Search Results for "command line"

SipTunnel is used to tunnel SIP UDP datagrams (and also RTP datagrams) through NAT using TCP protocol. It also can be used for many other purposes.

Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file.

VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.

Fix Skype is a python script and Win32 binary that is used to fix "future" timestamps in the Skype message database, such as those caused by the end of DST or by dual-booting a Linux system with time set to UTC from anywhere west of GMT. Essentially, you feed it your username and the number of hours by which some timestamps are in the future, and it subtracts that amount of time from any timestamp greater than "now".

Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib

Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.

What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.

VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K

Google Contacts to Grandstream xml Phonebook Format Converter

This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.

miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.

Acv is a PHP program which allows you to monitor Asterisk servers by viewing active calls, channels, and overall server performance numbers.