Asterisk VoIP News

New URL: AsteriskVoIPNews.com - Update Your Book Marks

2006-04-17T10:52:00.000-07:00

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

Asterisk VoIP News Update: We are Moving to a New Platform :)

2006-04-15T11:26:00.000-07:00

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

CANADA VoIP 911 Update - Executive Summary

2006-04-13T12:29:00.000-07:00

Click Here:
CANADA VoIP 911 Update - Executive Summary - has moved - Click Here

[Nerd Vittles] 100 Great Springtime Projects For You & Your Free Asterisk@Home PBX

2006-04-12T11:34:00.000-07:00

Click Here:
[Nerd Vittles] 100 Great Springtime Projects For You & Your Free Asterisk@Home PBX - has moved - Click Here

Skype Buys VoIP Startup Sonorit

2006-04-12T11:01:00.000-07:00

Click Here:
Skype Buys VoIP Startup Sonorit - has moved - Click Here

AstriCon Update: Europe Early Bird Ends Saturday

2006-04-11T22:50:00.000-07:00

Click Here:
AstriCon Update: Europe Early Bird Ends Saturday - has moved - Click Here

Trial Version of Asterisk Interface Available

2006-04-11T10:19:00.000-07:00

Click Here:
Trial Version of Asterisk Interface Available - has moved - Click Here

Spring is here :: Test the Asterisk SpringCollection 2006!

2006-04-10T09:48:00.000-07:00

Click Here:
Spring is here :: Test the Asterisk SpringCollection 2006! - has moved - Click Here

Announcing Astmanproxy 1.20

2006-04-07T20:57:00.000-07:00

Click Here: Announcing Astmanproxy 1.20

Integrics ITSP 1.6 released

2006-04-07T10:54:00.000-07:00

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

Why VoIP Needs Crypto

2006-04-06T11:09:00.000-07:00

Click Here: Why VoIP Needs Crypto?

From PBX to VoIP: Making the Change

2006-04-06T11:07:00.000-07:00

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

GoogleTalk Gets a Facelift for Business with Asterisk

2006-04-05T10:49:00.000-07:00

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

Uplink Connects SIP & Skype

2006-04-04T10:34:00.000-07:00

Click Here for Uplink Connects SIP & Skype

iotum helps makes Asterisk relevant to more Enterprise Users

2006-04-04T10:23:00.000-07:00

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

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

2006-04-03T07:50:00.000-07:00

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

2006-04-02T15:38:00.000-07:00

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

2006-04-01T17:17:00.000-08:00

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

Connect GoogleTalk to your Telephone with FreeSwitch

2006-03-31T12:53:00.000-08:00

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.

Arisng Group Evaluates Asterisk for International Clients

2006-03-31T09:01:00.000-08:00

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 releases web activated telephony service

2006-03-30T11:35:00.000-08:00

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

Don't Guess Who's Coming to Dinner: Use Asterisk

2006-03-30T00:11:00.000-08:00

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

Veratel offers AES128 bit Encryption for IAX

2006-03-29T10:36:00.000-08:00

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

Linux LiveCD VoIP Server

2006-03-29T10:32:00.000-08:00

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

Asterisk Tools for Mac OSX

2006-03-29T10:19:00.000-08:00

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

Asterisk Architectural Freeze for 1.4

2006-03-28T10:27:00.000-08:00

I want to remind everybody that we have our scheduled architectural freeze, this Friday, for the 1.4 release. What this means is that if you have features that you would like to see in 1.4 THAT REQUIRE CHANGES TO THE HEADER FILES, those changes need to be finalized by this Friday. We may push back the date by a couple days, if we have enough on the bugtracker to discuss, but your patches on the bugtracker must be applicable to the current trunk, and you must have addressed all concerns listed on the bugtracker before Friday for your patch to be considered for 1.4.

Features that do not require changes to the header files are not architectural in nature; those features have until the beginning of May to be gotten ready.

I hope to see a flurry of activity on the bugtracker, so we may go forward with each of our planned freeze dates, culminating in the release of 1.4 by the end of June.

--
Tilghman

WIST - Web Interface for SIP Trace

2006-03-27T15:11:00.000-08:00

Click Here for WIST - Web Interface for SIP Trace

Asterisk 1.2.6 and Zaptel 1.2.5 Released

2006-03-27T12:05:00.000-08:00

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.6 and Zaptel 1.2.5. Both of these releases include a number of important bug fixes, and users are encouraged to upgrade their systems when possible. See the included ChangeLog files for more details on what has been fixed.

The releases are available on the Digium FTP servers as PGP signed tarballs and also as PGP signed patch files, to ease upgrading from the previous versions. The keys used to sign these files can be verified by using the keyserver at pgp.mit.edu.

Thanks for your support of Asterisk and Zaptel!

x100p.com Announces World's First Low Profile FXO PCI Card

2006-03-27T08:26:00.000-08:00

The X100P provides a single, full featured FXO interface for connecting the Open Source Asterisk PBX server to the PSTN (Public Switched Telephone Network).

This PCI card allows Asterisk to make calls to or receive calls from a traditional analog phone line. The X100P is affordable and ideal component for building Interactive Voice Response (IVR) and Voicemail applications.

It supports all standard enhanced call features including Caller ID, Call Conferencing, and Call Waiting/Caller ID. It features the Latest Revision of the Original DAA chipsets with Caller ID Fix.

By combining the X100P and the Power of Open Source Asterisk PBX, one can easily, economically implement sophisticated yet flexible call services. Such services ranging from multi-menued IVR, multi-protocol VoIP gateways, directory services to business class voicemail.

Click Here for more Information

TERENA Secretariat Switches to VoIP/Asterisk

2006-03-27T08:25:00.000-08:00

The TERENA Secretariat recently migrated its phone system to an open-source-based Voice over Internet Protocol (VoIP) solution. VoIP is a technology for transmitting ordinary telephone calls over the Internet very cheaply or for free. TERENA had planned to replace the Secretariat phone system this year and wanted a system that is more reliable than the traditional systems and provides a number of new features.

VoIP Technology is just one of the areas that the TERENA Task Force on Voice, Video and Collaboration (TF-VVC) is involved with. TF-VVC promotes the ongoing development and testing of available collaboration technologies and services and defines, develops and tests new video, voice and collaboration services. It was felt that any new TERENA Secretariat phone system should support the new technologies being pioneered by the task force. Several options were considered and a decision was taken to opt for an open solution which was free from any vendor lock-ins and proprietary protocols.

TERENA decided to use the Open Source software package "Asterisk" as the heart of the new Private Branch Exchange (PBX). All legacy phones in the office were replaced with IP phones, which all use the Session Initiation Protocol (SIP) to communicate with the PBX. The PBX itself still uses ISDN lines to make calls to people on ordinary phone networks but it is also hooked up to the Internet, so callers can use SIP to make (free) calls via the Internet.

Click Here for the Full Article

GEOTEK Phonebook for Asterisk Released

2006-03-25T13:16:00.000-08:00

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

Realtime Interview with Peter Csathy at SightSpeed

2006-03-24T10:43:00.000-08:00

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

University of Queensland (UQ) dials Asterisk for VoIP

2006-03-23T13:23:00.000-08:00

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

Even more cool README for the test branch

2006-03-22T18:37:00.000-08:00

Thanks to Mike Taht - the Asterisk Rock Band Leader - we now have a README with hyperlinks to the bug reports in Mantis.

More Information

A good way to discover the test branch before you help me test it.

Today, I've updated the func_realtime so it loads properly (thanks bweschke) and added an update to the cdr_radius driver. The radius CDR driver is now a mix of two drivers - the best of two tested drivers where the developers decided to combine their work. Good work in a community fashion!

Thanks for all the help and all the contributions. Keep testing!

/Olle

A Marriage Made in Heaven: Sprint Cellphone + Asterisk@Home = Unlimited U.S. Cell Phone Calls for $5

2006-03-22T07:23:00.000-08:00

Click Here for Sprint Cellphone + Asterisk@Home = Unlimited U.S. Cell Phone Calls for $5

LumenVox and Digium Partner to Offer Speech-Enabled for Asterisk Business Edition

2006-03-22T07:18:00.000-08:00

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

VoIP offers wealth of opportunity

2006-03-21T08:56:00.000-08:00

By the time you read this column, Cebit 2006 will be over and the exhibitors will have flown home. But judging from the number of products and vendors demonstrating their offerings it is clear that voice over IP (VoIP) has gone mainstream.

