Best sip software for free download

HHD Accurate Network Monitor - ip packet sniffing, monitoring and protocol analyzer software tool that can sniff internet protocols performing tcp / ip, udp / ip communication traffic analysis. LAN connection sniffer and internet data capture utility
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MiaRec is a software for recording your conversations in any of Voip protocols (H.323, SIP, MGCP, Cisco SCCP).
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Globe7 is a free downloadable software integrated with Video phone, IM, & Real time video streaming, powered by online advertising. Globe7 enables you to make free calls from pc to pc, pc to mobile & pc to landline by watching videos in Globe
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MAX Phone is the latest VoIP application from MAIN Telecom, allowing users to enjoy rich real-time communications.
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Eliminate PBX headaches with 3CX Phone System for Windows!

Evolve your communications with 3CX Phone System for Windows - an office phone system/IP PBX that completely replaces your proprietary PBX, supports standard SIP soft/hard phones, VOIP services and traditional PSTN phone lines. 3CX Phone System is far less expensive than a traditional PBX and can reduce call costs substantially by using a VOIP service provider. Its web based administration makes phone system management easy. 3CX Phone System eliminates the phone wiring network and allows users to hot desk simply by taking their phone. Key Features:

* Complete phone system: Provides call switching, routing & queueing
* Purchase cost dramatically lower than a traditional hardware PBX
* Scaleable - Unlimited extensions and phone lines. No proprietary expansion modules needed!
* Web based configuration & status indication - Easy phone system management!
* Unified messaging: Receive voice mail via e-mail
* Auto attendant (e.g. 1 for sales, 2 for support etc.)
* Reduce long distance and inter office call costs
* No more expensive proprietary system phones: Use standard SIP phones
* Eliminate the phone wiring and make moving offices easier
* Out of the box configuration for many VOIP providers

Download the FREE edition with no time outs or limitations from: http://www.3cx.com/ip-pbx/index.html
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VaxVoIP SIP SDK provides tools and components to quickly add SIP (Session Initiation Protocol) based IP-Telephony make and receive phone calls feature in your web pages and software applications. It accelerates the development of SIP based soft phone with your own GUI (graphical user interface) and brand name.
It delivers superior voice quality by integrating advanced digital voice processing features including acoustic echo cancellation, noise cancellation and adaptive jitter buffering.
In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK.

- ACOUSTIC ECHO CANCELLATION OR SUPPRESSION
In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK. Hands-free or Internet telephony imposes several problems. The principal one is due to the coupling between loudspeaker and microphone. The loudspeaker signal is echoed back to the microphone and transmitted back to its origin. As a result the far-end participant perceives this as an echo.

- NOISE CANCELLATION OR SUPPRESSION:
VaxVoIP SIP SDK offers advanced Noise Cancellation technology that allows significant suppression of any background noise and provides high quality of output speech.

- ADAPTIVE JITTER BUFFER
Jitter buffers are used to smooth delay variations in received audio by buffering the packets and adjusting their rendering. The result is a smoother delivery of audio to the user.

- PACKET LOSS CONCEALMENT
Packet Loss Concealment (PLC) is a technique used to mask the effects of lost or discarded packets. PLC is generally effective only for small numbers of consecutive lost packets, for example a total of 20-30 milliseconds of speech, and for low packet loss rates.

- NAT AND FIREWALLS FRIENDLY
User can set SIP outbound proxy inorder to make and receive phone calls behind the NAT/firewall.
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Free VOIP / SIP phone. Use your PC as a phone. Works with any SIP / VOIP provider/ phone service. Use instead of the soft phone provided by your VOIP provider which locks you into their VOIP phone service.

Easy to use interface includes dial pad and buttons to transfer or forward a call. Set-up conference calls with a few mouse-clicks. Includes call log, phone-book, call status & statistics and more. Available in English, German, Spanish, French & Portuguese. Free download from http://3cx.com.

