VoIP Information, VoIP News and VoIP Technology

--> Area Code

iotum helps makes Asterisk relevant to more Enterprise Users

For the more than 250,000 Asterisk IP-PBX installations around-the-world, getting the right phone call, at the right time, and on the right device just got easier thanks to iotum. Today, at VON Canada in Toronto, iotum announced the beta availability of the iotum Asterisk integration module; two relevance-enabled call management applications for Asterisk and a rich new set of developer API's.

iotum's Asterisk integration module connects its award-winning Web 2.0 call management applications to Asterisk IP-PBX's to help users prioritize which calls are important, and which can wait, based on who's calling, and what the user is currently doing. The iotum Asterisk integration module consists of source code, installation instructions and access to an iotum test account.

"Today's announcement represents the next step in iotum's platform strategy", said CEO Alec Saunders. "As of today, members of the world-wide community of Asterisk developers and integrators can use our platform to build new and compelling relevance-enabled revenue generating applications."

Once the integration is complete, this non-commercial beta will allow Asterisk users to filter, rank, and prioritize incoming calls using iotum as well as offer users the ability to easily schedule conference calls from within Microsoft Outlook using iotum's Pronto Conference Calling feature. With, Pronto Conference Calling participants join a conference call by dialing the organizer's usual phone number eliminating the need for everyone to dial into a conference bridge or remember pass codes.

The iotum Relevance Engine API allows Asterisk developers to embed advanced, contextually driven call management capabilities into their own applications. The API allows services to be built which use the iotum Relevance Engine, the world's first smart platform to intelligently assess the relevance of a phone call, and route it to the most appropriate device on any network.. The API is for use in a variety of different kinds of applications, including collaboration, CRM, and personal productivity applications.

For more information about iotum, visit the company's web site:
http://www.iotum.com

A new twist on VoIP? - A new Jajah killer and serious Skype competition?

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

AstManProxy 1.20pre Released

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

VoIP Caller ID Spoofing - Still Dangerous

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


  Next Entries »