On the one hand the Chinese and Taiwanese manufacturers are trying to bring new products to market, and on the other we have some seriously large players extolling the virtues of their particular brands of VoIP.

The legacy PABX vendors simply cannot succeed in the face of competition from VoIP and Session Initiation Protocol (SIP)-based telephony. While the PABX market may continue for a few years as a service-only operation maintaining legacy equipment, anyone who buys a non-IP telephony system (particularly smaller firms) is probably wasting their money. This is particularly the case where costs can be reduced significantly through the use of open-source PABX systems, such as Asterisk.

Click Here for the Full Article

GDS Voice Conferencing Solution released today!

2006-03-21T08:45:00.000-08:00

Nearly every company today uses Voice Conferencing in daily business to maximize productivity. There are many conferencing solutions in the market but the challenge is to integrate a scalable solution that will meet your needs and also delivers cost-efficiency and good return on investment.

GDS Voice Conferencing solution is a cost effective, feature rich Enterprise Voice Conferencing solution based on native Asterisk voice conferencing application.
GDS Voice Conferencing solution is an feature rich Enterprise Voice Conferencing solution built on top of native Asterisk MeetMe application.
For more info visit:
GDS Voice Conferencing Solution Info

Overview
* Multiple conference types (scheduled, recurrence, reservation-less)
* Intuitive web interface for conference management, personal contact management, user management and system administration
* Ability to easily manage conferencing attributes like announce user leave/join, wait for marked user and to associate contacts and its roles within the conference (listen only, admin mode etc.)
* Monitor live conferences (mute/un-mute participant, kick out participant, lock conference, view on line participants, its attributes etc.)
* Easy import of existing contacts
* Integrated personal contact management for simple invitation and notification
* User role based privileges
* Port resources management (TDM and VoIP)
* Recurrence and conflict conferences management
* Automatic email notifications and reminders
* API (application programming interface) that allow development of integrated or custom application
and more...

Boris Zolotarev
boris.zolotarev@gdspartners.com

China to open up VoIP market?

2006-03-20T19:19:00.000-08:00

Speculation is growing that China could be about to relax restrictions on its voice over IP (VoIP) market.

A report last week in The Beijing News said the Chinese government has granted a VoIP licence to a southern Chinese telecoms company for a pilot programme, and telecom carriers and virtual network operators (VNO) will be allowed to apply for the licences starting in 2007.

So far Chinese telecoms operators that have received government approval to trial VoIP services declined to do so because they believed it would threaten their fixed-line services revenues, according to the newspaper.

Click Here for the Full Article

FreePBX 2.0 Preview

2006-03-20T19:16:00.000-08:00

Anyone who has spent more than a few minutes trying to figure out Asterisk’s configuration files can quickly appreciate a graphical user interface to make managing the myriad of files much easier. One of the best open source projects has been the Asterisk Management Portal (aka AMP). While AMP has been a fantastic tool, its original design did not take into account all of the features that would eventually become available and need to be added to the system. Eventually, a complete rewrite of the code was going to be needed to modularize the system and change things around in order to sustain the product for a long time. Thus, FreePBX was born. In this article, we look under the hood at what FreePBX is and how it works.

What is FreePBX
Many people think that FreePBX is a competitor to Asterisk@Home, this is far from accurate. Asterisk@Home is an ISO image that automatically installs CentOS, AMP, Flash Operator Panel, Asterisk Recording Interface, and many other tools and preconfigured dialplan options. FreePBX is simply the replacement for one component of Asterisk@Home, replacing AMP as the configuration file editor. In upcoming versions of Asterisk@Home, it will include FreePBX rather than AMP. If you build your own Linux server, install Asterisk, you can simply install FreePBX to help you manage your system.

Click Here for the Full Preview

New astGUIclient VICIDIAL Released: 1.1.10

2006-03-17T19:28:00.000-08:00

We've released another update to our Asterisk GUI Client suite: 1.1.10

http://astguiclient.sf.net/

The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app inbound/outbound call center software. This package is free as in GPL. (The suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks.

For this revision, we have focused on fixing bugs and several new features like Answering machine detection integration and Scheduled callbacks for VICIDIAL. We have also tested the suite on Asterisk versions through 1.2.4

All client web-apps and administration pages are available in English, Spanish and Greek, with rough translations of French, German, Italian and Portuguese for the client web-apps only.

Check out the project blog for more information:
http://astguiclient.blogspot.com

Let me know what you think.

Thanks,
-MATT

FreePBX 2.0.1 Released!

2006-03-17T19:11:00.000-08:00

The Asterisk Management Portal (AMP) is now known as FreePBX.

FreePBX 2.0.1 is now available for download. A **BIG** thank you goes out to the project developers for all their hard work, and to beta testers for running FreePBX through it's paces!

This exciting new release boasts a better user experience, additional functionality, and a new module system.

The module system is designed to be simple, powerful, and easy to use existing code with. It only imposes a minimal API, with just a few requirements to make it work. Please see the project wiki for more information: http://freepbx.org

As usual, previous versions (AMP) will be automatically upgraded by the install_amp script. Please note, that when you install, no modules will be enabled. You must visit the web interface and click to install them.

Please report any bugs using the sourceforge bugtracker.

Bug Trackers & Mailing Lists: http://sourceforge.net/projects/amportal
Project Wiki: http://www.freepbx.org
Documentation Wiki: http://docs.freepbx.org
IRC: #freepbx on freenode

Changes For: 2.0.1
- AMP is now "freePBX"
- New module system allows for drop-in functionality
- All previous AMP functionality ported to new module system
- Added Modules: Conferences, Time Conditions, Asterisk CLI, Online Support
- Inbound routes can set ALERT_INFO variable for SIP devices
- Outbound Routes can now use an Authenticate Password File
- Queue Static Agents can have penalties applied
- Ringgroups can play an announcement to caller before dialing
- Ability to force Emergency Caller ID for devices using an Emergency Outbound Route.
- Much improved form validation for all modules
- Requires Asterisk 1.2.x
- Using native music on hold support - no more mpg123!!
- FOP .24
- ARI 00.08.03 - now with AJAX!
- Initial sqlite support!
- GUI improvements
- Default is to use freePBX database authentication.
- SVN has been adopted for version control

--
Ryan Courtnage
Coalescent Systems Inc.
Tomorrow's Telephony Today
403.244.8089
www.coalescentsystems.ca
www.gabcast.com

Calling Circles Desktop - Outlook/Asterisk Integration

2006-03-17T09:04:00.000-08:00

Click Here for Calling Circles Desktop - Outlook/Asterisk Integration

Asterisk Users to Get Free Call Management App from Fonality

2006-03-16T12:42:00.000-08:00

Excerpt: IP-PBX phone systems provider Fonality announced this week the introduction of Heads-Up Display (HUD), a new call management application that provides businesses with real-time, easy-to-use call control and management features. HUD comes in two versions, the HUDlite, which is a free call management application for the Asterisk Open Source PBX; and the HUDpro, which is an advanced call management application that enhances PBXtra, the company's IP-PBX platform.

"The Digium-Fonality relationship is an important one to us," said Spencer. "Fonality's new HUD application provides Asterisk users with an innovative and extremely productive way to improve their operations with call presence awareness and call management."

The HUD seems to be like a presence monitor. Through a color-coded desktop interface, HUD lets employees see when others in the office are on a call, to whom they are talking to and whether calls are internal, external or in a queue. HUDlite, available next month, provides drag-and-drop calling and call controls, call monitoring and barging and on-the-fly recording. HUDpro is currently available and it provides additional features, including advanced multihierarchical permission systems, enterprise-class secure instant messaging (IM), complete integration with PBXtra and configuration and support from Fonality."

Click Here for the Full Article

Asterisk Predictive Dialer for Your Outbound Call Center

2006-03-16T07:58:00.000-08:00

Predictive Dialers are used by outbound call centers to keep their call center agents talking on the phone. Indosoft has recently upgraded its predictive dialer technology based on open source Asterisk.

It can provide live connects within 10 to 15 seconds from the time an agent wraps up a call. In order to achieve this, a predictive dialer has to dial more phone numbers than the anticipated number of agents available to answer the calls, should the calls be picked up. Generally when a call is picked up and there are no available agents to answer the call, it gets dropped and the person on the other end does not hear an agent. The FCC in United States and the CRTC in Canada have strict guidelines governing dropped calls. Indosoft's predictive dialer technology is designed to be compliant with these guidelines so that the predictive dialer drops less than 3% of the total number of calls connected, excluding answering machines. The PBX running the predictive dialer is expected to play a recorded message announcing the dropped call with details in conformity with the regulation.

