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System :: Communications :: Telephony

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Wammu

 Mobile phone manager.
Wammu is mobile phone manager running on Linux, Windows and possibly other platforms, where Gammu and wxPython works. The communication is made by Gammu library. Wammu main features are: complete support for contacts, todo, calendar; read/create/save/send/backup smses; sms composer for multi part smses; display smses including pictures and ringtones playback; support for backup and import; export ...
Licence : GPL


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Twinkle
 Softphone for your voice over IP and instant messaging communications.
Twinkle is a softphone for your voice over IP and instant messaging communications using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages. Twinkle supports basic presence functionality using a presence agent that must be offered by your SIP provider. Only the basic online and offline presence states ...
Licence : GPL


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Phone Manager
 Control aspects of your mobile phone.
Phone Manager is a program created to allow you to control aspects of your mobile phone from your GNOME 2 desktop. . Current Phone Manager features include: Runs in the background; indicates status on the panel notification area. Display on-screen alert when text message (SMS) arrives. Text message (SMS) sending facility. Evolution Addressbook integration when sending messages.
Licence : GPL


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Linphone
 Linphone is an audio video phone with gnome interface.
Linphone is full featured SIP softphone making people having free VoIP calls with voice, video and chat over the internet. It supports several codecs (speex, gsm, ilbc, g711, h263-1998, mpeg4, theora). It is able to use and register with SIP proxies, gateways or registrar. It also exists as a small console application called linphonec and a beta release is availlable for windows XP.
Licence : GPL


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Yate
 Yet Another Telephony Engine.
Yate is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. Yate is written in C++ and supports ...
Licence : GPL


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Trixbox
 Telephony application platform.
Trixbox is a telephony application platform based on Asterisk Open Source PBX Software. The trixbox application platform makes it fast and simple to install and configure your business-class telephony system. With unlimited extensions, voicemail-to-email, music on hold, call parking, analog lines or high density T1/E1 circuits and many other features, trixbox provides all the functionality your ...
Licence : Open Source


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I Hear U
 Voice over IP (VoIP) application.
HU is a Voice over IP (VoIP) application for Linux (using Qt), that creates an audio stream between two computers easily and with the minimal traffic on the network. The main features are: Peer-to-Peer: the communication takes place directly between the computers (UDP and TCP both supported), without need of session protocols (such as SIP or H323) or other servers in the middle. Good audio ...
Licence : GPL


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BitPim
 View and manipulate data on many CDMA phones.
BitPim is a program that allows you to view and manipulate data on many CDMA phones from LG, Samsung, Sanyo and other manufacturers. This includes the PhoneBook, Calendar, WallPapers, RingTones (functionality varies by phone) and the Filesystem for most Qualcomm CDMA chipset based phones.
Licence : GPL


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Sofia-SIP
 Open-source SIP User-Agent library.
Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP implementation follows RFC3261 and related key RFCs. ...
Licence : GPL


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OpenSS7
 SS7/SIGTRAN/IDSN/VoIP STREAMS Comunications Protocols.
The OpenSS7 Project provides the Linux Fast-STREAMS SVR 4.2 compatible STREAMS framework for Linux and a wide range of Telecommunications Protocols utilizing the STREAMS framework: SS7 (Signalling System No. 7), SIGTRAN (Signalling Transport), ISDN (Integrated Services Digital Network), VoIP (Voice over IP), GSM. The aim of the project is a complete open-source SS7 stack and related telephony ...
Licence : GPL


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Voix Manager
 Asterisk management Switchboard Application.
Voix Manager Beta 2.2 has been released, we have added a new template so is now possible to organize the extensions in groups, for every group is possible turn on or off the visibility. We have changed the click behavior on the extensions from double click to one click, this is useful for touchscreen users. More improvements in the graphic layout and in the tootips, and fixed some bug. VOIX ...
Licence : Commercial


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OpenSER
 Mature and flexible open source SIP server.
OpenSER is a mature and flexible open source SIP server (RFC3261). It can be used on systems with limitted resources as well as on carrier grade servers, scaling to up to thousands call setups per second. It is written in pure C for Unix/Linux-like systems with architecture specific optimizations to offer high performances. It is customizable, being able to feature as fast load balancer; SIP ...
Licence : GPL


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