Features include

* Integrated phone book (allows you to assign caller ID to a name)
* Supports several SIP profiles
* Displays caller ID
* Individual ring tones (per caller)
* Drag & drop handling
* Call log
* Incoming calls: identify, accept, reject or ignore
* Record phone calls
* Multiple calls (HOLD)
* Terminal portability (TP)
* Call transfer (ECT)
* 3 party conference (3PTY)
* Call diversion (FORWARD)
* Call deflection (CD)
* Call completion (CCBS and CCNR)
* Message Waiting Indication (MWI)
* EarlyB3Connect (tones from trunk line)
* Keypad
* VoIP support for SIP connections
* Support G.711 (A-Law and u-Law), G 726, GSM, iLBC and Speex codecs.
* STUN support for NAT/firewall traversal.
* Search using ENUM
* Installation provided as MSI for easy deployment

3CX VOIP Phone for Windows also functions as a client for 3CX Phone System for Windows, a software-based VOIP PBX that completely replaces a traditional proprietary phone system. More information on http://www.3cx.com/phone-system/
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IVM is a telephone answering machine, voice mail, call attendant, info-line and interactive voice response (ivr) program for Windows computers with VoIP, voice modems or professional telephony line devices.
Just run the install file and IVM will be ready to answer calls. But you can also create a sophisticated telephone interactive response system. You will find many exciting ways to use IVM to provide information, serve your customers and save time.
Typical Applications
~ PSTN and VoIP Voicemail (from a single line answering machine to hundreds of voice mailboxes).
~ In-coming caller ID logger or display.
~ Call attendant (using the menu call transfer features) to transfer and direct calls.
~ Telephone information lines or AudioText type systems.
~ Automated telephone order taking.
~ Automated telephone surveys (using the number entry features linked to a database).
~ Credit card telephone account payment.
~ Dial in computer control or information (eg. Dial in to test security alarm or to restart server).
~ Many more telephone related applications.
Features
~ Multiple telephone line support (1 - 64 lines simultaneously).
~ Visual display for menu setup with drag and drop functionality.
~ Caller ID logging and optional on-screen caller ID flash display.
~ Automatic message forward by email or upload to internet.
~ Call transfer.
~ Remote Access
~ Supports VoIP connections using the SIP protocol.
~ Save in-coming messages as wav files .
~ Automatic hours feature.
~ Unlimited (selectable) out-going messages and menus.
~ Call simulator test your systems off-line.
~ Automated outbound calls and messages
~ Unlimited voice mail boxes
~ Text-to-Speech voice synthesis as an alternative to recording or importing wav or mp3 files.
~ Ability to open files or run other software to process data or report information. Free plugins offer internet and odbc sql connectivity, fax back services, hardware control and more.
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VOIP Phone Buddy for SKYPE

VOIP Phone Buddy for Skype adds automatic telephone dialling to almost any windows application, using SKYPE VOIP telephony, including Microsoft CRM, Microsoft, Access, Microsoft Outlook, Goldmine, ACT, other CRM and Accounts packages etc.

Simply move the cursor to a phone number and press the Hot Key to activate "VOIP Phone Buddy for Skype" phone dialer. The VOIP Phone Buddy for Skype and Jajah dialer then looks at the phone number, where the cursor was previously, and formats the international country code automatically. If the phone number does not contain the international country code then the user can select the code from their own Favourite countries in the Phone Buddy. The system also provides redial and Quick dial (favorite phone numbers) facilities too.

Automatic phone dialing from almost any Microsoft Windows Application!

When the call is selected Skype and Jajah will call you on your own phone and then dial the phone number you selected, using cheap or free phone call!

"Phone Buddy for SKYPE" features:
- Free 30 day trial ($19 per user)
- Install in minutes !
- Use with almost any windows software !!!
- Cheap national and international calls using SKYPE Voice Over IP
- Free calls to other Skype users
- Call History logged
- Integration with many of the top CRM and Accounts applications, such as Quickbooks, Microsoft CRM, Microsoft, Access, Microsoft Outlook, Goldmine, ACT, etc.
- Ideal for single users and small businesses


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Der Klassiker unter den IP-Telefonen

Das Grandstream BT-101 Telefon ist eines der besten und preiswertesten Internet-Telefone am Markt. Wir haben uns nach ausführlichen Tests für den Hersteller Grandstream entschieden, da dieses Telefon das beste Preis-Leistungsverhältnis bei hervorragender Funktionalität bietet.

Kinderleicht konfigurieren

Die Konfiguration erfolgt einfach und bequem über ein übersichtliches Web-Interface. Große Tasten machen die Bedienung auch für in Ihrer Wahrnehmung eingeschränkte Menschen kindeleicht und eröffnet so neue Zielgruppen für VoIP Technologie. Es verfügt über hervorragende Klangqualität und wird einfach an Ihren vorhanden Breitband-Internetzugang angeschlossen.