Predictive dialers are computer algorithms that decide how many phone numbers the PBX should dial out, for a given number of agents. The optimization in the predictive dialing algorithm tries to determine the number of connects at any given time. The parameters are generally a function of the quality of leads, the time of day and the immediate statistical past. Predictive Dialers with tone detection to identify Busy, No-Answer and other call terminations do not have high degree of accuracy in identifying the call termination. Asterisk provides TDM and VoIP termination options that come with Digital signaling essential for reliable and fast detection. Asterisk has a CTI capable TCP based Manager Interface. A good session manager for any call center software should use this interface to manage the predictive dialing algorithm.

This advanced predictive dialer is tightly integrated with Asterisk and is a sub-component of the telephony and CRM of Indosoft's outbound contact center technology. Asterisk PBX is a full featured open source Enterprise PBX software that dramatically reduces the cost of building any large call center using its wonderful telephony platform. At very little cost, it provides all essential functionality required for an enterprise call center. The Digium Quad PRI (T1) boards are good quality TDM interface at extremely low cost. Digium-Asterisk will become the dominating platform for contact center industry in years to come.

Indosoft Inc. has many successful deployments of its predictive dialer in contact center industry today. Indosoft is a Digium Asterisk partner and provides fully blended solutions for Contact Centers, Audio Conferencing Bridge, Real-time call blocking for Do Not Call list enforcement, Hosted PBX, IVR and Recording.

AudioCodes Teams with Digium -- Media Gateways + Asterisk Software

2006-03-15T15:17:00.000-08:00

AudioCodes and Digium, the creator of Asterisk and pioneer of open source telephony, announced a partnership to formalize product interoperability between a range of AudioCodes media gateway platforms and the Asterisk open-source software application.

AudioCodes SIP Gateway products, the Mediant 1000; TP260/SIP; and the MediaPack will undergo testing and a certification process to determine interoperability and compatibility when integrated with Digium's Asterisk Business Edition.

The companies said the results of interoperability testing will help Asterisk developers and value added resellers (VARs) to better design and deploy high quality and scalable SIP-based solutions.

More Information:
http://www.audiocodes.com
http://www.digium.com

Take Your Free VoIP Test Today!!!

2006-03-15T08:24:00.000-08:00

Goto: http://www.start-voip.info/
and take your VoIP test. You have 10 mins to take the test. Anyone who passes and emails there name, screenshot of time and proof they passed. I will add you too my own Grad list.

AVN Blog Graduated List:

Asterisk User/Provider Database

2006-03-15T08:06:00.000-08:00

The title explains it all. Goto the site and adds yourself to be counted:

AsteriskCounter

Note: I have thought about developing one of these of my own for the blog. If there is an devs out there that would like to work with my on a version of this but with a different feature set please email me.

MCC Billing Solution for Asterisk v.1.3 Released

2006-03-15T07:41:00.000-08:00

MCC - Billing solution for Asterisk PBX

Current version: 1.3 + 1.3.1 Patch

MCC is a web-based, user (and admin) friendly billing interface for Asterisk and VOIP.

MCC is open source software licensed under the GPL

Features of MCC:

-Unlimited SIP, IAX and Mobile/PSTN devices assigned to user
-Unlimited tariffs with different rates
-Rate Table viewable in Currency of choice
-Profit counting!!!
-Stats by countries
-Blocking of users
-Show Balance, Expenditure, Payments and number of Calls on each account
-Call Data Records (also in CSV/PDF)
-Advanced customer management and portal management
-Integrated PayPal and Hanza.net commerce modules
-View and Store Customers payments
-Manage Pre Paid and Post Paid customers
-Full Credit control by User Account

Concurrent calls for every user

MCC Requirements:

Asterisk
PostgreSQL
Apache + PHP

Homepage: http://www.paskambink.lt/mcc

SipReality Announces Release of TotallySip Softswitch

2006-03-14T21:51:00.000-08:00

SipReality Limited, a developer and distributor of Voice Over Internet Protocol (VoIP) softswitch and end user device provisioning systems is pleased to announce the Commercial Release of it's TotallySip Softswitch platform.

TotallySip has been developed as a carrier grade softswitch solution to service both the emerging VoIP Internet Service Providers (VISP) and much larger and geographically dispersed Competitive Local Exchange Carriers (CLEC) as well the Incumbent Carriers (ILEC). The system is fully scalable from a single node operating with end user provisioning to Class IV and Class V switch features. TotallySip integrates easily with many vendor gateway solutions allowing for maximum flexibility in using existing infrastructure to augment services into the VoIP arena.

Low cost of entry combined with flexible integration and of course extensive features makes TotallySip stand out as one of the most economically viable Softswitches on the market today. Clustering between nodes in geographically disperse or centralized locations is simple and efficient while still allowing for single point management of the entire system. Least Cost Routing (LCR) is done via NPA/NXX with OCN based support coming soon. (LERG subscription required for OCN based routing).

Paul Falcon, President of SipReality stated, "We are very happy to reach our goal of delivering a high availability yet reasonably priced softswitch solution to the market. Today we hit a milestone. Expect to see more modules and add-ons for TotallySip to be available in the coming months extending the ease of use and functionality far beyond any competitive product."

Click Here for the Full Release

Digium Announces New Hardware Products at VON

2006-03-14T16:55:00.000-08:00

Digium Inc., the creator of Asterisk(TM), and pioneer of open source telephony, today announced the availability of new hardware solutions to enhance Asterisk transcoding and echo cancellation performance for VoIP and PSTN gateways. These new products include the TC400P VoIP transcoding card and the TE420P and TE415P four-port T1/E1/J1/PRI cards with onboard hardware echo cancellation.

"Our product team is always working to develop solutions like these that ultimately further the open source movement in VoIP," said Mark Spencer, president of Digium. "Not only are we constantly striving to improve Asterisk's performance, but we also want to contribute to the overall VoIP experience, while keeping costs low."

Click Here for the Full Release

New ncurses Asterisk Manager Interface

2006-03-14T16:44:00.000-08:00

I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at: http://sig.lange.googlepages.com/assman

The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been running it for quite some time on a production server w/o any major glitches. Soon as the sf.net approves the project I will have SVN and the latest versions online.

Feedback is welcome as well as requested features.

Thanks.

--
Sig Lange
http://www.signuts.net/

A2Billing (Asterisk2Billing) Release v1.1

2006-03-14T16:21:00.000-08:00

Great day for the callingcard-fan ! Just a little mail to let you know that a new version of A2Billing 1.1 (Asterisk2Billing) is available! Many features have been added, lot of bugs solved and hundreds of good improvement made, so there we go -> http://www.asterisk2billing.org

The key newest features :
* Ecommerce product with API addons - Integration with OsCommerce
* Speeddial-support for UIs (Customer & Admin)
* Add DB backup/restore tool
* Currencies support management - yahoo financial (cront for auto update) Add new model for update currencies from Yahoo , now currencies are in Database in cc_currencies table. Remove rates.inc and any information about.
* Signup autocreates SIP/IAX
* New features for PEAK & OFF-PEAK
Add new model for ratecard , removing week day and adding starttime and endtime instead.
* Add Voip Provider
* Add the RATECARD SIMULATOR
* Add support for Jiax web phone
* notenoughcredit_assign_newcardnumber_cid
IF the CARD doesn't have enough credit, request to enter a new cardnumber.
* Assign the CallerID to the new cardnumber
* Predictive Dialer Features
* Manage Campaign, Phonelist, Import Phonelist.
* Customer Interface (Agent) have the ability to call a predefined amount of Phone numbers.
* Support call at Zero-Cost & Negative cost (plus param = maxtime_tocall_negatif_free_route)
* CallerID authentication improvement
- (new param : notenoughcredit_cardnumber ;
cid_auto_assign_card_to_cid ; cid_auto_create_card ;
cid_auto_assign_card_to_cid)
* Popup Select Card - avoids long load (issue for user that have create lot of cards)
* PAYPAL SUPPORT - IPN - Customer can buy credit through paypal
* DID SELLING SUPPORT + DID monthly billing - features to sell to your customer preconfigured DID.
Customer would have the opportunity to redirect those to his phonenumber and even deploy a Follow-Me
* and lot of bug fixed and much more fancy stuff...

