h.323
Vulnerability in Microsoft Internet Security and Acceleration Server 2000 H.323 Filter Could Allow Remote Code Execution (816458) 1.0
Vulnerability in Microsoft Internet Security and Acceleration Server 2000 H.323 Filter Could Allow Remote Code Execution (816458) is created to allow an attacker to overflow a buffer in the Microsoft Firewall Service in Microsoft Internet Security and Acceleration Server 2000. more>>
Vulnerability in Microsoft Internet Security and Acceleration Server 2000 H.323 Filter Could Allow Remote Code Execution (816458) 1.0 is created to allow an attacker to overflow a buffer in the Microsoft Firewall Service in Microsoft Internet Security and Acceleration Server 2000. An attacker who successfully exploited this vulnerability could try to run code of their choice in the security context of the Microsoft Firewall Service. This would give the attacker complete control over the system.
The H.323 filter is enabled by default on servers running ISA Server 2000 computers that are installed in integrated or firewall mode.
Mitigating factors:
- ISA Servers running in cache mode are not vulnerable because the Microsoft Firewall Service is disabled by default
- Users can prevent the risk of attack by disabling the H.323 filter
H323 Caller 1.0
H323 Caller 1.0 is a useful tool that allows you to connect to a H.323 voice or VoIP gateway and dial a certain number (optionally with text information to be displayed at the addressed phone) more>>
H323 Caller 1.0 is a useful tool that allows you to connect to a H.323 voice or VoIP gateway and dial a certain number (optionally with text information to be displayed at the addressed phone).
Since H323Call is a command line program it can be used with VBScript, DOS and other script languages to easily create applications like alarm signaling, server problem notification and many others.
Major Features:
- Connect to a H.323 voice or VoIP gateway and dial a certain number.
- Command line program that can be used with VBScript, DOS and other script languages
- Create applications like alarm signalling, server problem notification and many others.
- Very easy to use.
Requirements: Microsoft Windows 2000 or higher
Phone Patch 1.3b5
H.323 web phone proxy more>> Most web phone programs must be directly connected to the internet to operate correctly. Phone Patch allows H.323 compliant web phone programs to operate from a private network to the net, even if your computer is not directly connected.<<less
SJphone 1.60.289a
SJphone is a VOIP softphone that allows you to speak with any other softphone more>>
It supports both SIP and H.323 standards and is fully inter-operable with most major VOIP vendors and Service Providers

PicoPhone 1.64
A simple Internet phone application with chat more>>
I made it, because all the Internet phones around are quite heavy and use complicated protocols, such as H.323, which has problems with NAT routers. PicoPhone uses a simple UDP-based protocol, which works very well with NAT. The program accepts connections on UDP port 11676 and makes connection to that port.
Optionally a port number can follow the address (the colon is used as a separator), if the connection has to be made to another port (the NAT router on the receiving side should convert the port number to the default 11676).
PicoPhone allows multiple concurrent calls, but the audio device should allow multiple output streams to be opened simultaneously. Conferences are not set up automatically, i.e., if A calls B and C, B and C wont hear each other: B has to call C, or C has to call B for the complete conference to be set up.
On incoming calls the program plays the file ringin.wav, which should be present in the default windows location for wav files or in the same directory of PicoPhone.
Operation
Enter a fully qualified domain name or an IP address and then press the Call key. You can put a comment before the address, but in this case you should enclose the address between < and >. If not specified, the address uses the standard PicoPhone port, 11676. A port can be specified after the address, using a colon as a separator. The syntax can be therefore:
address, or
address:port, or
name , or
name
When the second two forms are used, and also the first two forms, when the address is a DNS name and not an IP address, the address will be saved and it will be available from the drop down box for the next time.
Firefly Softphone 3.2 Build 5774
pc to pc calling & voice mail more>> Firefly, bridges the barriers between instant messaging and VoIP. FireFly adds to its offering an upcoming direct bridge to standard telephone lines with very low prepaid calling-rates anywhere in the world, and (also forthcoming) full interoperability with other VoIP networks and with the H.323, MGCP, SIP protocols.<<less

GNU Gatekeeper (GnuGk) 2.3.1
A free H.323 gatekeeper to build VoIP networks, route calls and do authentication and more>>
A free H.323 gatekeeper to build VoIP networks, route calls and do authentication and accounting.<<less
A free H.323 gatekeeper to build VoIP networks, route calls and do authentication and accounting. GNU Gatekeeper (GnuGk) 2.3.0 - GNULicense:Freeware
License:Freeware
License:Freeware
Mizu VOIP Server 3.2.14
The Mizu VOIP Server is an all in one enterprise grade soft switch solution with support to SIP and H.323. All Class5 features including sophisticated routing, access rules, CDR record generation, billing and accounting. more>> <<less

JoinPhone Lite 1.67
You can see the world! P2P video capture & phone JoinPhone Lite! more>>
Yate 1.1
Yate is a next-generation telephony engine more>>
Voice, video, data and instant messaging can all be unified under Yates flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
Yate can be used as a:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and client
- IP Telephony server and client
- Call center server
- IVR engine
- Prepaid and postpaid cards system
PC-Tone Softphone 1.01
A configurable H.323 softphone which supports NAT behind firewall. A selectable service providers gatekeeper IP and Codec at users choice. more>>
PC to PC, PC to Phone, Phone to PC
H323 compliant
Touch-tones [DTMF]
Dial/ Redial/Hangup
Dynamic CODEC Selection
Caller ID
Call Timer
Backspace/Clear/Delete
Microphone & Speakers Levels
Last Caller-IDs
Recent Calls Dialed
Recent Calls Received
Sound Device Selection
New Familiar-looking Menu
Phonebook
Speed Dial
Benefits:
Easy to install
Intuitive user interface & Menu.
Easy to navigate and configure.
G.723/729/GSM codecs included
NAT/Firewall support
Multiple gatekeeper selection
Customised skin can be provided by Author
Marratech for Windows 6.1
group videoconferencing more>> Marratech enables secure, multi platform group collaboration over the Internet.
It combines the markets best voice (VOIP) quality, a full featured whiteboard, multiparty video and instant messaging.
It is available for Mac OS X, Windows and Linux. It is compatible with SIP gateways and phones and with H.323 end points.
Test the software for free by using the public servers listed at http://www.marratech.com/meetings/ or download a Marratech Free server at http://www.marratech.com/free/<<less
GnuGk 2.2.7
It forms the basis for a free IP telephony system (VOIP). more>> It forms the basis for a free IP telephony system (VOIP).
The GNU Gatekeeper (GnuGk) was designed to be a full featured H.323 gatekeeper. It forms the basis for a free IP telephony system (VOIP).
Main features:
- can be run as a Windows service
- accounting and call authorization via SQL database, Radius, file or external application
- flexible call routing
- number rewriting (calling and called)
- support for NAT traversal
- full H.323 proxy
- TCP interface to applications
- CTI functions (eg. VOIP call-center, call transfers)
- gatekeeper clustering support (neighbors, parent/child, alternates)
- H.235 security
- graphical user interface
- its free, including source code

ManageEngine VQManager 6.2
ManageEngine? VQManager is a powerful, web-based, passive VoIP monitoring software tool. VQManager monitors real-time VoIP QoS metrics to give instantaneous reports, self-configurable alerts & in-depth fault probing ? all from a single web interface. more>>
ManageEngine? VQManager is a powerful, web-based, passive QoS monitoring tool for VoIP networks. It can monitor any device that supports SIP, H.323, Cisco? Skinny and RTP / RTCP. VQManager monitors real-time VoIP QoS metrics to give instantaneous reports, self-configurable alerts & in-depth fault probing ? all from a single web interface.
How does VQManager help keep your VoIP infrastructure up and optimized?
- Real-time, 24x7 non-intrusive monitoring of various QoS metrics allows speedy identification of VoIP faults and quality deterioration.
- VQManager pro-actively notifies faults by means of operator alerts, Email notifications and SNMP traps.
- Complete call level and end-point level 24x7 QoS analysis makes possible in-depth fault probing through call traces and call summaries.
- VQManager?s supports monitoring of any SIP, H.323 or Cisco? Skinny device. Also, by processing Cisco (4, 4.x, 5), Swyx 6.0, Vovida Vocal, Tekelec 5.6, Shoretel 6.1, PortaOne and Asterisk CDRs, VQManager keeps the administrator updated with the happenings in the VoIP network regardless of the devices used.
- The tool?s powerful and flexible reporting generates over 30 different reports giving insight into usage trends, bandwidth utilization & capacity planning.
VQManager requires no other special hardware/software for support and its remotely accessible, user-friendly interface saves precious time & greatly improves administrator productivity. With minimal to zero deployment, VQManager is your fastest and most cost-effective way of ensuring an optimized VoIP communication infrastructure.
NOTE: Downloaded software would have support for monitoring unlimited IP phones for the first 30 days of usage. On completion of 30 days, support is reduced to monitoring of 10 IP phones.
System Requirements: Pentium 1Ghz and above, 1 GB RAM
Enhancements: H.323 support, Concurrent calls reporting, Configuration wizard for easy set-up, Voice Quality graphs enhanced with drill-down options, Full support for Cisco CallManager 6.x environments
<<lessitalk2u 1.0
Free PC to PC calls to and from anywhere in the world more>>
The software and PC to PC calls are free as well as some other features:
Built in Instant Messaging
User online status (Real Time)
User profile and search
File Transfer
Free PC 2 PC Calls
Caller "On Hold"
No Firewall to Firewall setup
No Spyware or Adware
H.323
Web Click 2 Call
And Much More...
iTALK2u is quick and easy to install. Just download it, register, and within seconds you can plug in your PC headset and call your friends on iTALK2u. iTALK2u calls have the best sound quality and are the most secure with end-to-end encryption. Best of all, iTALK2u does not require you to reconfigure your firewall or router.
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