Search Results for "telegram-bot-api"
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21 projects for "telegram-bot-api" with 2 filters applied:
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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FreePBX (formerly Asterisk Management Portal) is a project to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with web-based administrative interface.
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Enterprise-grade ITSM, for every businessFreshservice is an intuitive, AI-powered platform that helps IT, operations, and business teams deliver exceptional service without the usual complexity. Automate repetitive tasks, resolve issues faster, and provide seamless support across the organization. From managing incidents and assets to driving smarter decisions, Freshservice makes it easy to stay efficient and scale with confidence.
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Twisted Protocols for communication with FreeSWITCH PySWITCH is a Twisted Python library that allows you to communicate with FreeSWITCH using inbound and outbound EventSocket connections.
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cdr-view
Webface witch extract CDR (Call Detail Records) from MySQL base.
Webface witch extract CDR (Call Detail Records ) directly from MySQL base. May be used as SOHO billing system. -
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cyber-wrt PBX
Cyber-wrt pbx is embedded system in DD-Wrt router
############################################### #--- CYBER-WRT by Christian Zeler / Comdif Telecom ---# #----------------- http://cyberhotel.comdif.com ------------------# ############################################## You can find 2015 news Cyber-hotel versions on http://cyberhotel.comdif.com including cyber-hotel on freepbx - Cyber-hotel decktop. . Cyber-wrt is a complet Asterisk IPBX and Hotspot system embedded in DD-Wrt router. Last version is build for ASUS RT-AC56U... -
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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
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VoIPGMap
Graph active VoIP calls on Google Maps.
A bash shell script which will read active call destinations in E.164 format from a source (such as Asterisk), then run the called number through some plain text and MySQL databases to find the best latitude/longitude match. It then creates a .html file with Google Maps API JavaScript code which will display a single call origination point and the active destinations (calls) on a Google Maps world map. -
Gemini 3 and 200+ AI Models on One PlatformBuild, govern, and optimize agents and models with Gemini Enterprise Agent Platform.
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This project aim to provide a web based application to plan Conference Call using Asterisk application MeetMe(). The engine is based on Vaadin language to improve the portability of the application, Hibernate and some AGI script based on PHP.
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FreePBX module for ENUM lookup API at enumplus.org.
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Open UC Framework (openucf) is a unified communication server which supports telephony, presence, instant messaging and contact management.
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[es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد
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OpenBTS
GSM L1-L3 stack with SIP network interfaces
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface. -
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SerMyAdmin
Web interface for OpenSIPS
I'm sad to say that, unfortunately, the serMyAdmin project is being discontinued starting in September 1st, 2010. After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I... -
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A web-based system that allows you to create groups of exten and generate reports of telephony usage, simple and easy to implement.
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An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.
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SipManager is a full-feature IP-Centrex PBX web app for hosted/multi-client PBX based on Asterisk and OpenSer. VoIP/ToIP platform (PBX, Fax, workflows, etc.). User-friendly web interface. Realtime architecture with SQL database. ASP. http://www.ovvoe.com
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FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.
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AS3 skype API for the Skype. Flash has access to the system Webcam, Audio and flash 9+ has all the capabilities to build a Socket communications. So we can have a Flash based skype client on the web. This can be a widget for instant chat & voice call.
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sippet
C++ SIP stack based on Chrome source code
...The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration.
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