Voice Over Internet VoIP

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

This is a new, easy to use phonebook application that installs in minutes on any Asterisk server. It has a pleasant, ergonomically designed web interface that allows to look up phone numbers and to modify and update the central Asterisk caller database, so that callers can be identified by name on all SIP telephones.

The most recent incoming calls are listed up front with names when avaibable. Phone numbers may be imported from other applications or even retrieved online via LDAP from Exchange od eDirectory. By clicking on the telephone number a call can be initiated without the need for MS-TAPI or any client software. (Click-to-Dial) There is also a mini dialer that can be used for telephone integration with other applications.

The phonebook may be used as a central phonebook for smaller companies, as an add-on for existing company address applications, as a tool to specifically deal with CallerID identification in Asterisk or as a cute click-to-dial application. (CTI)

Click Here for More Information

Yesterday I had the pleasure of chatting with Peter Csathy. Peter's the CEO of SightSpeed. We talked about VoIP, video, and SightSpeed's fascinating software offering. I'd like to share our conversation with you.

The first thing that's important to acknowledge has to do with Peter "drinking his own kool-aid," which I'll mention again later. We used SightSpeed to share a video conversation for this interview. This is the second time I've done an interview using SightSpeed, and in both cases, it's been an awesome collaboration tool.

Click Here for the Interview

Avaya Inc. is looking to tap into Philippines-based application developers to create vertical solutions for some of its new products, including its one-X line of SIP-based IP telephony products.

Having completed its campus-wide wireless network last year, the University of Queensland (UQ) in Brisbane has joined the handful of enterprises deploying the open source Asterisk IP-PBX for staff and student VoIP.

Scott Sinclair from the university’s strategic technologies group told Computerworld new technologies are always being investigated and VoIP could reduce call costs, particularly between the smaller campuses which are already linked by fibre.

“We have a commercial ISP as part of the university so providing commercial VoIP with Asterisk would be good,” Sinclair said. “We’re looking at a number of products but the easy and inexpensive way to get into [VoIP] is with open source.” While making a name for itself among open source and IP telephony circles, Asterisk, which runs on Linux and Unix, has little to show for widespread enterprise adoption. Its flagship end-user sites include Melbourne-based department store chain Adairs, and Copiah-Lincoln Community College in Mississippi.

“So far we have successfully integrated Asterisk with the traditional TDM and are now looking at the presence functionality it provides,” Sinclair said.

“We only have a small deployment but it’s been successful so far. Being able to advertise the multiple places where you are is a powerful feature.” UQ’s Asterisk system consists one x86 server running Red Hat Linux. Sinclair is excited at the possibilities of VoIP for some 5500 staff and 35,000 students when “their e-mail will become their phone number using the SIP protocol”.

About 10 people are using Asterisk now, but UQ will soon begin a pilot project with one of its residential colleges to supply VoIP to students’ rooms. This will involve some 200 users.

Click Here for the Full Article

Nerd Vittles today provides a tutorial on linking a Sprint cell phone to Asterisk@Home to all but eliminate the cost of outbound cell calls nationwide. Last week Sprint announced the availability (beginning today!) of a new add-on for existing and new Sprint cellphone customers. For $5 more a month and a little Yankee ingenuity, you now can make unlimited FREE calls between your Sprint cellphone (or multiple PCS phones if you're on a shared plan) and your residential phone number regardless of the wireline carrier. In short, your home phone service need not be with Sprint. If you have Sprint home phone service, then the new PCS to Home service will be free. In either case, no cellphone minutes will be assessed for inbound or outbound calls between your Sprint cellphone and your home number … ever. In fact, they’ll show up on your statement as PCS-to-PCS calls which are also free.


The Nerd Vittles Recipe:

-Cheapest Sprint Cell Phone Plan - $35
-PCS to Home Add-On Service - $5
-Asterisk@Home Server for Linux or Windows – FREE!
-Home Phone Service Switched to BroadVoice or AxVoice BYOD Plan - $9
-TelaSIP VoIP Unlimited Residential U.S. Calling Plan $15
-Nerd Vittles DISA Script – FREE!
-Unlimited Monthly Calls from Home OR your Sprint Cellphone - PRICELESS ... and FREE!

Click Here for the Full Nerd

LumenVox, an innovator of speech recognition technology, announced that Digium Inc., the creator of Asterisk, and pioneer of open source telephony, is currently integrating LumenVox's Speech Engine into their Open Source and Business Edition PBX's.

"Speech recognition enhances customer interactivity with an Asterisk PBX," said Mark Spencer, president of Digium and creator of Asterisk. "Additionally, the integration with the LumenVox Speech Engine enables the Asterisk development community to cost-effectively build and deploy speech solutions with performance characteristics to support even the most demanding speech requirements."

"One of our missions as a company is to work towards popularizing speech recognition," said Ed Miller, president of LumenVox, "and to provide world-class technology. The Asterisk community is innovative and adaptive and we are pleased to be a part of Digium's open source communications revolution."

The Speech Engine performs recognition on audio data from any audio source, and allows for dynamic language, grammar, audio format, and logging capabilities.

Click Here for more Information