Other good stuff as well :
- WIKI -> http://wiki.asterisk2billing.org/
I hope it will help to build quickly a serious user manual, I know
that it's pain in $%& to understand the soft.
- FORUM -> http://forum.asterisk2billing.org/
Damn !!! Que demande le peuple !!!
- DEMO -> http://demo.asterisk2billing.org/
- UNLIMITED FREE CALL ON PSTN -> ... forgot the link!

Seriously bit of helps (documenting, dev...) would be greatly appreciated so if someone is willing to help/contribute, please contact me directly! Enough talk it's time to enjoy this new version, have fun and don't forget to send me your comments :P

Cheers,
/Areski

Help Article: Configuring iax.conf for IAX2 clients

2006-03-14T07:30:00.000-08:00

Click Here for Configuring iax.conf for IAX2 clients

Digium and Zimbra Bring Asterisk VoIP to there Collaboration Suite at Spring VON

2006-03-14T07:23:00.000-08:00

Digium Inc., the original creator of Asterisk and pioneer of open source telephony, and Zimbra, a leader in open source next-generation collaboration and messaging, today announced the integration of Voice-over-IP calling capabilities into the Zimbra Collaboration Suite (ZCS), by leveraging Asterisk. With this partnership, Digium and Zimbra are leading the way to the first open source Unified Messaging platform.

"We're really pumped that Zimbra and Digium are able to provide Unified Messaging this quickly by leveraging open platforms. Now, I can initiate a conference call with my engineering team, quickly access my voicemail or call home from the road - all through my e-mail," said Satish Dharmaraj, Zimbra co-founder and CEO. "What's beautiful about this is that it's all based on open standards. We've implemented a SIP integration with ZCS that has been tested and proven on Digium's open source Asterisk VoIP system."

Click Here for the Full Release

Esnatech Unveils Asterisk Based Real-Time Communication Solution

2006-03-14T07:20:00.000-08:00

Esnatech, a provider of enterprise real-time communications solutions, today announced it has joined the Digium(TM) Partner Program to deliver enterprise based Unified Communications solutions integrated with Asterisk(TM) IP telephony platforms.

Esnatech's Telephony Office-LinX is a next-generation, real-time communications platform integrated within an organization's telephony network. The release of IP integration with Asterisk will provide small business organizations a truly Unified Communications platform incorporating IP telephony with integrated IP applications targeted specifically for the enterprise business marketplace.

Digium is the creator and primary developer of Asterisk, the industry's first Open Source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched and Ethernet architectures. The Digium Partner program is designed to promote Digium and Asterisk related products. The goal is to form a closer relationship between Digium and companies who have incorporated Digium and Asterisk technology into their products.

Esnatech's Asterisk integration provides pure SIP-based Unified Communications solution. It provides secure server-based wireless messaging with access to messages conveniently from virtually any communication device, including office telephones, cell phones, PDAs, PCs or any Web browser. It bundles a suite of communication solutions including location-based routing, fax, online and offline access to presence management and text messaging, speech-enabled routing and corporate dialing, desktop call control, IVR and CRM integration, all bundled into one integrated SIP based platform.

Click Here for the Full Release

Newbie's Guide to Asterisk@Home 2.7: Unabridged Installation and Upgrade Guide

2006-03-13T09:28:00.000-08:00

Click Here for Newbie's Guide to Asterisk@Home 2.7: Unabridged Installation and Upgrade Guide

Ekiga 2.00 aka "The Oberoi Release" available!

2006-03-13T09:20:00.000-08:00

Ekiga is a SIP and H.323 compatible VoIP, IP Telephony, and Video Conferencing application that allows you to make audio and video calls to remote users with SIP or H.323 hardware and software. It supports all modern VoIP features for both SIP and H.323.

Ekiga is the first Open Source application to support both H.323 and SIP, as well as audio and video. Ekiga was formerly known as GnomeMeeting.

After more than one year of work, Ekiga 2.00 is finally available. Ekiga is now the first Open Source software to support both SIP and H.323 in the same application. GnomeMeeting was already a pioneer among the Open Source Voice over IP softphones, and we hope that Ekiga will continue in this path.
Among the features, you can find:

* Full SIP Support
* Full H.323 Support
* Audio and Video Support
* Call Transfer (SIP and H.323)
* Call Forwarding on Busy, No Answer, Always
* Call Hold
* DTMFs Support
* Basic Instant Messaging
* Ability to Register to Several SIP Accounts Simultaneously
* Possibility to Use an Outbound Proxy (SIP) or a Gateway (H.323)
* Message Waiting Indications (SIP)

Among the new features:

* Better Audio Quality
* Support for Wideband Codecs (16 kHz)
* Echo Cancellation
* Easier NAT traversal
* Improved camera Support
* Improved Video4Linux2 Support
* DBUS Support

Click Here for more Information

RIM to "Push" Voice Calls Into Corporate Desks

2006-03-13T09:15:00.000-08:00

The maker of BlackBerry email devices - RIM has acquired Ascendent Systems, a leading provider of solutions that simplify voice mobility implementations in the enterprise.

It is a SIP standards-based software solution that augments existing PBX (Private Branch Exchange) and IP-PBX (Internet Protocol Private Branch Exchange) systems and supports heterogeneous telephony environments to “push” voice calls and extend corporate desk phone functionality to mobile users on their wireless handset or any wireline phone. Ascendent will become a wholly-owned subsidiary of RIM. Terms of the agreement were not disclosed.

Click Here for the Full Article

Development news :: T38 passthrough support

2006-03-10T17:20:00.000-08:00

Friends in the Asterisk.org community,

There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users.

However, developers can't really get anywhere without a dialog with the users. You know what you need, you know what is missing and how you would like to
make Asterisk a better choice.

I am planning to send out a description of new features now and then, to inform you about what is going on, but also to get some feedback. The bug tracker is not only a tool for developers, but also for testers and users to react to changes and contribute.

*** ITU T.38 -- Fax over VoIP

Fax over VoIP is a hot issue. VoIP service providers encourage people to switch to VoIP but often forget to mention that faxing over VoIP is like russian
roulette. On a local LAN, it might work if you pick a clear channel codec like G.711. Steve Underwood, member of the Asterisk developer team, has writen a good article about the problems involved and the solutions for it on his web site, the URL is: http://soft-switch.org/foip.html

T.38 is an ITU standard for fax over VoIP. To simplify, the idea is to decode the fax audio stream at the ingress point, convert it to a data stream that is not
sensitive towards jitter or delays and encode it into audio again if needed at the other end of the call - if you can't convert it to an image somewhere in the middle and print it directly, or send it by e-mail.

*** T.38 PASSTHROUGH in Asterisk

Steve is the main contributor behind the work for T38 support in Asterisk. He's also
the author of spandsp - the fax application that many use in Asterisk. The first part
is to be able to send T38 calls to your Asterisk PBX and make Asterisk recognize
this and forward the data stream to another endpoint that supports T.38.

Asterisk won't be an T.38 endpoint, but will handle T.38 calls properly, regardless
if the T.38 was offered in the original call setup, or if the caller suddenly sends a fax in the middle of a call (a re-invite). The requirement is that the incoming channel and the outbound channel both supports T38. If not, the call will be
declined in a proper way.

When this is tested and stable, work will continue to see if we can make Asterisk an T.38 endpoint.

This is a very important addition to Asterisk. There is code for testing available.
If you are interested, please check this URL in the bug tracker:
http://bugs.digium.com/view.php?id=5090

I think this is a big step for Asterisk. Do you? If so, don't forget to say "thank you" to Steve Underwood - Coppice!

Have a nice weekend!

/Olle

Signate's CEO appears on Interviews with Ronald Lewis

2006-03-10T09:43:00.000-08:00

Signate, a VoIP open source telephony company, is led by its present CEO, William Boehlke. Boehlke's 25 years of senior level experience has afforded him opportunities with leaders such as Adobe, Baan, Borland, Lucent Data Networking, PeopleSoft and Knight-Ridder and successful start-ups such as Forte Software, Siebel Systems and Tibco.

Click Here for the Interview with Signate's CEO

Vonage cries foul over Canada VoIP "Tax"

2006-03-10T09:40:00.000-08:00

Note: This is a little off-topic for Asterisk but I felt you still needed to read this if you have not heard because it could affect us in the future.