Voraussetzungen für den Betrieb

Für den Betrieb des BT-101 benötigen Sie einen Router an Ihrem Breitbandanschluss. Der Router muss dazu über einen 10 MBit-Ethernet-Anschluss verfügen (bei allen handelsüblichen Geräte vorhanden).
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One4000 SoftGlobalPhone, a Phone-in-the-Phone that lets you take advantage of free broadband VoIP calling and super low worldwide rates without changing your operator. 9 ways of calling from all of your phones, including PDA, cellulars and your PC. Download the 3D designer phone for your PC screen. Use it as a soft phone and to manage calls with your regular phones at super low rates worldwide from One4000 Telecom. Call from your PC, your cellular/mobile phone or home phone to any other PC, cellular/mobile or land phone in the world. It’s free calls and you can call to any PC with a soft phone, not just the ones with One4000 installed. PC to regular phones it’s not free, but rates are very low and you also get super low long distance and international rates if you want to call from your mobile or home phone to your friends all around the world. Besides, you get premium voice quality. Download and install automatically. Open up your One4000 SoftGlobalPhone, with five different skins, by clicking on the icon on the desktop. Choose from which one of your registered phone numbers you want to make your call by clicking on one of the buttons: Phone, GSM or PC. Enter the phone number you want to call by clicking on the dial pad (just the way you would have done with a real phone) and make your call. Alternatively you can use the address book included and just click on the name of the person you want to call. Then click on the Green Phone button. You will be called up and connected in seconds. You can also download a free small software program, One4000 MobileGlobalPhone, to your mobile/cell to initiate calls and get our super low rates. With this little software program in your mobile you can also choose which one of your registered phone numbers you want to be called up on. This software is directly downloaded into your mobile through an SMS if the phone supports Java and WMA. Take your service with you, where ever you go in the world.
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Adore SIP Softphone comes with an intuitive Graphical User Interface and can be easily customized. It's designed with SIP technology that can be used to make outbound phone calls from PC to any telephone. Since sip softphone uses Microsoft's g723.1 codec, it can be used in all versions of Microsoft Windows without the need of external codecs and gives excellent call quality even in dialup connection.

SIP dialer softphone delivers SIP-based communications and services for PC-to-Phone, Phone-to-PC and PC-to-PC services.
For your own SIP Softphone skin customization, please contact sales@adoreinfotech.com for pricing.

Features of SIP Softphone:
Customized Skin Interfaces
Call Timer
Display Balance
Call Forwarding
Call Transfer
Line Hold
Last Number Redial
Local Signalization (Dial tone, busy, ring back, etc.) for user comfort
Mute
Touch-Tones
Address Book
Microphone Volume Control
Speaker Volume Control


Benefits of SIP Softphone:



Small Application (less than 400KB)·
Use Microsoft g723.1 codec
Works with any Full-Duplex sound card
USB handset and headset support.
Works well on all versions of Microsoft Windows (Like 95, 98, 98SE, Millennium, NT4, 2000, XP, 2003)
NAT/Firewall support, stable SIP, RTP ports
Specify NAT IP to be written in SIP messages
Auto-Configuration of settings for easy deployment
STUN support for NAT detection and classification
Configuration Wizard
Uses New RFC 3261 compliant stack

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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 and websites. 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 suppression, noise suppression, reverb suppression and mute microphon

Key Features

* 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, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec)
* Encrypt SIP account settings (protect your SIP account settings in websites)
* Secure Weblicensing (protect your license in websites)
* Multi-User conference support
* Multi-line (simultaneous calls) support (Multiple Concurrent calls)
* Call Hold support
* Call Transfer support
* Instant text messaging (MIME) support and typing indication
* Mute microphone/speaker for each line
* DNS SRV resolution for SIP servers (RFC 3263)
* Stereo codec (L16)
* RTCP
* Auto-answer
* Do Not Disturb (DND)
* Adaptive jitter buffer
* Adaptive silence
* Advanced configurable digital voice processing features

...and much more. Try it today!
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SIP Phone DLL for .NET and ActiveX - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP Phone DLL 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 and websites. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.

