Search Results for "libamd.so.1"

Open Source - GPLv3 inc images. *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer browsing. ...

...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 reposurgeon. 2013-05-29: Maintainers of github fork "pyst2" contacted to join forces. Note that the last release here fixes the same bug as on github, otherwise github version seems to only contain cosmetic changes and examples. 2012-07-06: An offer to maintain the project has been received. 2012-01-29: There have been two or three forks that are being maintained. ...

pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/

آواسینتا ویرایشی اختصاصی شده از نرم افزار ویپ آستریسک می باشد که از زبان فارسی نیز پشتیبانی می نماید. این برنامه رایگان بوده و روی سیستم عامل اوبونتو ارائه می شود. برای دریافت اطلاعات بیشتر د ر مورد سیستم تلفن آواسینتا می توانید به صفحه اصلی آواسینتا در شبکه پلاس مراجعه نمایید https://samiyacode.ir/avasinta/

The program consist of an API (daemon) to handle a VOIP box of the type USB-B2K which allows to plug a classical telephone on an USB port (skypebox, wangobox, yealink...) in order to integrate this with any type of program of the type VOIP under GNU/LINUX. The functionalities are the following: 1. Switching to mode USB/PTSN 2. Detecting Pick up/ Hang up signal from the telephone 3. Generating Pick up tune 4. Generating Ring tone. 5. Handling telephone dial. This project is forked from https://savannah.nongnu.org/projects/usbb2k-api/ and https://sourceforge.net/projects/kb2kskype/

One of the first secure telephony programs. Not limited to VOIP but can also directly encrypt a telephone line by using modems. Calls are encrypted using Diffie-Hellmann key exchange and several symmetric ciphers. Download is possible from Germany only, see homepage

ss7mon will implement the following features for SIGTRAN and SS7 family of protocols: - Message sniffer. - Call flow tracing. - Traces DB. - Distributed tracing. - QoS monitoring. - KPI monitoring. - Web interface. - CLI command line interface. - ...etc.

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

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

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.

...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.

This is interface of itpp for python.

Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.

- ARIA Cipher Suite Support for SRTP

Quintum Tenor VoIP Gateway Call Detail Record (CDR) Advanced Server. This CDR Server can collect Call Detail Records (CDRs) from multiple Tenors. The data is collected simultaneously and continuously. Supported MySQL and file storages.

This an open source project for Paliha P750 and similar corded phones http://www.paliha.ru (only Russian language). The original project "Teleport-Paliha" was written for Windows OS, but I want the same on my Linux!

popz is a java implementation of the osi layers one and two over normal phone connections using frequency modulation techniques. This is really useful if you have to check your mail from the himalaya and have only your laptop and a mobile phone with you

CallFilter applies rules to incomming Phone Calls. Much like a network firewall.

OpenSMSC is ETSI GSM based realtime, distributed, highly available Short Message Service Center. It transfers short messages between mobile phones(MO and MT) over SS7 network and to applications(ESME) using open standard protocols like SMPP, HTTP & SOAP

rtusd is a foundation set of real time enabled User Space Devices/Drivers designed to link to HRT high resolution Timer controlled POSIX threads and link re-factored Linux hdw drivers for 2.6.18+. rtusd propogates Linux RT to user space generically.

Dgvchannel é um driver que permite a utilização das placas DigiVoice com o Asterisk(c). A Digivoice tem uma placa E1 (1 ou 2 spans) e uma placa FXO (4 ou 8 canais) que são suportadas por este channel. Acesse http://www.digivoice.com.br

eTalk is a modular and skinnable communication software for Enlightenment. Its written in C, EFL and has support for instant messaging, voice and video plugins.

This is a Java wrapper for Ffmpeg. The JMF (Java Media Framework) plugins system lets one use JMStudio or other application to play H.263/rfc2190 streams. Support for H.263+ and MPEG-4 is currently under test, please contact us for further information.

VVOIP is voice and video over IP is a P2P application without any protocol like SIP (no server is required) etc.. application. It use decent codecs for video and voice encrypted streams over IP. Moreover it is also comptiable with VoIP modems, you'd con