Search Results for "mega-voice"
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Showing 92 open source projects for "mega-voice"
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FusionPBX
A domain based multi-tenant PBX & voice switch for FreeSWITCH
FusionPBX is a multi-tenant PBX and voice switch for FreeSWITCH, a highly scalable, multi-threaded, cross-platform communication platform. FusionPBX can be used either as a single or domain based PBX, and offers several functionalities: as carrier grade switch, call center server, voicemail server, fax server, conference server, VoIP server, multi-tenant appliance framework and so much more. -
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BlackBelt WASTE- ipv4 Tor i2p AI Voice
Modern, AI-Smart, WASTE p2p for ipv4, Tor and i2p + Voice Conference.
Open Source - GPLv3 inc images. A WASTE client. Download and create your own WASTE networks. Move 1000's of GB's at 100MB+ per sec. (800 Mbits per sec) FULL pause and resume capable. Voice Conference, Chat, Transfer files and Participate in Forums in a secure environment. For Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win 10, Win 11, Linux (WINE). *** User Based Access Control - for voice, chats, file transfers and uploads. (useful in NULLNETS) *** Distributed Autonomic-Performance-Tuning - A Goal-Seeking Swarming-Semiotic AI *** AI Connect - AI Managed Connections. *** Self-Organising Anti-Spoofing Technology *** Geared Multi-threading, providing the smoothest performance possible *** Advanced Threat Detect and Manage Technology *** Voice Conferencing Over WASTE *** RNN - Recurring Neural Net - AI Noise Reduction *** Differential Files Transfer - Seriously fast data backups -
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VOIP-VOICE-TO-TEXT&ANALYS
Convert VoIP calls to text and analyze them with AI
The VoIP voice-to-text software for Issabel is an intelligent, AI-based solution that converts calls into accurate Persian text. After each call, the audio file is sent to the GPT-4O AI engine, producing editable transcripts. The software also provides AI-powered call analysis, extracting key points, customer requests, satisfaction levels, and sensitive topics, all stored in the database. -
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ictdialer
Open source auto dialer for voice, fax and SMS campaigns
An open source auto dialer and broadcasting platform built on FreeSWITCH and the ICTCore framework. It runs outbound campaigns across voice, fax and SMS from one interface. Campaigns are multi-tenant, so a service provider can give customers their own logins, contacts and reporting without running separate installations. IVR flows are built with a drag and drop designer rather than written by hand, which matters when the person designing a survey or a press-one campaign is not the person who administers the switch. ... -
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Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
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Longing for the good old days of Asterisk@Home? Welcome back to the steroid-enhanced version. PBX in a Flash 3.0 & Incredible PBX 2020/2021/2022/2027 are the latest Lean, Mean Asterisk Machines, high-performance, turnkey Asterisk PBXs that are easy to upgrade. Features include Rocky8, CentOS/SL 7.x, Ubuntu 22.04 & 20.04, Debian 10 and Raspbian 10 support with Asterisk 20/18/16 and FreePBX 16/15 GPL modules. Add-ons include one-click installs of Incredible Fax and many other Asterisk...
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JavaComm
Desktop communication tool: chat, telephony, multimedia
JavaComm is a client–server communication platform offering chat, telephony, screen sharing, conferencing, a media player, and configurable media streams. Users can define and manage their own streams for flexible playback. All server-based communication runs over HTTPS (SSL/TLS), while file transfers are handled directly between users without passing through the server. The JavaComm Server runs on any Jakarta EE Web Profile 10 platform with the required extensions, using Tomcat 10.1.60 as... -
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ictcrm
Open source CRM with built-in contact centre integration
An open source CRM built on SuiteCRM and wired into our contact centre stack, so the customer record and the phone call are not sitting in two separate systems. The CRM side is what you would expect: accounts, contacts, leads, opportunities, cases and campaigns, with the reporting and workflow SuiteCRM already gives you. What we added is the telephony integration. Calls can be placed from a record, an inbound call can bring the matching record up, and campaign activity flows back into... -
MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
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ictcore
Unified communications framework and REST API for developers
ICTCore is the communications framework the rest of our open source stack is built on. It sits above FreeSWITCH and gives web developers a REST API for voice calls, fax, SMS and email instead of a dialplan to learn. The idea is that a communications workflow should be something you write in your own application. You describe what should happen, and ICTCore deals with the switch, the media and the state machine underneath. It is the engine behind ICTFax and the open source ICTDialer, so it is exercised daily by real deployments rather than being a demonstration piece. ... -
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AppKonference , a high-performance Asterisk conferencing module, is a fork of AppConference focused on voice.
