Best sip software for free download

Functional empire dual-mode phone for a small price

The Pirelli DP-L10 combines cellular and Voice over IP and offers ease of handling. When a WLAN hotspot available, notify the Pirelli DP-L10 the availability of cost-effective and allows talks on Internet telephony. With numerous features, this dual-mode phone with a high quality and reliable device for the use of mobile Voice over IP at a low price. The Pirelli DP-L10 will SiPHome without cellular contract sold, and is thus all the customers of T-Mobile, Vodafone, E-Plus and O2 use.

Internet and mobile telephony in the highest voice quality

The Pirelli DP-L10 is a GSM-Triband-Handy and hence as In the United States usage. In WLAN-Modus supports the WLAN standard. This is the wireless connection to the wireless access point, depending on the configuration manual on the Verschlüssellungs- standards WEP, WPA and WPA2 secured. With the support of current VoIP voice codecs, the DP-L10 even at lower bandwidth highest voice quality and has Sprachqualitäts- features like CNG (Comfort Noise Generation), VAD (Voice Activity Detection) and echo suppression.

Numerous features

In addition to the many features, such as the phone book for 500 entries, contact synchronization with Outlook, SMS, MMS, WAP 2.0 and e-mail functionality, 25 polyphonic ringing tones and an integrated VGA camera, meets the Pirelli DP-L10 all current high standards of the current mobile generation.

Power and comfortable call termination

The battery provides a comfortable talktime of up to 4 hours, and a standby time of up to 180 hours. Before each interview, the user via radio button to choose which Telefonart he wants to make phone calls. Moreover, even GSM-Telefonate executed whe

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Implement a professional voice response system into your business quickly and efficiently. CT Developer Studio helps you develop and host automated call handling systems which can be tailored to individual needs. Develop complex IVR software or implement a simple inbound call center, set up automatic dialing or create a fully automatic call management system in Windows. Setting up an IVR system becomes fast and straightforward using CT Developer Studio.

Get started in minutes using one of the many supplied demonstration templates. Quickly modify the details to suit your own requirements thanks to CT Developer Studio graphical interface. Drag and drop components and draw connections with a mouse onto a flow chart to specify the route for your calls. This visual development approach closely mimics the "What You See Is What You Get" approach of modern text processors, allowing you to model your resulting voice application in a straightforward way. SQL database support ensures fast and error-free operation under heavy loads even in the busiest environments. CT Developer Studio is designed to handle hundreds of phone lines and concurrent phone calls on a regular Windows PC.

The intuitive graphical IVR designer can create your IVR system in a matter of minutes, and the rich functionality and the ease with which CT Developer Studio can be customized provides you with the means of creating the call handling system you want. CT Developer Studio intuitive, icon-based interface removes the complexity of developing advanced call centers, IVR scripts and call routing applications by seamlessly integrating telephony and computers. Complex functions are built into simple graphical icons.

CT Developer Studio is an ideal tool for developing, hosting and managing advanced voice applications. CT Developer Studio supports a wide variety of telephony cards such as Intel/Dialogic, NMS, and Ai-Logix.

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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)
* NEW in v2.4! Encrypt SIP account settings (protect your SIP account settings in websites)
* NEW in v2.3! Secure Weblicensing (protect your license in websites)
* NEW in v2.2! 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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Quorum is software which runs as a phone conference server on any Windows PC. Callers dial into the server to join telephone conferences.
Callers can connect to a conference call using either standard or VoIP lines on external telephone numbers or through an office phone system. The number of people who can join a conference call is limited only by your bandwidth.

When the caller selects the option to create a new conference Quorum will assign and read back the unique and secure conference number. The conference creator then gives this number to each conference participant. Whenever any person dials the server and enters this conference number, they will join in the current conversation with the other members of the conference.