Consumer IP telephony service Vonage has filed a complaint to Canadian regulators over plans by local telco, Shaw Cable, to charge a C$10 ($8.60) a month premium to customers of VoIP service. The charge ostensibly covers to cost of providing a higher quality connection to VOIP (Voice over Internet Protocol) users. Vonage describes the levy as a "thinly veiled" VoIP tax.

By using internet connections to make long-distance calls instead of conventional voice circuits users have the potential to make far cheaper calls. Vonage argues Shaw's fee undermines the healthy development of the market.

Click Here for the Full Article

ruby-agi-1.1.2 released

2006-03-09T09:20:00.000-08:00

Release notes of: ruby-agi-1.1.2
March 07, 2006

In this release bug # 3722 has been fixed
Details of this can be found Here

Feedback, suggestion, feature request, bug report is always appreciated.

For more information, please visit projects homepage:
http://rubyforge.org/projects/ruby-agi/

To install ruby-agi,
% gem install ruby-agi
and to update exiting ruby-agi
% gem update ruby-agi

Thanks,
Mohammad Khan
info beeplove com

MINNESOTA: TwinCities Asterisk Users Group -Saturday 03/11/2006

2006-03-09T09:15:00.000-08:00

SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC
"Keep in touch with the World"

The next Asterisk Users Group meeting has been scheduled for this Saturday March 11th at 11:30am.

Meetings are held monthly on the second Saturday of each month, excluding July and December.

Meetings are held at Sound Choice Communications LLC:
Google Map Directions

Sound Choice Communications is located in Bloomington Minnesota, just 1/2 mile west of the Mall of America. The address is: 7839 12th Ave S, Bloomington Minnesota 55425. We are just south of Hwy494 on 12th Ave. 12th Avenue is one exit west of Hwy 77 (Ceder Ave).

This month we'll hear from Shane Young and Dave Walters as they discuss integrating Asterisk with Tivo, Home security, Home Audio, and possibly X-10.

If you're having a problem with Asterisk, bring your questions to a meeting for free help. We love helping new users!

Come to a meeting to meet other asterisk users, see asterisk solutions, win a door prize, eat food, or for the good company, to look for work, if your looking for employees, to go out for a drive, to get out of your house, whatever, JUST COME TO THE MEETING!

Last month we gave away two licenses for the Cepstral Text to Speech software voices. Thank's Cepstral for your support!

In November we gave away an autographed copy of the O'Reilly book "Asterisk - The Future of Telephony". All three authors, plus Mark Spencer personally signed the book.

New visitors can help themselves to FREE FXO Interface cards (So you can connect your phone line, and have a timing source for meetme and IAX protocols). Some members have been known to swap hardware at the meetings. Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to see. Have you been to a meeting recently?

Please come and share your own ideas and learn from others. As always, free food.

We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything.

Meeting starts at 11:30am and parking is available in the rear of the building. Runs about 2 hours or less, and we'll order Pizza to the meeting for lunch.

Look forward to seeing you there.

VoIP Info Link

If you have a product or service you'd like to introduce to our members, send a private message to ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next month's sponsor.

Asterisk Based - Starface Released

2006-03-08T11:39:00.000-08:00

starface is a professional Voice-over-IP solution, which can reproduce the complete voice-communication of companies on a software basis.

Characteristic for starface Softswitch PBX is the innovative, process-oriented usage concept, which uses the advantages of the software basis, to reproduce the process of voice-communication ergonomically and intuitively.

The browser-based concept makes sure, that starface can be used on every client with a standard-browser - without any need to install additional software. By this means, starface potentiates the access on centralized resources - out on business or through mobile clients.

Click Here for more Information

Business 2.0 Names Fonality Top 25 Startup

2006-03-08T11:32:00.000-08:00

Fonality, the leader in affordable IP-PBX systems for small businesses and the world's largest distributed deployment of Asterisk, today announced that Business 2.0 has named Fonality one of "The Next Net 25." According to the magazine, Business 2.0 editors selected Fonality as one of 25 companies "in the vanguard" of the "new Web revolution."

Editors selected companies "whose approaches help illuminate where the Web is headed and where the opportunities lie." "The Next Net 25" includes five categories. Fonality was selected as one of five companies within the category titled "The New Phone."

Click Here for the Full Release

[Nerd Vittles] Asterisk Call Queues: The Smarter Way to Manage Incoming Calls

2006-03-07T08:47:00.000-08:00

Excerpt: "Ever wished you could screen incoming calls and route them to another person, or to an autoattendant, or to voicemail without the caller knowing what you're up to? Need a free Automatic Call Distribution (ACD) System for your business (with Elevator Music no less) that will balance incoming call workload among your employees? Ever wanted to prioritize incoming calls from different groups of callers? How about a simple way to access hidden features on your Asterisk system when you're away from your home or office?

Well, the latest Asterisk (1.2.x) now can do all of this without breaking a sweat... with Call Queues. And, with Asterisk@Home 2.5 and its included Asterisk Management Portal, you can build a complete Call Queue System in about half an hour."

Click Here for the Full Nerd

It's a Cisco Day - Cisco: Unified Communications System

2006-03-07T08:43:00.000-08:00

The new Unified Communications System aimed at streamlining business processes, and helping to drive productivity. Unified Communications (UC) will feature new presence, desktop tools, mobile integration and network intelligence to improve business agility and customer interaction. Cisco Unified Communications is fully embracing the SIP standard on their desktop phones.

Based on the Cisco Service-Oriented Network Architecture (SONA) announced in December 2005, the Cisco Unified Communications system is an open and extensible platform for real-time communications based on presence, mobility and the intelligent information network. It uses the IT data network as the service delivery platform helping workers to reach the right resource the first time by delivering presence and preference information to an organization's employees.

Click Here for the Full Article

Cisco Sees The Light On VoIP Protocol (Finally)

2006-03-07T08:36:00.000-08:00

Cisco Systems' IP-based PBX system was the only major system still not supporting a standard protocol that would cut the cost of voice over IP and pave the way for a new generation of VoIP applications. That all changes this week, as mounting customer pressure and the standard's potential finally convinced Cisco to get behind the Session Initiation Protocol.

The network-equipment vendor has planned a set of announcements that have the Session Initiation Protocol at their core. In addition to a SIP-compliant CallManager 5.0, they include SIP capabilities for Cisco IP phones, presence-awareness software, and multimedia communications software.

The messaging protocol delegates how VoIP phones establish contact and use call waiting, among other things. It will let customers mix and match VoIP products from different vendors. Support in CallManager 5.0 should make it possible for a customer to choose a cheaper SIP-based alternative to Cisco's VoIP phones, which can cost upward of $500 apiece. Cisco plans to support Research In Motion's BlackBerry and forthcoming Nokia dual-mode phones with Call Manager 5.0, though just about any standard SIP-based phone from two dozen or so vendors should work with the IP PBX.

Click Here for the Full Article

Asterisk 1.2.5 Released

2006-03-04T10:01:00.000-08:00

Asterisk 1.2.5 is now available for download on the ftp. See the ChangeLog for details about what has changed.

Click Here to Download

As mentioned in the release announcement for Zaptel 1.2.4, our releases now contain some extra files. The Asterisk release is available as asterisk-1.2.5.tar.gz. However, there is also a patch against the previous release as an option for a smaller download, asterisk-1.2.5-patch.gz.

For both the release tarballs and release patches, we have provided SHA-1 sums and PGP signatures. To verify the releases, you will need the public keys of both russell@digium.com and kpfleming@digium.com. Both are available on the keyserver, pgp.mit.edu.

Thank you for your continued support of Asterisk!

-- The Asterisk Development Team

The AstriCon Europe Tour: June 2006

2006-03-02T10:19:00.000-08:00

AstriCon returns to Europe this summer with a three-city tour. Events will be held in London, Paris and Berlin. The program at each tour stop will include top-notch technical speakers, a developer meeting, an exhibition of Asterisk products and services, an introductory seminar, and plenty of local content.

Click Here for AstriCon Information

Contact Center News: VoIP, Nuasis, SureWest, Unity4, Digium Asterisk PBX

2006-03-02T10:09:00.000-08:00

Nuasis Corporation, which calls itself "the IP contact center company," announces the signing of a "major contract" with Southwest Gas Corporation.

The Nuasis NuContact Center was preferred by Southwest Gas, Nuasis officials say, "in part because of its ability to manage multiple contact center sites with a single software application." The system will be used to apply common rules, policies and resources across Southwest Gas locations.

Independent telecommunications holding company SureWest Communications announced operating results today for the quarter and year ended December 31, 2005. Total operating revenues grew from $211.8 million in 2004 to $218.6 million in 2005, yielding net income of $6.4 million in 2005 as compared to a loss of $1.1 million in 2004.

