Search Results for "gnu-efi" - Page 3

dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.

P3nfs makes the drives of Symbian/Epoc devices available under Unix. Its name comes from the Psion Series 3a, which was the first supported device. Current devices include Nokia and Sony ericsson phones.

PartiSIPation is a sip user agent which has an easy replaceable gui. So it can be used as a softphone as well as it can be integrated in any kind of application, e.g. online games.

An extension to the standard Asterisk PBX expanding it's capabilities to include messaging.

OSP Module enables SER (SIP Express Router) to use the Open Settlement Protocol for secure Authorization, Routing and Accounting. OSP is an ETSI protocol, Technical Specification 101 321

Siemens EWSD PSTN Exchange can dump its configuration into regenfiles (or commandfile, cmdfile). rg2xml interprets these files and transforms information to XML

This projects intend to port to Windows / Interix the famous and performant SIP Express Router from IPTEL (SER)

Very fast and very flexible SIP server

GSM Multiplexer Daemon is intended to be used for GSM Modens. It implements (a subset of) the GSM 07.10 standard, it can be used to multiplex one or more logical channels over one physical serial channel to the modem.

Asterisk based SIP b2bua realization. It's patched asterisk and some AGI script. Features: full vovida's b2bua radius emulation, extended radius attributes, radius failover, LCR, Call failover, Codec based routing and other useful things.

Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home/corporate/operator level markets.

sipsak is a command line tool which can send simple requests to a SIP server. It can run additional tests on a SIP server which are usefull for admins and developers of SIP enviroments. Note: the project is maintained on GitHub https://github.com/nils-ohlmeier/sipsak

The Milkfish is an Embedded Communications Software Project. Main objective is to provide a basic NAT traversal solution for cheap hardware enabling a practical setup of multiple SIP phones in a LAN sharing a WAN connection with one public IP address.

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...

Video Applications for Asterisk (libh43m, app_h324m, app_mp4, app_rtsp, app_transcode, app_swf) H324m stack, mp4 record and playblack, rtsp support, video transcoding and swf file playback.

A Load Balancer for a VoIP software on a Mobile Node device that choose among three net interfaces to send packages to an other Load Balancer on a Fixed Node. The LB's aim is to lose the lowest number of packages, choosing the best net interface.

Maintained version of old asterisk applications ported to 1.4 that for copyright reasons cannot be included into official Digium's releases and that for some reason the author is not keeping up-to-date. The project also require Zaptel instead of DAHDI.

SIP over IPv4 and SIP over IPv6 are different worlds, making it impossible to register IPv4-only phones on IPv6-only services, and so on. SIPproxy64 is a proxy that translates between the two worlds. It is small enough to run inside a router.

CyanogenMod 7 for Samsung Galaxy S Series Phones Sources: http://github.com/CyanogenMod

Remotely turns Grandstream GXV3000 video phone into a spy camera.

This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.

This is interface of itpp for python.

A $10 Analog Telephone Adapter (ATA) that allows low cost routers such as the WRT54G to connect to an analog telephone. The application is low cost VOIP, in particular for the developing world.

This project is aim to develop module named app_avi.so for asterisk to add the ability of play avi file.