Features:

~ Runs on almost any Windows PC.
~ Handles unlimited simultaneous conferences. The number of calls and conferences is limited only by your bandwidth.
~ Can connect via VoIP using the international standard SIP protocol. If VoIP is used no special hardware is needed.
~ Can connect to ordinary phone lines (or analog PBX extensions) directly using telephony devices.
~ Can connect to one Skype Number (so a Skype user can connect) using the Uplink Skype to SIP Adapter.
~ Easily allocate new conference IDs on the phone.
~ Join any pre-defined conferences by using the allocated conference number.
~ Voice prompts are easily changed via the user interface.
~ VoIP calls can be made using free VoIP SIP Service Providers saving call costs if they are an issue.
~ Installs and ready to run conferences in just minutes.
~ Very simple user interface for day-to-day easy business use.

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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.
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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MAX Phone is expanding the soft phone experience by making it even easier for users to make VoIP (Voice over Internet Protocol) and Video over IP calls.
MAX Phone supports SIP industry standard, and is fully inter-operable with any RFC SIP 3261 compliant servers.

MAX Phone delivers SIP-based communications and services for PC-to-Phone, Phone-to-PC and PC-to-PC services.

Standard Telephone Features:

MAIN Telecom Multimedia Communicator MAX Phone has all the standard telephone features, including:

* 4 lines with ability for extension.
* Caller ID.
* Call rejection.
* Call waiting.
* Call forward & blind transfer.
* Hold.
* Call record using ASTERISK engine.
* Microphone and Speaker volume control.
* Mute Microphone.
* Redial, Auto answer & Do not disturb.
* Direct IP to IP Calling.
* Video call.

Enhanced Features and Functions:

MAX Phone also supports the following VoIP features and functions:
* Import contacts list from comma-separated values (CSV) files, CSV standard is supported by Microsoft Outlook, Microsoft Excel ...etc.
* Export contacts list to comma-separated values (CSV) files.
* Separate Calls history for each registered user.
* Separate Contacts list for each registered user.
* Audio & Video tuning wizard.
* Acoustic echo cancellation, redundant audio coding, dynamic jitter buffer and adjustment, automatic gain control, voice activity detection.
* Support for the following audio codecs: G.711, G.722.1, G.723, GSM, DVI4 and SIREN
* Support for the following video codecs: H.261 and H.263.
* Automatic selection of the best codec based on the other party’s capability, available bandwidth, and network conditions, MAX Phone switches the codec within a call in response to changing network conditions.
* UPnP-enabled NAT traversal.
* Support for DTMF.
* Live Update.
* Private Labeling and User Interface customization.

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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 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 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 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)
* NEW in v2.4! Encrypt SIP account settings (protect your SIP account settings in websites)
* NEW in v2.3! Secure Weblicensing (protect your license in websites)
* NEW in v2.2! 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 DLL - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP 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.
It accelerates the development of SIP/ RTP compliant soft phone with
a fully-customizable user interface and brand name. The SIP 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 cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here is a list of the main features of the conaito SIP DLL:
* 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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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)
* NEW in v2.3! Secure Weblicensing (protect your license in websites)
* NEW in v2.2! 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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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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The Merak Mail Server comes with all the optional modules which can be evaluated after a 5 minute installation. You have the possibility to backup and restore your server configuration including all settings and accounts with a single click. Merak Mail Server is compatible with all servers supporting the Internet standards. Merak Mail Server supports all cutting-edge security features including anti-virus protection, anti-spam engine and open relay protection options.

- SMTP / POP3 / IMAP4 / SIP / HTTP / FTP / LDAP / Jabber / Proxy
- SNMP
- SSL (128-bit encryption for all protocols)
- Compatible with Microsoft Windows and various main stream Linux distiributions
- Outlook & Outlook Express, Eudora, Bat! and majority of other mail clients compatible
- Multiple Domains & Users
- Mailing Lists & List Servers
- Groups
- Remote Accounts
- Executables
- Catalogs
- Notifications

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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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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

Download the FREE edition with no time outs or limitations from: http://www.3cx.com/ip-pbx/index.html

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CommView for WiFi is a special edition of CommView designed for capturing and analyzing network packets on wireless 802.11a/b/g networks. CommView for WiFi gathers information from the wireless adapter and decodes the analyzed data.With CommView for WiFi you can see the list of network connections and vital IP statistics and examine individual packets. Packets can be decrypted utilizing user-defined WEP or WPA-PSK keys and are decoded down to the lowest layer, with full analysis of the most widespread protocols. Full access to raw data is also provided. Captured packets can be saved to log files for future analysis. A flexible system of filters makes it possible to drop unnecessary packets or capture the essential packets. Configurable alarms can notify the user about important events such as suspicious packets, high bandwidth utilization, or unknown addresses. Packet Generator utility is available for editing and sending packets via your wireless network adapter.CommView for WiFi features full decoding of the following protocols: ARP, BCAST, BGP, BMP, CDP, DAYTIME, DDNS, DHCP, DIAG, DNS, EIGRP, FTP, G.723, GRE, H.225, H.261, H.263, H.323, HTTP, HTTPS, ICMP, ICQ, IGMP, IGRP, IMAP, IPsec, IPv4, IPv6, IPX, HSRP, LDAP, MS SQL, NCP, NDS, NetBIOS, NFS, NLSP, NNTP, NTP, OSPF, POP3, PPP, PPPoE, RARP, RADIUS, RDP, RIP, RIPX, RMCP, RPC, RSVP, RTP, RTCP, RTSP, SAP, SER, SIP, SMB, SMTP, SNA, SNMP, SNTP, SOCKS, SPX, SSH, TCP, TELNET, TFTP, TIME, TLS, UDP, VTP, WAP, WDOG, YMSG, 802.1Q, 802.1X. CommView for WiFi is a helpful tool for WLAN administrators, security professionals, network programmers, or anyone who wants to see the full picture of their WLAN traffic. This application runs under Windows 2000/XP/2003 and requires a compatible wireless network adapter.

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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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CommView is a program for monitoring Internet and Local Area Network (LAN) activity capable of capturing and analyzing network packets. It gathers information about data passing through your dial-up connection or Ethernet card and decodes the analyzed data. With CommView you can see the list of network connections and vital IP statistics and examine individual packets. Packets are decoded down to the lowest layer with full analysis of the most widespread protocols. Full access to raw data is also provided. Captured packets can be saved to log files for future analysis. A flexible system of filters makes it possible to drop packets you don't need or capture only those packets that you wish to capture. Configurable alarms can notify you about important events, such as suspicious packets, high bandwidth utilization, or unknown addresses. CommView is a helpful tool for LAN administrators, security professionals, network programmers, or anyone who wants to have a full picture of the traffic going through one's PC or LAN segment. This application is designed for Internet users and small and medium sized networks and can run on any Windows 98/Me/2000/XP/2003 system or Windows XP 64-bit Edition on AMD Opteron and Athlon64 processors. It requires an Ethernet, Wireless Ethernet, or Token Ring network card supporting the NDIS 3.0 driver standard, or a standard dial-up adapter. CommView features full decoding of the following protocols: ARP, BCAST, BGP, BMP, CDP, DAYTIME, DDNS, DHCP, DIAG, DNS, EIGRP, FTP, G.723, GRE, H.225, H.261, H.263, H.323, HTTP, HTTPS, ICMP, ICQ, IGMP, IGRP, IMAP, IPsec, IPv4, IPv6, IPX, HSRP, LDAP, MS SQL, NCP, NDS, NetBIOS, NFS, NLSP, NNTP, NTP, OSPF, POP3, PPP, PPPoE, RARP, RADIUS, RDP, RIP, RIPX, RMCP, RPC, RSVP, RTP, RTCP, RTSP, SAP, SER, SIP, SMB, SMTP, SNA, SNMP, SNTP, SOCKS, SPX, SSH, TCP, TELNET, TFTP, TIME, TLS, UDP, VTP, WAP, WDOG, YMSG, 802.1Q, 802.1X.

Related software: network, monitoring, sniffer, capturing, traffic, packet, …

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


Related software: gateway, voip, skype, voice, pstn, phone, modem, call, driver, sip, h323, forward, pci, fxo, …