Search Results for "automatic1111-stable-diffusion"
Sort By:
Showing 95 open source projects for "automatic1111-stable-diffusion"
View related business solutions-
Full-stack observability with actually useful AI | Grafana CloudBuilt on open standards like Prometheus and OpenTelemetry, Grafana Cloud includes Kubernetes Monitoring, Application Observability, Incident Response, plus the AI-powered Grafana Assistant. Get started with our generous free tier today.
-
Try Google Cloud Risk-Free With $300 in CreditUse your credit across every product. Compute, storage, AI, analytics. When it runs out, 20+ products stay free. You only pay when you choose to.
-
1
Synapse
Matrix reference homeserver
...Rooms can be located using convenience aliases like #matrix:matrix.org or #test:localhost:8448. Synapse is currently in rapid development, but as of version 0.5 we believe it is sufficiently stable to be run as an internet-facing service for real usage! Create and manage fully distributed chat rooms with no single points of control or failure. Eventually-consistent cryptographically secure synchronization of room state across a global open network of federated servers and services. Send and receive extensible messages in a room with (optional) end-to-end encryption. ... -
2
IssabelPBX
Issabel PBX - Unified Communications
Open Source and Unified Communications partners created a new platform based on an Elastix® fork (currently purchased by 3CX) to provide the community with continuity, peace of mind and support needed to continue with their PBX and operation developments. Contribute to the funding of Issabel on https://www.patreon.com/issabel -
3
VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
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. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or... -
4
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
-
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.
-
5
ICTFax
Open source Fax server software for inbound / outbound internet Faxing
Note: for binaries and installation instruction please visit official website. http://ictfax.org/ ICTFax is Open Source Fax Server software to send and receive Fax over Internet & Emails developed over Freeswitch, ICTCore and Angular Framework , It works with T.38 and G.711 pass through and feature email to fax, fax to email, web to fax, fax to web and also support fax machines to send and recieve Fax using ATA. Installation... -
6
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. This helps sales and support teams make faster decisions, improve response quality, and enhance customer... -
7
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
-
8
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. -
9
FusionPBX Live Medium
Project about FusionPBX on Live Medium
If you want to quickly set up a multi-tenant PBX on matter of few minutes, this is a suitable choice. - Demo environment. You can start to add extensions, gateways and dialplans, and quickly have a fully working PBX environment. - Import previous FusionPBX backup to Live environment. After this you will have a quickly working PBX on few minutes. - Install on hard disk. With this option you can install the environment to Hard Disk. Username and password for PostgreSQL is ... -
8 Monitoring Tools in One APM. Install in 5 Minutes.AppSignal works out of the box for Ruby, Elixir, Node.js, Python, and more. 30-day free trial, no credit card required.
-
10
Asterisk Channels Live
This is a PBX dashboard showing the peer,queue,agent at real time
http://www.astchannelslive.com/sourceforge.php This is a dashboard work with Asterisk(FreePBX) any other installation: 1. Showing the peer status 2. queue,agent at real time 3. Hangup, Spy,.... -
11
Free Telephony Project
Free software and hardware for telephony
Legacy SVN repositories for projects from David Rowe http://rowetel.com Mailing lists for Codec 2 and SM1000. -
12
Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on... -
13
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/
-
14
Kazoo Live Medium
Kazoo Live Medium. From Null to a Full Featured PBX on minutes
From Null to a Full Featured PBX in matter of minutes!! If you want: - To boot or install quickly Kazoo on your server. - To test about how it works, without installing on your hard drive. - To install on cluster environment - To install if your previous platform go downtime. This is a suitable option. What is included? Monster-UI Open Source Apps, like SmartPBX, CallFlows, PBX Connector, Voicemails, Faxes, Accounts and Number Manager Kazoo v4.3.x Kamailio ... -
15
AstCDR
AstCDR
Sistema de CDR para Asterisk Puro em Portugues do Brasil. -
16
Komendant ERP
ERP, IP PBX, Accounting, CRM, Time tracking, and etc.
Comprehensive business management system The full-cycle automation ERP system - [ Full Freeware ] IP PBX - Complete synchronization with CRM system - Call accounting for PBX (PBX billing) - Phone call control and monitoring - Phone recording systems Accounting, Financial management - Trade management, warehouse - Management and tax accounting Corporate information system - Client relations management system (CRM) - Project, tasks, orders management - On-line conferences... -
17
Taki
SIP softphone
Cross-platform SIP softphone -
18
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/ -
19
MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. . -
20
A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools, covering important functionalities (MI,statistics) and modules (acc,siptrace,drouting,dialplan) of OpenSIPS. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips-cp .
-
21
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.
-
22
VoiceOne gives you the ability to install and compleatly configure a pbx platform based on Asterisk 1.8 with an easy to use web GUI, which would be a framework to build a communication server adding various plugins.
-
23
A convenient alternative to single-purpose touch screens using IP phone application APIs. This plug-in provides alerting, information display and home interaction capabilities, and is customized and controlled from within the HomeSeer web interface.
-
24
PolycomVVXControl
A command line utility for remote controlling Polycom VVX IP phones
Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file. -
25
Asterisk CDR import
Simple web interface for mapping CDR from VoIP provider to Asterisk
This is a simple tool to map information for the cost of phone calls given by your VoIP provider, to the call log inside your Asterisk. As a result, you will know how much each of your employees has spent for phone calls. Additionally, this tool shows information about the assigned DID and CLIP numbers in Asterisk. This is helpful to identify obsolete, duplicate and multiple assignments.
