Voice Over Internet VoIP

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.

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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

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.

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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.

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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

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."

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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 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

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: