Alternatives and similar repositories for FreeSWITCH-panel

Users that are interested in FreeSWITCH-panel are comparing it to the libraries listed below

• openvoips / OV500
  OpenSource Freeswitch & Kamailio Billing, rating and Routing Software
  ☆114Updated this week
• asipto / siremis-php
  Web Admin Interface for Kamailio
  ☆103Updated 10 months ago
• ricardojlrufino / webphone-sip
  WebRTC SIP based VoIP client software (+chrome extension)
  ☆111Updated last year
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆95Updated last year
• hnimminh / sip-healthcheck
  Monitor SIP server and Notify whenever downtime/latency detected.
  ☆15Updated 5 years ago
• gnostix / freeswitch-monitoring
  Project to monitor FreeSwitch status and health
  ☆46Updated 8 years ago
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆37Updated 7 years ago
• mwolff44 / pk-sbc
  P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine
  ☆108Updated 3 weeks ago
• havfo / SipCaller
  React SIP user agent
  ☆56Updated 6 years ago
• chanon-m / sip2ban.github.io
  Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS
  ☆14Updated 10 years ago
• sipwise / sems
  SIP Express Media Server, very fast and flexible SIP media server
  ☆70Updated 3 weeks ago
• alepolidori / janus-webrtc-phone
  SIP Phone WebRTC for your browser
  ☆59Updated 4 years ago
• os11k / sipp2freeswitch
  sample sipp scenarios for testing freeswitch
  ☆34Updated 3 years ago
• gmaruzz / saraphone
  SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…
  ☆183Updated last year
• pascomnet / asterisk_sbr
  Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.
  ☆44Updated 3 years ago
• AGProjects / mediaproxy
  Media relay for RTP/RTCP and UDP streams
  ☆47Updated 2 months ago
• wangduanduan / siphub
  sip capture server by hep。work with OpenSIPS, Kamailo, and FreeSWITCH。
  ☆69Updated 6 months ago
• davehorton / drachtio-fs-load-balancing-proxy
  Load-balancing SIP proxy for Freeswitch
  ☆28Updated 9 years ago
• voiceip / tinyphone
  Minimalist Windows / OSx / Linux SIP Softphone with API Control
  ☆152Updated 9 months ago
• davehorton / drachtio-siprec-recording-client
  SIP outbound proxy based on drachtio and freeswitch that includes siprec client functionality
  ☆20Updated last year
• OpenSIPS / call-api
  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…
  ☆56Updated last month
• voiceboys / sbcOS
  Barebone Opensource Powered SBC
  ☆114Updated 5 months ago
• agranig / pcap2mermaid
  A tool to create mermaid sequence diagrams from pcap files for SIP call flows
  ☆22Updated 6 years ago
• iNextrix / ASTPP
  Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects
  ☆169Updated 7 months ago
• 2600hz / kazoo-configs-kamailio
  ☆36Updated last week
• sippy / rtp_cluster
  RTP Cluster is a front-end for multiple RTPproxies
  ☆40Updated last year
• caruizdiaz / kamailio-ws
  Kamailio scripts to call from websocket UA to classic UA, and vice versa.
  ☆44Updated 11 years ago
• altanai / opensipsexamples
  sample configurations and settings for opensips for various use-cases
  ☆63Updated 6 years ago
• akscf / mod_openai_asr
  Freeswitch ASR module
  ☆14Updated 7 months ago
• unispeech / asterisk-unimrcp
  UniMRCP modules for Asterisk
  ☆51Updated last year