Unity4 will deploy Indosoft contact center technology in Sydney, Australia. The computer telephony backbone will be based on the Digium Asterisk PBX.

Indosoft will supply the Dialer technology, integrate a soft-phone into the Unity4 Agent Desktop and co-develop a contact center specific ACD architecture and GUI interface for Asterisk.

Click Here for the Full Article

Sangoma CEO Addresses Open Source VoIP at VON Spring

2006-03-02T10:06:00.000-08:00

What:
Hardware scalability is a major concern for those working in the Open Source VoIP space. Come hear Sangoma Technologies (www.sangoma.com) David Mandelstam and his fellow panelists discuss "VoIP in the real world" as it relates to issues of security, high availability and flexibility.

Who:
David Mandelstam - President/CEO David and his research and development team focuses on Sangoma's family of AFT (Advanced Flexible Telecommunication) T1/E1/J1 voice/data cards that are engineered for today's demanding soft PBX, IVR and VoIP applications, such as Asterisk and Yate.

Where/When:
VON Spring Conference & Expo, San Jose, CA Free and Open Source VoIP: World Communication through World Cooperation Tuesday, March 14, 2006, 9:15am - 10:35am

Click Here for Info on Spring VON

Milliwatt Analyzer available

2006-03-02T10:03:00.000-08:00

Here it is: Mwanalyze
Download

It performs a Fourier analysis for a fixed frequency and tells the amplitude.

The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple of 0.5 ms, thus the valid frequences are: 2000 Hz, 1000 Hz, 666.666666667 Hz, 500 Hz, ...

Furthermore the application computes the ripple on that tone. In order to detect audiogaps and short noise on the line, one can define a treshold and a timeslice duration (typically 1s to 0.1s), and the application will compute the ripple for each timeslice and count the timeslices with a ripple greater than the given treshold.

Thus the application is a tool to verify the line quality, e.g. for least-cost-but-not-too-bad-line routings.

For conveniance Mwanalyze also generates a tone of the frequency it analyzes. Thus a bidirectional operation, and test for frequencies other than Milliwatt's 1000 Hz are possible. Anyway Milliwatt is much much more economic to CPU and RAM!

For details see inline documenation or output while loading the module app_mwanalyze.so!

Now, I will try to contact to dev-list, in order to put this application to future releases.

Roger.

[Nerd Vittles] Follow-Me Roaming with Asterisk: Transparently Integrating Mobile Phones Into Your Dialplan

2006-03-01T06:34:00.000-08:00

Click Here for Nerd Vittles Follow-Me Roaming with Asterisk: Transparently Integrating Mobile Phones Into Your Dialplan

Xorcom TS-1 solid state Asterisk server released

2006-02-28T11:25:00.000-08:00

Click Here for Xorcom TS-1 solid state Asterisk server released

OpenSER v1.0.1 released

2006-02-28T11:21:00.000-08:00

OpenSER is a project spawned from FhG FOKUS SIP Express Router (SER). The reason for this new venture is the lack of progressing and contributions to the SER project from the other SER team members as well as the reticience to new contributions from project's community members. We want to accelerate the integration of public contributions to the SER project.

OpenSER promotes a new management policy (OPEN) -for new code acceptance and code-through propagation- and development approach -design and architecture. We have decided to bring more dynamics into SIP world by creating this new project that can benefit of TLS and so many other contributions. We welcome your contributions to the success of this project.

Click Here for more Information

Turbolinux Launches IP-PBX Software to Support Broadband IP Phone Services

2006-02-27T12:56:00.000-08:00

Turbolinux, Inc., a global leader of Linux-based solutions, today announced the sales launch of IP-PBX software InfiniTalk, which is based on the open source software Asterisk. InfiniTalk improves and upgrades the IP telephone environment for significant cost reductions.

InfiniTalk IP-PBX software is generally considered the best choice for a low-cost, next generation standard IP telephone system. The combination of the Linux and Asterisk open architecture and rich hardware allows customization of the software for specific applications and customers. InfiniTalk software supports the majority of standard telephony equipment. InifiniTalk also supports the newer broadband IP phone services and fiber optic technology provided by NTT East and West Corporation. These capabilities work to create a cost-effective, IP phone system in the business environment.

Click Here for the Full Article

AstriDevCon Europe 2006

2006-02-22T15:40:00.000-08:00

From May 8-11 in Pisa, Italy, a group of Asterisk developers will be getting together for four days of hacking, coding, testing, designing and otherwise beating on the Asterisk code base. The event will be hosted at the University of Pisa (thanks to Luigi Rizzo), and will be low-key and open only to serious Asterisk developers and contributors. We are expecting to keep the attendance to 15 people or less, and we already expect to see these people:

Attendees:

Mark Spencer (Digium)
Kevin P. Fleming (Digium)
Luigi Rizzo (University of Pisa)
Christian Stredicke (Snom)
Christian Richter (BeroNet)
Olle Johansson (Edvina.net)
Joachim Vanheuverzwijn (zoa, Securax)

If you wish to participate, please contact me off-list so I can make arrangements with you. We will need to have the final list of attendees in place by March 15th or so, so that hotel accommodations and conference room space can be scheduled. Luigi has suggested that we stay at the Hotel Roma which is within walking distance of the University, the main square and the Tower, so that seems the best choice.

Note that this is _not_ a sponsored event (other than the university providing workspace and network access)... each attendee will be responsible for their own travel, lodging and meals.

The timing of this event means that Asterisk 1.4 will have already entered feature freeze mode and will be in beta-testing, so the goal of the conference is to concentrate on architectural changes and other work targeted for Asterisk 1.6 (to be released around the end of 2006). If it goes well and is productive, we may arrange a similar event in the USA for later this year so that those who cannot attend in Europe can participate.

Posted By: Kevin P. Fleming

Dev: Test my test-branch!

2006-02-21T13:25:00.000-08:00

The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback.

For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is written by me. There are large contributions from other developers, code that I maintain in several open subversion branches in order to help them stay up to date with their work.

To assist the testing group and make life easier, I've combined a lot of patches into one superbranch for testing. I've added the README further down.

** PLEASE help the community, please test this branch.

Check it out like this:

SVN Checkout

Then cd into test-trunk and run "make" then "make install"

Report any bugs in the proper open bug in the bug tracker. If you like new functions, add a comment that this works for you. Provide feedback, make our work easier.

Run "svn update" from time to time to get the latest version. Any changes from trunk will be merged into this code. Read the README.test-this-branch file to get more information.

Thank you for your help!

P.S: Obviously, this is test code, not recommended to be closer than 2 miles from your production servers.

----- README.test-this-branch

----------
TESTING BRANCH - WELCOME!!
----------
Asterisk is developed by the Asterisk.org user community. The development team does not only consist of coders, but also testers and people that write documentation and check for security problems.

This is a combined branch of many patches and branches from the bug tracker that needs your testing. Please test and report your results in the bug tracker reports for each patch.

What's in this branch?
----------------------
This branch includes the following branches:

- sipdiversion: Additional support for the Diversion: header
- jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854)
- videosupport: Improved support for video (#5427)
- peermatch: New peer matching algorithm (no bug report yet)
- rtcp: Improved support for RTCP (#2863)
- dialplan-ami-events: Report dialplan reload in manager (#5741)
- sipregister: A new registration architecture (#5834)
- subscribemwi: Support for SIP subscription of MWI notification (#6390)

Coming Here Soon:
- iptos: New IPtos support, separate audio and signalling (#6355)
- metermaids: Subscription support for parking lots (#5779)
- multiparking: Multiple parking lots (#6113)

And the following stand-alone patches
- New CLI commands for global variables (#6506)
- Additional options for the CHANNEL dialplan function

All of these exist in the bug tracker

* PEERMATCH: New object match for incoming calls. Skip the "user" :-)
---------------------------------------------------------------------
In this code, we will match incoming calls like this:

- First user on From: user name
- Then peer on From: user name *** NEW ****
- Then peer on IP and port

This means that in most configurations, you can configure a phone entry as "type=peer" instead of "type=friend". Subscriptions will work much better with just one object to match.

/Olle

Introducing Telephone Reminders 2.5: The Asterisk Telephone Reminder System

2006-02-21T08:42:00.000-08:00

Excerpt: Using nothing but a phone call, you can schedule reminders for the near or distant future, specify different numbers for the return calls, and customize a recorded message for each call. In short, it's perfect for appointment reminders, birthday reminders, anniversary reminders, and anything else you want or need to remember.

