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VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database.

Pyst consists of a set of interfaces and libraries to allow programming of Asterisk from python. The library currently supports AGI, AMI, and the parsing of Asterisk configuration files. The library also includes debugging facilities for AGI. 2014-04-17: Moved the version control to GIT. To check out see the tab "Code". Note that the whole history including ancient CVS, then some time in monotone, then subversion was united into one GIT repository thanks to ESR's...

Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp

The OSP Toolkit is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.

Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip

This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c

KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org

LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git

The Motorola XT881 is a 3G Mobile phone based on Android 4.0.4

This project is for people who want to access one or more HFC-S PCI A based ISDN BRI cards over the DAHDI interface and maybe want to use other DAHDI hardware in parallel, too. DAHDI driven hardware can be used by Asterisk, FreeSWITCH, Yate and other telephony software. The HFC-S PCI A based cards are very cheap and thus ideal for private BRI users. The OSLEC is being developed since a long time ago and said to be very good. The superiority of "hardware echo cancellers" is a myth. This...

*** MOVED ON GOOGLE CODE ***

Java version of Dialogic's SwitchKit API. This API is actually a more "Java Centric" API, with listeners, filters, and a very extensible and modular framework that finally uses JNI to access the native libraries provided by Dialogic.

BladeWareVXML is a portable VoiceXML 2.1 interpreter that is an enhanced version (performance, usability and integration) of OpenVXI. A commercial version, with documentation, sample code, and support options, is available from the Commetrex Website.

Welcome to the Linux-IrDA project. The overall goal of this project is to make an implementation of the IrDA (tm) standards specifications for the Linux kernel. The code is licensed under the GNU Public licence (GPL) and is now included in Linux-2.2.

BeepagePHP provides a PHP-based web client to the beepage client/server paging system. Beepage is a Unix-based, Internet-aware text paging system that implements the Telocator Alphanumeric Protocol (TAP). Source code is hosted at http://git.io/rgikgg

An open project for a number of activex controls for use in the Windows environment. The controls have been sold commerical before by Eurosource (http://www.eurosource.se) but are now free complete with source code.

An Open H.323 based channel driver for the the Asterisk PBX. This code has now been intergrated into the Asterisk source tree. Please see the Asterisk source tree for any future updates of this software.

PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide

OpenSnom aims to implement a Fully Free Firmware for the 150Mhz MIPS (INCA-IP - PSB 21553 E) Based SIP Phone Series (320/360/370) The Phone is based on a Embedded Debian with a lot of additional proprietary Software. Until approvement the project is host