andrius / asteriskLinks
β¨π Asterisk PBX in π³ Docker β Smallest Asterisk ever! π
β331Updated this week
Alternatives and similar repositories for asterisk
Users that are interested in asterisk are comparing it to the libraries listed below
Sorting:
- Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCPβ532Updated 3 years ago
- Demo of scalable Asterisk on Kubernetesβ174Updated 2 years ago
- Docker image providing Asterisk PBXβ268Updated last month
- Asterisk Management Interface (AMI) to Web-socket proxyβ93Updated 2 years ago
- Stream Asterisk audio over Websocketsβ189Updated 5 months ago
- ARI examples in Python and JavaScript.β112Updated 10 years ago
- Some dockerfiles for whipping up an asterisk serverβ291Updated 5 years ago
- β90Updated 2 weeks ago
- Asterisk AGI script that uses Google's translate text to speech service.β222Updated last year
- Node.js client for ARI. This library is best effort with limited support.β281Updated last year
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sarβ¦β183Updated last year
- Web Admin Interface for Kamailioβ103Updated 11 months ago
- IVOZ Provider - Multitenant solution for VoIP telephony providersβ218Updated this week
- A SIP call processing server that can be controlled via nodejs applicationsβ305Updated last week
- Setup for a WEBRTC client and Kamailio server to call SIP clientsβ308Updated 2 years ago
- Dashboard for Queues in Asterisk and FreeSWITCH. app_queue panel for Asterisk and mod_callcenter in FreeSWITCH. Get news -> http://eepuβ¦β191Updated last year
- HOMER 7 Docker Imagesβ96Updated last year
- ctxSip is a Javascript based SIP client that uses WebRTC and WebSockets to connect to your SIP server. The UI is designed to be launched β¦β337Updated last year
- Asterisk PBX configuration syntax checkerβ68Updated 3 years ago
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Servicesβ225Updated 2 weeks ago
- Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCHβ57Updated 3 years ago
- NATS or RabbitMQ message bus Asterisk REST Interface proxy system implemented in Goβ85Updated 2 months ago
- Asterisk Queues Dashboard with amiwsβ55Updated 2 years ago
- Modern and flexible SIP/VoIP cli toolβ375Updated last month
- β134Updated this week
- Call API is a front-end layer for managing advanced SIP call flows. It listens for WebSocket connections and talks JSON-RPC 2.0 over themβ¦β56Updated 2 months ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projectsβ170Updated 8 months ago
- An open source Session Border Controller π The SBC you dream about π½ LibreSBC will help you save thousands of dollars.β457Updated 3 weeks ago
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/β125Updated 3 years ago
- A fully featured browser based WebRTC SIP phone for Asteriskβ636Updated last year