Click Here for the Full Nerd

Create your own Voice-over-IP PBX using Asterisk

2006-02-21T08:29:00.000-08:00

Asterisk is already hard at work in South Africa. Its being used as a PABX, for call-recording, for both small and large call-centres, for voice conferencing, and in CTI (Computer Telephony Integration). Its providing inter-office "free" calling, and very inexpensive international calling. In every case, Asterisk-based solutions are a fraction of the cost of the traditional equivalents.

Voice-over-IP (VOIP) is built right in to Asterisk. Connecting into the traditional phone network is done using interface cards readily available in South Africa, or by connecting into an ITSP (Internet Telephony Service Provider) via VoIP over the Internet.

Asterisk provides all the functions of even the most expensive traditional PABXes simply with software running on an ordinary PC. What's more it comes with all the open-source goodness that Tectonic readers know and love.

At Connection Telecom we like to say that the coming together of VoIP and the open source world is resulting in the most dramatic change in the world of telephony since we last heard "Nommer Asseblief?" ["Number Please" in Afrikaans].

Click Here for the Full Article

Asterisk on OpenWrt

2006-02-18T10:00:00.000-08:00

Asterisk is free software that lets you create a fully functional, easily customizable, private branch exchange (PBX). Businesses like Asterisk because they can save money by using it, and because it is open source, they can add functionality to it easily and inexpensively. Asterisk is also becoming popular with home office users -- so much so that it spawned a new project called Asterisk@Home, which released its 1.0 version last year. Now there's even a version of Asterisk that runs on OpenWrt, a Linux distribution designed to run on your wireless router. I found it to be worthwhile, but I wouldn't depend on it for my home office.

I installed Asterisk on OpenWrt White Russian RC4 on a Linksys WRT54GS wireless router. It's my first Asterisk installation. I admit that I scraped the knuckles on both hands getting Asterisk correctly configured, but now that I've done it, I would say it was worth all the frustrations it caused me. Not only do I now have a functional personal PBX, I've also learned a little about the black art of telephony along the way.

nstallation was a snap. All I had to do was point my browser at the WRT54GS's IP address, log in at the OpenWRT Admin Console, and then click the install button next to the Asterisk and Asterisk-sounds packages. The install was finished, but I still had a long way to go.

Click Here for the Full Article

Article: Open source Asterisk PBX getting more Popular

2006-02-16T08:14:00.000-08:00

Asterisk, the open source PBX system made by Digium, is gaining ground with companies and governments alike. The cost savings over proprietary PBX systems can be substantial, but Mark Spencer, president of Digium, told TG Daily in a short interview at the Southern California Linux Expo, that "choice" is the main reason companies adopt Asterisk. "Customers have a choice in how they configure they system, in the hardware they buy or the graphical interface they want," says Spencer. This choice also allows companies to add extensions or make changes in seconds compared to days with a traditional PBX.

Click Here for the Full Article

Zaptel 1.2.4 Released

2006-02-15T19:44:00.000-08:00

The Asterisk/Zaptel development team is pleased to announce the release of Zaptel 1.2.4.

This release contains a number of bug fixes, along some with new functionality:

* The driver for the Xorcom Astribank has been incorporated into this distribution. Xorcom will provide primary support and driver maintenance for customers using this product.

* The driver for the Digium Wildcard TDM2400P has been upgraded to support revision B of the VPM100M echo cancellation module.

* The special parameters required for the Digium Wildcard TDM400P when used on the Australian PSTN are now automatically set when the opermode is set to 'AUSTRALIA'.

The release is available on the Digium FTP Servers under the name zaptel-1.2.4.tar.gz, and also as a patch from version 1.2.3 (in file zaptel-1.2.4-patch.gz).

In addition, beginning with this release we have included an SHA-1 sum of the files (in files zaptel-1.2.4.tar.gz.sum and zaptel-1.2.4-patch.gz.sum) and GPG signatures (in files zaptel-1.2.4.tar.gz.sign and zaptel-1.2.4-patch.gz.sign) verifying that
this is an official Zaptel release.

You can retrieve the public keys for kpfleming@digium.com and russell@digium.com from the keyserver, pgp.mit.edu.

Next Montreal Asterisk Meeting - 02/21/2006 - Featuring a conference call with Mark Spencer

2006-02-15T11:14:00.000-08:00

This is a reminder about our next meeting.

It will be held from 6pm to 8pm, February 21 at Modulis Offices which are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal.

Thanks to Claude Patry, we will be having a 20 minute conference call with Mark Spencer.

If you'd like to ask Mark a question, please send it to me by email. We are limited to 5 questions, and will do our best to select those to be presented.

Please confirm your attendance at this meeting by replying to this email.

See you Next Week,

--
Adrien Laurent
adrien@modulis.ca
www.modulis.ca
(514) 284-2020 x 202

PIKA Technologies Announces Support for Asterisk PBX

2006-02-15T10:40:00.000-08:00

PIKA Technologies today announced that they have integrated PIKA's high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software
layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability between PIKA high-density analog boards (Daytona MM) and Asterisk PBX software.

"The Asterisk development community can now benefit from advanced features for fax and echo cancellation in high density analog applications, made possible by PIKA's DSP processing power on the board," stated Wojciech Tryc, Enterprise VoIP architect at PIKA Technologies. "Because of the native bridging for TDM calls, latency is nearly eliminated in this implementation. The solution is very reliable, as we have witnessed not only in the lab, but in live customer environments."

Asterisk developers can be up and running quickly with PIKA Connect for Asterisk and PIKA hardware. "For those familiar with using the Asterisk platform, no additional training is required. They can take advantage of the PIKA solution with minimal effort or investment," said PIKA Technologies, Wojciech Tryc.

For more information on PIKA Connect for Asterisk go to:
Pika Technologies Asterisk Info

Help Article: Configuring voicemail.conf for Asterisk

2006-02-15T10:15:00.000-08:00

Click Here for Configuring voicemail.conf for Asterisk

Firmware version 1.3.1 released for Aastra IP Phones

2006-02-14T20:50:00.000-08:00

Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of IP phones (480i, 480iCT, 9112i and 9133i).

The firmware and release notes (no updated admin and user guides yet) are available for download at:
http://www.aastra.com/support/enterpriseip

Contrary to what the version numbering would suggest, this is a significant update with many new features and bug fixes. See the release notes for full details, but here are some hightlights for Asterisk users:

- Context-sensitive softkeys. Softkeys can now be configured for each of the following call states: idle, incoming, outgoing and connected
- Speed dial using the BLF key
- Per-line outbound proxy
- Use the Icom key to make intercom calls
- Further XML enhancements
- Voice quality (transmit level) issues resolved
- Keypad now continues to work when a second incoming call appears

And much more.

Nerd Vittles: Introducing TeleYapper 2.5: The Free Asterisk Message Broadcasting System

2006-02-14T09:23:00.000-08:00

Click Here for Nerd Vittles: Introducing TeleYapper 2.5: The Free Asterisk Message Broadcasting System

Video conferencing over Asterisk unveiled

2006-02-12T18:15:00.000-08:00

Adiance have succeeded in becoming one of the world's first companies to add native video support for Asterisk, offering the most advanced video solutions - such as, video conferencing, and video broadcasting.

Released recently, this technology will be integrated into Sidance's existing range of products - including inbound/outbound VoIP Call Center Software Solution - Sidance Enterprise 2006, Voice & Video Broadcasting System, VoIP telecom gateway, Soft phone with multi-user Video Conferencing support and other Asterisk VoIP software solutions.

This technology empowers users to do with video virtually everything they can do with voice on a powerful Asterisk solution platform. Combination of open source Asterisk software and state of the art Sidance technology enables customers to build VoIP video solutions at more than 40% cost savings as compared to available solutions in the market.

Adiance's evolutionary technology will help service providers to offer on-demand video broadcasting/ conferencing services to their Asterisk users at huge savings with the elimination of proprietary video infrastructure hardware. Video conferencing right now is very costly and time consuming technology. With VoIP - Video Over IP, it is cheap and fast. With this technology, Video is sent using normal Internet bandwidth as is the case with audio in Voice over IP. It is also possible to offer Video voice mails, Video on Hold application, Video recording, Text messaging and other services.

