



Get your free VOIP Solution software with answering machine with free voicemail account. No charges, no strings, completely free phone number you can give out as your own. You get free voice mail answering machine included for free too!
Secure VoIP for private internet talk by encrypting voice communications on computer network. It securely transports voice data through strong encryption technology, advanced key exchange process, and user authentication using digital certificates
VOIP H323 DLL Soft Phone SDK brings H.323 protocol support for ActiveX. With this SDK one can create a simple program to connect and start speaking with anyone with a direct IP Address or use the Gateway or the Gatekeeper to connect to /or from PSTN.
Telepati SIP Phone Freeware allows you to make PC-PC phone-phone calls over the Internet. Developed using Research Labs VOIP SIP Phone SDK, this free soft phone brings SIP protocol support for ActiveX. Tested and works with 75% of the providers.

VSoft3
$0 - MoBif Berhad
The VSoft Phone is a simple to use internet telephone system that can reduce your call costs. It uses the latest technology to let you make calls to friends, family, and business associates anywhere in the world without any chunky telephone equipment. Put simply, it turns your PC into a telephone.
The VSoft Phone utilises a Broadband internet connection to allow a user to make VoIP calls to any phone number worldwide.
VSoft features includes unlimited free VoIP to VoIP call (PC to PC), free video call, low cost VoIP to home/mobile call, send SMS, confrence, SMS to call, VoIP bridge. It is free to download and easy to install.

PBXPress Live CD Demo
$499 - PBXPress Communications, Inc.
PBXpress is a modern-day replacement for traditional PBX systems. Instead of working with wires and cables, it utilizes Internet technologies. In particular, it uses VoIP (voice over IP) technology to transmit calls.
All of your phones are connected and provisioned via an IP network, your calls can be delivered to any place on Earth via Internet at a fraction of the usual cost, and all system configuring is done from your web browser! In addition to new exciting features, PBXpress offers the reliability of good old telephony service: in case of any problem with your Internet connection, PSTN fallback is engaged, and the call goes via one of the telco phone lines.
Worried about who made all those calls to the Dominican Republic? Wish you had some record of your previous verbal agreement with a vendor? Want to analyze whether your new salesperson does the right thing on the phone? No problem: PBXpress can record every Introduction conversation made through it, storing them on CD-ROM or DVD-ROM in a format which can be played back on any computer.
You've tried to use a residential VoIP company's service in your office, but couldn't get any audio? The most likely reason for this is that your firewall is too strict, and does not allow network traffic to pass. PBXpress can act as a bridge across the firewall for VoIP calls, eliminating such problems.
This demo is a bootable CD. You must burn it as an ISO image file, not a data file. Boot your PC from the CD you have just burned. You may need to access your computer's BIOS setup (typically, press Del or F2 immediately after restarting your computer) in order to set CD-ROM as the first boot device.
As soon as your PC boots from the CD, you will see PBXPress graphical user interface accompanied by the on-screen emulator of the actual PBXPress box LCD. If a CD burns successfully, you will see a progress bar. After the initial phase is finished, you will see an IP address. Point your Web browser to the IP.

Solution4VOIP VoipSwitch
$1500 - Solution4VOIP
SOLUTION4VOIP is a platform that allows to implement various types of Voice Over the Internet Protocol (VOIP) services, with retaining shared, uniform management interface. The feature that distinguishes this platform is the implementation of an integrated, embedded billing system that cooperates with SQL – MS SQL or MySQL databases’ servers.
This solution results in the simplicity of preparing the system, by the operator, to be fully functional for providing services and also for administrating it in the future.
VoipSwitch’s software consists of the following modules:
VoipSwitch manager – the main part of the application. It allows to monitor the whole incoming traffic. Apart from the current connection status and the type of logged in clients, it also presents a number of additional information on the processes that are taking place.
VoipSwitch Config management interface. It is a tool for configuring the whole system. It has a number of features that allow to manage clients’ billing and analyze traffic information, based on the statistical data.
Web CDR the module that allows clients to check their current account status and the history of the calls made. There is a possibility of exporting data to the file from this level.
PC to Phone Client softphone based on the g723.1 codec.
Webphone softphone that can be initiated directly from the website.
Callback Client client’s software that allows to initiate calls between two telephones.
Webcallback client’s callback version that is initiated directly from the website
SMS callback module that cooperates with the SMS operators. It also allows to create access number for SMS callback service using the mobile phone
IVR module module responsible for playing back voice messages utilized by various services e.g. 2 stage dialing, account balance or IP PBX
Online Shop web based system that allows resellers to create users accounts, manage them, manage the tariffs, and analyze the information on the traffic that is taking place etc.
Reseller’s module web based system that allows resellers to create users accounts, manage them, manage the tariffs, and analyze the information on the traffic that is taking place etc. .
CallShop windows based application that is to be installed in internet callshops, it gives real time information on connection status, the telephone number, time of the connection, its cost etc

VoIP EVO SDK for Windows and Linux
$899 - conaito Technologies
conaito VoIP EVO SDK - A simple way to VoIP-enable your applications and websites.
The conaito VoIP EVO SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users. conaito VoIP EVO users are arranged in a tree-structure where each node is a room/channel where users can talk, send instant messages and share files. More than 40 users can be in each room/channel and participate in a conference while resource usage, both memory-wise and CPU-wise, is still kept at an acceptable level for gaming and other types of resource intensive tasks.
Here is a list of the main features of the conaito VoIP EVO VoIP client:
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (Speex Codec)
* Full Duplex - VoIP conferencing solution (multi-user conferences)
* Comes with Windows as well Linux server (running as service).
* Work with DirectSound as well with Windows Audio.
* Remote console server administration (e.g. use telnet to control the server)
* Administrator EVO Server API (VoIP EVO Admin COM control) - 2 SDK versions (VoIP EVO Client SDK and VoIP EVO Admin SDK) are available.
* Record conferences to MP3 - Voice conferencing recording (*.mp3 format)
* Cryptography (SSL for TCP data transmission and BLOW FISH for UDP data
and much more!
Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a teaching tool, live support, chat, meeting tool, a multiplayer game or any other type of application which requires users being able to talk, type messages and send files to each other. For conaito VoIP EVO clients to be able to interact with each other they must connect to a conaito VoIP EVO server which keeps track of where users are located and forwards voice-data packets between clients. The conaito VoIP EVO server is able to serve 1000 simultaneous clients and can be hosted in both LAN and internet environments.

VoIP SIP SDK for .NET and Win32 COM
$999 - conaito Technologies
VoIP SIP SDK - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.
Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications.
It accelerates the development of SIP/ RTP compliant soft phone with
a fully-customizable user interface and brand name. The conaito VoIP SIP SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.
Here is a list of the main features of the conaito VoIP SIP client:
* Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).
And much more! Try it today.

VoIP SIP SDK
$999 - conaito Technologies
VoIP SIP SDK - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.
Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications.
It accelerates the development of SIP/ RTP compliant soft phone with
a fully-customizable user interface and brand name. The conaito VoIP SIP SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.
Here is a list of the main features of the conaito VoIP SIP client:
* Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).
And much more! Try it today.
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