Idefisk supports now IAX and SIP !

Main features:

• SIP + IAX protocols
• IAX2 protocols
• Available codecs
• GSM
• ulaw
• alaw
• speex
• ilbc
• STUN support
• STUN server per account
• Three lines
• Up to Two accounts
• Echo cancellation
• Codec settings per account
• Account password encryption
• Address Book
• Quick Dial

Feature Keys:
• Hold button
• Transfer button
• Quick Dial button
• Numpad
• Slide bars for speakers and microphone volume control
• History button

Website: www.asteriskguru.com

Cheers
Daniel

Web-Call-Back is the module for Joomla! that allows, in cooperation with a Asterisk PBX server, to perform a call back to a visitors provided phone number directly from your website. To make use of this module a Asterisk PBX server or compatible software PBX is required. The modules settings allow fully customization of the messages, in multiple languages with JoomFish installed, and styling via CSS. It requires JavaScript on the client browsers side as it performs its work in client-server communication with the server side module in Joomla. As overview Web-Call-Back allows: * To enter any international phone number on the frontend * Initiation of the call-back, either by internal extension first or outbound call with transfer to internal extension * Fully customizeable messages for phone number form and status messages of the module * Date & Time rules to be defined to show the module only during office hours or any specified period of times * Show messages or hide the module when the internal extension or the PBX system is offline and the call back is unable to get performed. * Output of debug information on communication with the Asterisk PBX. Prerequisite for this module to perform its functions is a Asterisk PBX system for call initiation and transfer. This module requires knowledge and ability to configure a Asterisk PBX system, without these individual prerequisites this module will become difficult to startup

VICIDIAL is enterprise predictive dialer for phoneCUBE. It's a opensource solution, therefore it's free...

With this solution we are able to offer a enterprise predictive dialer solution for call centers.

Key-Features are:

Ability for an agent to call clients in succession from a database through a web-client
Ability to display a script for the agent to read with fields like name, address, etc. filled-in
Ability to set a campaign to auto-dial and send live calls to available agents
Ability to dial predictively in a campaign with an adaptive dialing algorithm
Ability to dial on a single campaign across multiple Asterisk servers, or multiple campaigns on a single server
Ability to transfer calls with customer data to a closer on the local system or a remote Asterisk server
Ability to open a custom web page with user data from the call per campaign
Ability to autodial campaigns to start with a simple IVR then direct to agent
Ability to park the customer with custom music per campaign
Ability to send a dropped call to a voicemail box per campaign if no agent is available
Ability to set outbound CallerID per campaign
Ability to take inbound calls grabbing CallerID
Ability to function as an ACD for inbound and fronter/closer verification calls
Ability to have an agent take both inbound and outbound calls in one session(blended)
Ability for agents to log in remotely and have calls redirected to any phone number
Ability to start and stop recording an agent's calls at any time
Ability to automatically record all calls
Ability to call upto two other customer numbers for the same lead
Ability to schedule a callback with a customer as either any-agent or agent-specific
Ability in Manual dial mode to preview leads before dialing
Ability for agents to be logged in remotely anywhere with just a phone and a web browser
Faster dispositioning of calls with agent key-binding (HotKeys)
Definable Agent Wrapup-time per campaign
Ability to add custom call dispositions per campaign
Ability to use custom database queries in campaign dialing
Recycling of Busy calls at a specified interval without resetting a list
Dialing with custom TimeZone restrictions including per state and per day-of-the-week
Dialing with Answering Machine Detection, also playing a message for AM calls
Multiple campaigns and lead-lists are possible
Option of a drop timer with safe-harbor message for FTC compliance
Variable Drop call percentage when dialing predictively for FTC compliance
Internal DNC list can optionally be activated per campaign
All calls are logged and statuses of calls are logged as well as agent time breakdowns
Load Balancing across multiple inbound or outbound Asterisk servers is possible
Several real-time and summary reports available
Real-time campaign display screens
3rd party conferencing(with DTMF macros and number presets)
3rd party blind call transfer
3rd party conferencing with agent drop-off
Ability to set user levels and permissions for certain features and campaigns
Ability for managers to listen-in on agent conversations
Ability for managers to enter conversations with agents and customers
Web-based administration
Client web-app web pages available in English, Spanish, Greek, German, French, Italian, Polish, Portuguese and Brazillian Portuguese
Admin web pages available in English, Spanish, Greek and German

and much more....

Website: http://astguiclient.sourceforge.net/vicidial.html

Let me know if you like to know more.

With best regards
Daniel

The inventor of PGP did it again. He is the driving force behind the new ZRTP protocoll.

With ZRTP we are able to encrypt SIP phone calls!! No one can listen into your calls anymore....

The solution is distributed as SDK (software development kit) and as
SIP phone (Zfone).

The ZRTP protocol used by Zfone will soon be integrated into many standalone secure VoIP clients, but today we have a software product that lets you turn your existing VoIP client into a secure phone. The current Zfone software runs in the Internet Protocol stack on any Windows XP, Mac OS X, or Linux PC, and intercepts and filters all the VoIP packets as they go in and out of the machine, and secures the call on the fly. You can use a variety of different software VoIP clients to make a VoIP call. The Zfone software detects when the call starts, and initiates a cryptographic key agreement between the two parties, and then proceeds to encrypt and decrypt the voice packets. It has its own little separate GUI, telling the user if the call is secure. It's as if Zfone were a "bump on the wire", sitting between the VoIP client and the Internet. Think of it as a bump in the protocol stack.

Website: zfoneproject

I will test the Zfone with phoneCUBE in the next couple of days.

Regards
Daniel

Finally we can build "decent" voice recognition systems. LumenVox
is providing us a solution which is good enough to build great speech recognition IVR's.

Go ahead: Speech-enable our phoenCUBE, give the freedom of hands-free phone interactions to our clients, or simply provide an automated interface for customer service.

The Speech Engine is directly and seamlessly integrated with the phoneCUBE platform and Dial Plan through a unique connector bridge that Digium created. Now we can easily build speech-enabled IVR's by using the familiar Dial Plan scripting language or the C-API.

It provides speech application developers with an efficient development and runtime platform, allowing for dynamic language, grammar, audio format, and logging capabilities to customize every step of their application. Grammars are entered as a simple list of words or pronunciations, or in the industry standard Speech Recognition Grammar Specification.

I will test the solution and update you, on how we can sell and implement it for our clients.

Cheers
Daniel