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ChinaMobilePhoneNumberRegex
Regular expressions that match the mobile phone number in China
...Since MNP has been piloted in some areas, for users who have changed to another carrier, the mobile phone number prefix can no longer reflect its current carrier. According to the relevant documents of the MIIT, the voice call function is allowed in 145 / 147 / 149 prefixes, carriers can decide whether to provide voice call function according to the needs of their customers. At present, the 147 / 149 prefixes that support voice calls is available. Operated by China Transport Telecommunication & Information Center. -
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NVG589
Source Code content for Open Source packages used by the NVG589
Source code content for Open Source packages used by the NVG589, which is a residential gateway VDSL CPE with 802.11n wifi, 4 port ethernet switch, 1 EWAN port, 2 voice ports, HPNA port and a USB port. -
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NVG599
residential gateway VDSL CPE with 802.11a/b/g/n/ac wifi, ...
The NVG599 is a residential gateway VDSL CPE with 802.11a/b/g/n/ac wifi, 4 port ethernet switch, 1 EWAN port, 2 voice ports, HPNA port and a USB port. -
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Telemarketing (Junk) Call Blocker
Blocks unwanted calls using caller ID and a modem
...RECENT ACTIVITY: (11/2014):Added Raspberry Pi support for two USB modem brands. (01/2015): Added Raspberry Pi support for USR5686G modem. (06/2015): Added full-function program for ATian voice/FAX modem. (01/2016): Added jcEdit.py file edit program. (05/2016): Confirmed Sewell modem same as ATian modem. (07/06/2016): File download count passed 10,000. (11/2016): Changed contact email to: walsheath@gmail.com (01/2017): Updated jcEdit.py (See Discussion menu for details). (03/2018): Uploaded final jcblockAT update . (06/2018): Final jcblockAT update success (see Discussion menu or README2 file). (07/2018): Uploaded final jcblock update. -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call Centers and Contact Centers for Quality monitoring (QM) purposes. -
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NVG348Q
ARRIS NVG34x-Series Voice, Video and Data residential gateway
The ARRIS NVG34x-Series Triple Play (Voice, Video and Data) and Dual-Play (Video and Data) Residential Gateways are designed to deliver robust video, high-speed data and (optionally) primary line telephony, all delivered over the VDSL/VDSL2/ADSL2+ broadband network. This full-featured, high-performance Gateway provides a cost-effective way for Service Providers to migrate seamlessly from legacy ADSL networks to VDSL/VDSL2 networks via a converged services platform, made possible with the ARRIS 9x CPE Software. -
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SIP Data Filter (SiDaFir)
Simple and efficient tool for SIP trace filtering
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. Thanks to its simplicity, SIP messages are often used in creative ways for which these were not originally designed (e.g. using periodical OPTIONS packets as NAT keep-alive instead of using STUN or TURN) and thus SIP traces of the captured traffic often contain "useless" traffic from the actual signalization point of view. ... -
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Airean
GSM air interface
Airean is a proof-of-concept application showing that eavesdropping of GSM traffic (voice calls, mobile locations, SMS messages) is possible using a standard desktop PC with a small investment in hardware. The application requires a hardware interface to capture signals. Currently the following devices are supported: * Ettus Research USRP * Great Scott Gadgets HackRF (will be implemented when the device is released) Airean has the following capabilities: * Scan the GSM spectrum * Show nearby GSM base stations on a map * Monitor broadcast channels for active mobile terminals * With knowledge of session key: decode downlink SMS messages and audio * By interfacing to external programs: obtain session key from a captured communication with high probability For more information, see the Wiki section. -
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for example, HTTP servers). ... -
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SVG1501 and SVG2501
OSS project for the SVG1501 and SVG2501 Wireless Voice Gateway
This is the OSS redistribution project for the SVG2501, SVG1501 Series SURFboard® Wireless Voice Gateway. "The Motorola SVG2501 SURFboard® Wireless Voice Gateway is a fully integrated telecommunications solution that combines the functionality of a DOCSIS® 2.0 cable modem, an embedded multimedia telephone adapter, 4-port router with advanced firewall, an 802.11b/g wireless access point, and a field-replaceable Lithium-ion back-up battery—all in a stylish enclosure for the sophisticated consumer. ...