Contact:
Siji George
14, Empire Tower
CG Road, Ahmedabad
Gujarat India 380009
Phone: 91 9879200499
http://www.adiance.com

Click Here for the Full Release

Dev Info: Revised Codecs/ Implementation

2006-02-11T11:15:00.000-08:00

I noticed that the various files in codecs/codec_*.c contain a large amount of replicated code in the newpvt, framein, frameout callbacks, buffer definitions and so on.

Additionally there are several bugs in there, from null pointer dereferences (e.g. on malloc failures, the code does check, resets the pointer to NULL and then proceeds to use it as if it were good), to (less severe but terribly confusing) comments that have nothing to do with the code that follows (they refer to the file used as a
template).

I have committed in:

http://svn.digium.com/view/asterisk/
team/rizzo/base/

a revised implementation of the codecs interface, where most of the
common functions are moved to translators.c, so the individual
codecs can just use the generic functions in most cases.

Please read the comments in

include/asterisk/translators.h

that describe the architecture (hopefully it is clear enough; if not, ask).

The results are very interesting - codec_*.c reduced from ~5000 to ~3600 lines, and the code is very consistent now.

One area that can still be improved a lot is the generation of 'sample' frames for each codec. Right now, except one or two cases, those frames are just chunks of silence of various lengths, which is not the best input to evaluate a codec's performance (used when building the translation matrix).

I would suggest to move to a slightly different approach where the input is the same for all - a piece of slin data - and we do a first pass using the slin-to-FOOtranslator to generate a frame in format FOO, and then use these frames as input for the actual evaluation. This would remove the need for a 'sample()' callback from all codecs that can do the slin-to-FOO translation, requiring them only for
those (none at the moment) that do not support direct or indirect translation from slin.

Testing and feedback welcome.

cheers
luigi

Asterisk VoIP gets South African Accent

2006-02-10T09:31:00.000-08:00

True Voice Communications and Connection Telecom have jointly released a South African package of over 340 free prompts for the open source Asterisk platform. The free prompts, recorded in a neutral South African English female voice, can be used to replace the usual Canadian accented prompts.

"Customers can now replace the standard North American-English pack with one more suited to their customers," says Connection Telecom CEO Rob Lith. "Research and our own experience have taught us that consumers want to deal with the familiar. This is also is a step toward legitimising South Africa as a business destination that should be taken seriously," says Lith.

"Although the previous voice of Asterisk, Allison Smith, will continue to have very dear place in all Asterisk techie hearts, we believe its time to adopt a more local flavour," says Lith.

The package also offers pre-recorded IVR prompts, which can be customised using the same voice talent as in the packages.

Click Here for the Full Article

Asterisk Native Sounds re-release

2006-02-10T09:24:00.000-08:00

Hello everyone,

It seems that the letter "s" did not make it into the original release.

Please visit www.astlinux.org and download the latest tarball. Or, if you just want "s" in all of the available formats, just grab this:

http://mirror.astlinux.org/sounds/s.tar.bz2

Sorry!
--Kristian Kielhofner

Help Article: Configuring X-Lite for Asterisk

2006-02-09T09:01:00.000-08:00

This tutorial is not a comprehensive review of X-Lite. The purpose of this post is to set up a soft phone for evaluation. Soft phones can save money over the purchase of SIP handsets, but I strongly encourage you to test the soft phones before making a purchase decision. Soft phone quality varies widely depending on network conditions, codecs and protocol. This tutorial will help you evaluate the feasibility of soft phones with Asterisk. Before we start with X-Lite lets set up SIP and dial plan.

1) Change directory to zaptel source directory.
example:
[matt@localhost ~]$ cd /etc/asterisk

2) I'm using nano to edit the file, but pico, vi, emacs, or any text editor will do.
example:
[matt@localhost asterisk]$ nano sip.conf

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 192.168.1.x ; x = Asterisk server IP address
allow = ulaw ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here

[9250]
type=friend
username=9250
secret=password
host=dynamic
context=from-sip
mailbox=9250
nat=no
canreinvite=no

[9251]
type=friend
username=9251
secret=password
host=dynamic
context=from-sip
mailbox=9251
nat=no
canreinvite=no

3) Okay, we have the configuration for two clients on the SIP server. Now we have too make two extensions. These extensions will not include voicemail.
example:
[matt@localhost asterisk]$ nano extensions.conf

[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]

[from-sip]

exten => 9250,1,Dial(SIP/9250,20)
exten => 9250,2,Hangup

exten => 9251,1,Dial(SIP/9251,20)
exten => 9251,2,Hangup

4) Now we need to download X-Lite and install on our Windows PC. Download the soft phone from http://www.xten.net/index.php?menu=download. Run the install. Open X-Lite click on the menu button.

Click on 'System Settings'. Then choose the 'SIP proxy' option. Click on default and continue to proceed with filling out the SIP client configuration. When you are done t should look like this:

5) Now you are ready to login and test some calls.

To Be Continued....Next Week

Note: My name is Matt Birkland, I work as a VoIP Engineer for VoiceIP Solutions an Asterisk Provider in the Seattle area. Every Week I will be submitting a one page Asterisk/VoIP tip of the week on the blog. Next week we will build on this subject by reviewing codes and at some point we'll move on to common network issues or IAX tunnels... haven't decided yet!

LinuxWorld Magazine - Asterisk: Getting Connected Part 2

2006-02-09T08:40:00.000-08:00

Excerpt:

"
In December's column, we installed the Asterisk PBX and configured two IP phones as extensions. This month, we'll connect our PBX to the telephone network for incoming and outgoing calls and set up the Digital Receptionist to route our incoming calls.

Connecting Asterisk to the Telephone Network Asterisk supports a myriad of ways to connect to the public telephone network. Asterisk provides standard technologies such as IAX (Inter-Asterisk-eXchange), SIP (Session Initiation Protocol), and PRI (with appropriate hardware). We'll be connecting with IAX because it's simple and widely supported by VoIP providers."

Click Here for the Full Article

Interview: Sangoma CEO David Mandelstam

2006-02-08T11:13:00.000-08:00

Here is a link to an interview with Sangoma's CEO David Mandelstam with Ronald Lewis.

Nerd Vittles: Installing Asterisk@Home on Your Windows PC for Free

2006-02-07T07:53:00.000-08:00

Click Here for Nerd Vittles: Installing Asterisk@Home on Your Windows PC for Free

Announce: New issue tracker for handling licensing issues for Asterisk, Zaptel and related projects

2006-02-06T20:40:00.000-08:00

In an effort to ensure that every licensing issue brought to our attention is handled fully and openly, we have created a new issue tracker for this purpose. It is located at:

http://licensing.digium.com

The tracker is open to the public, and we encourage all interested parties to post their concerns and participate in the discussions involved in resolving them. Digium's will actively respond and pursue resolution of each and every issue posted, and other concerned parties may also participate.

In the future, if you have a question or concern related to the licensing of any of these projects (including packaging, trademark and other related issues), please open an issue in the tracker rather than posting to one of the mailing lists.

Thank you for supporting Asterisk and Zaptel!

Dev: Automerge changes... finally worked out

2006-02-06T20:34:00.000-08:00

I finally figured out the source of the problems with automerging of developer branches from the trunk (this was not an issue with developer branches based on branches/1.2).

The core issue that the 'svnmerge-integrated' property was being used for two conflicting purposes: to track the branches/1.2 fixes that had been forward-ported into the trunk and also to track the trunk changes that had been merged into the developer branch. Obviously this cannot work :-)

To solve the problem, the branches/1.2 forward-porting properties on trunk are no longer using the standard svnmerge property names; they are now called 'branch-1.2-merged' and 'branch-1.2-blocked', and there are _no_'svnmerge-integrated' or 'svnmerge-blocked' properties on the trunk. This means that forward-porting patches from branches/1.2 requires specifying the property names to svnmerge; see the
branching/merging page on asterisk.org for an example.

All new developers branches made from trunk should work with automerging without any problem, but existing branches have an issue: the developer branch contains a property 'svnmerge-integrated', but the trunk does not, and running 'svnmerge merge' will try to update that property, resulting in a conflict. To resolve this issue, for each branch that you maintain, you will need to manually use svnmerge to bring it up-to-date to at least revision 9163 (where the property was removed from the
trunk); from that point forward, automerge will be able to manage the merges for you.

In the meantime, you will probably start to receive conflict notices because your branches cannot be merged... but once we are past this issue, automerging should work well and should no longer generate 'false' conflicts like it was doing before.

Posted by: Kevin Fleming