thehunmonkgroup / jesterLinks
Scripting toolkit for FreeSWITCH written in the Lua programming language
☆47Updated last year
Alternatives and similar repositories for jester
Users that are interested in jester are comparing it to the libraries listed below
Sorting:
- Load-balancing SIP proxy for Freeswitch☆26Updated 9 years ago
- async FreeSWITCH cluster control☆74Updated last year
- RTP Cluster is a front-end for multiple RTPproxies☆41Updated last year
- FreeSWITCH mod_amd☆42Updated last year
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated last month
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- Project to monitor FreeSwitch status and health☆43Updated 8 years ago
- FreeSWITCH dialer program for VoIP performance tests☆47Updated 8 years ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 9 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆37Updated 7 years ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆176Updated 2 months ago
- Voice quality probe for end-to-end measurement of jitter and packet loss☆41Updated 11 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆68Updated last week
- 100% Open-Source Packet Capture Agent for HEP☆175Updated 6 months ago
- HEP-EEP: Extensible Encapsulation Protocol (Specs & Technical Docs)☆49Updated 3 months ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆43Updated 11 years ago
- This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.☆57Updated 6 years ago
- FreeSWITCH Dockerfile☆39Updated 5 years ago
- The original single-server, multi-tenant, switch agnostic UI built by 2600Hz☆105Updated 10 years ago
- 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 them…☆55Updated last month
- Prometheus exporter for Kamailio SIP server☆59Updated last year
- ☆36Updated this week
- ☆16Updated 9 years ago
- Create SIP load test scenarios the easy way☆225Updated 5 years ago
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆50Updated 3 months ago
- zmq/json support asterisk AMI module. (zeromq, 0MQ)☆16Updated 2 years ago
- FreeSWITCH socket client written in go (http://golang.org)☆75Updated 10 months ago
- Asterisk module that provides the "Flite" dialplan application, which allows you to use the Flite text to speech engine with Asterisk..Wo…☆28Updated 2 years ago
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆44Updated 2 years ago