Voice Over Internet VoIP

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

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."

Click Here for the Full Article

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:

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 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