "libDingaLing", written in C, creates a layer of abstraction to allow for an easier transition as the Jingle protocol evolves and eliminates the need to deal with XMPP or XML and supports many concurrent instances within 1 application.

The library is currently considered to be in Alpha stage, has been compiled and tested on many computer platforms including Windows XP, Solaris, Linux and MacOS X. The only other existing implementation of this protocol released thus far is the GoogleTalk instant messenger application therefore the library has been designed with interoperability with this particular client in mind but also anticipates changes in the protocol to come along as it becomes more widely accepted.

The new endpoint module appropriately named "mod_dingaling" couples FreeSWITCH to libDingaLing and allows both inbound and outbound communication. With this technology, GoogleTalk calls can gateway to the PSTN or to other VoIP protocols such as SIP or H323.

FreeSWITCH, http://www.freeswitch.org is a new open source telephony project started in early 2006 designed to provide a modular platform on which to merge various technologies. Both libDingaLing and FreeSWITCH were written by Anthony Minessale II, a developer who after contributing to other telephony related open source projects, decided to start a new initiative that focuses on abstraction, modularity and cross-platform crossarchitectural design.

Arising Group, Inc. has begun rigorous testing of the open source VoIP PBX, Asterisk. It is hoped that the new communication platform will provide a robust and low cost telephone solution for their international clients who need to stay connected with their traveling field representatives.

George Karshner, Director of Business Development at Arising Group said that what attracted them to Asterisk is the versatility and functionality of the program. "It appears that Asterisk can perform all of the functions of a premium office phone system like voicemail, conference bridging, call queuing, and call detail records, plus higher functions usually required by trading firms, like talk detection, call monitoring, and remote call pickup" Karshner said. "Its flexible feature set is very promising", he added, “but we want to run a battery of situation tests before we deploy it".

Arising Group has several multinational clients who need to stay in touch with their workers in Europe and Asia. One China based client has reps in the U.S., Japan, Korea, Vietnam, Hong Kong, Taiwan, Singapore, Malaysia and Indonesia. All of these people must communicate frequently from places where Internet service is more reliable than cell phone connections. They are also looking to cut down their international calling costs. With this new VOIP technology, all the inter-office phone bills will be one low, flat rate, Internet connection call.

Click Here for the Full Release

VoXaLot is the Web activated telephony service "Web Callback". Using this functionality you can make a call from any phone, anywhere, anytime using VoIP rates - even if you don't have an ATA or VoIP phone.

So, how does it work? You need to have signed up with a VoIP provider that gives you call rates that you are happy with. You don't have to configure any equipment on your side - you just need to have an account with a third-party VoIP provider.

In addition to being able to call VoIP numbers without having any VoIP equipment, you can also take advantage of the cheap PSTN rates that many providers offer. To do this, you need to have accounts with two different providers.

Click Here for more Information

With apologies to Sidney Poitier, yes, even your doorbell can now be part of your Asterisk system. And it probably should. Kevin Flanagan and his wife run a ski lodge in Mt. Washington Valley, New Hampshire. For baseball fans, you'll be interested in knowing that Babe Ruth spent a lot of time hanging out in Room #2 at the Cranmore Mountain Lodge primarily because his daughter owned it in the 1940's.

Anyway, Kevin wrote us about his DOORBELL several months ago, and we've been chomping at the bit to publish his article but were just waiting for a lull in the Asterisk updates. I hate to even say that for fear that Asterisk@Home 2.8 will hit the street in the morning. So, today, we're going to show you how to hook up your doorbell to Asterisk. And, we'll throw in an intercom as well. When someone rings your doorbell, they'll get music on hold or a prerecorded announcement while your phones go crazy!

Click Here for the Full Nerd

Companies can take comfort in knowing its voice communications are secure by implementing AES128 bit encryption for its Asterisk service. This application requires IAX/2 and Asterisk 1.2.4 or above. Sign up for free and purchase this service. Veratel's AES128 bit Encryption for IAX is only $5.00/US per account, for an unlimited number of local/toll free DID numbers.

Click Here for more Information

The Linux LiveCD VoIP Server can be used to provide a Vonage type service, or to create a voip pbx for a campus or business with upto thousands of phones. It is based on the Open Standard SIP Express Router (SER) and Asterisk. It can serve as a SIP Proxy, VoIP PBX, VoIP gateway or Class 5 Softswitch

Live Demo Examples: FonoSIP.com and VoIP.brujula.net

Click Here for more Information

Hello Asterisk Users,

I am an Objective-C enthusiast and have been writing some clever tools to integrate Asterisk functionality with Mac OS X applications.

Please find my project on:
http://www.sf.net/projects/astrxtools4osx/

The objectives of my project are as follows:

1. Implement an Objective-C framework to communicate effectively with the Asterisk Management Interface

2. Address Book plugin to enable call back functionality

3. A System Preferences pane to allow administrators to easily configure Asterisk options on a Mac

4. Dashboard Widget that allows users to quickly call arbitary numbers

5. iTunes integration to stop and star iTunes to play when the phone rings etc.

The source code is in pre-Alpha stage at the moment but I am hoping to release a Beta at the end of next week. Please feel free to download and use these extensions. I hope they turn out to be useful and would appreciate any feedback.

Devraj