The conaito SIP Phone DLL 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 suppression, noise suppression, reverb suppression and mute microphon

Key Features

* 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, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec)
* Encrypt SIP account settings (protect your SIP account settings in websites)
* Secure Weblicensing (protect your license in websites)
* Multi-User conference support
* Multi-line (simultaneous calls) support (Multiple Concurrent calls)
* Call Hold support
* Call Transfer support
* Instant text messaging (MIME) support and typing indication
* Mute microphone/speaker for each line
* DNS SRV resolution for SIP servers (RFC 3263)
* RTCP
* Auto-answer
* Do Not Disturb (DND)
* Adaptive jitter buffer
* Adaptive silence
* PLC (Packet Lost Concealment)
* AGC, VAD, AEC, Noise Concealment

...and much more. Try it today!
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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.
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Teleon Skype Gateway for voice modem is a VoIP gateway driver for common voice modem card. Calls can be forwarded from PSTN to Skype remote user and vice versa, in fix transfer mode or free dial mode.
Its freeware version has a duration limitation for each call.

Features:
1) Automatic installation/uninstallation program helps you to install, uninstall and remove driver.
2) Call from Skype to PSTN and PBX: open a chat window in Skype, type "atd xx..x", then initiate Skype call. Call will be forwarded by Skype gateway card to PSTN or PBX number xx...x.
3) Call from PSTN to Skype or SkypeOut: Call to the PSTN phone number where the gateway card is connected. Call will be answered after a number of rings, send out a prompt tone to inform you dial, dial the Skype remote speed dial number yy and # or SkypeOut PSTN number 011xx..xx#.
4) Support PIN calls. You can set PIN code to authenticate PSTN to SkypeOut calls, like phone card system.
5) Call from PSTN to PSTN is supported if both side PC installed Teleon gateway.
6) Fix transfer mode: Incoming Skype calls will be forwarded to a fixed local PSTN number set by you, e.g. your cellphone number, or a fixed PBX extension. PSTN to Skype fix transfer is also supported.
7) Easy configuration for call forward authentication, access blocking. Only those who are in your contact list are allowed to make forward calls. You can also simply block all forward capabilities if you want.
8) It works at background as a driver. There are rich balloon tip notifications inform you the call status.
9) Remote call trace tool to monitor the call flow in local PC or remote PC.

System Requirements:
Windows 2000/XP. CPU above 900Mhz, 256MB RAM and 20MB hard disk space.
Requires a voice modem with speakerphone function( you can purchase from www.teleon.net if you have no such a card), such as Conexant, Smartlink, PCtel chipset modems.
Skype version 1.3.0.57 up

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MAX IVR Studio Features:
Performance and Effectiveness:
* Call key select menus and ability for caller to enter numbers (data entry).
* Unlimited (selectable) out-going messages and menus.
* Fully programmable setup with examples.
* call transfer (subject to your company PBX features).
* call simulator is included so you can test your system off-line.</li>
* extended message play on key press including support for wav , GSM and a number of other ormats.
* Text-to-Speech voice synthesis as an alternative to recording or importing wav or GSM files.
* ability to access any database through ODBC.</li>
* support socket programming where can be sent/receive data to/from another PCs.</li>
* support broadcasting actions such as send Email, Fax and SMS.</li>
* ability to record message or record all the calls.
* support any arithmetic operations.
* ability to edit the windows registry.
* ability to import external files or run other software to process data or report information.
Reliability, Scalability, and Flexibility
* easy to use, graphical representation of common tape-deck controls, Pull-down
menus and extensive on-line help files that greatly
* reduce the need for costly user training
* visual displays of audio waveforms make it easy to see the effects of editing
* open development and editing environment provides choice and flexibility in
importing and exporting audio files in a variety of formats import/export
functions facilitate the interchange of audio files with other desktop system
using WAV and GSM audio files
* easy to export IVR tree file and its audio files in only on ZIP file and can import this ZIP file
* easy to change the nodes, links colors when design trees or in simulator
* easy to edit the properties of each node through the properties panel
* easy to create/delete any variable and change its value through the variables panel
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SIP SDK for .NET and ActiveX - 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 and websites. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.

The conaito 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 suppression, noise suppression, reverb suppression and mute microphon

Key Features

* 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, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec)
* Encrypt SIP account settings (protect your SIP account settings in websites)
* Secure Weblicensing (protect your license in websites)
* Multi-User conference support
* Multi-line (simultaneous calls) support (Multiple Concurrent calls)
* Call Hold support
* Call Transfer support
* Instant text messaging (MIME) support and typing indication
* Mute microphone/speaker for each line
* DNS SRV resolution for SIP servers (RFC 3263)
* RTCP
* Auto-answer
* Do Not Disturb (DND)
* Adaptive jitter buffer
* Adaptive silence
* PLC (Packet Lost Concealment)
* AGC, VAD, AEC, Noise Concealment

...and much more. Try it today!
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