gonicus / fsqueuemonLinks
Web status monitor for FreeSWITCH's mod_callcenter queues and agents
☆38Updated 11 years ago
Alternatives and similar repositories for fsqueuemon
Users that are interested in fsqueuemon are comparing it to the libraries listed below
Sorting:
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆176Updated last month
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆191Updated 3 weeks ago
- Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent☆142Updated 4 months ago
- A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools…☆127Updated 3 weeks ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆303Updated last year
- Sip Express Media Server☆175Updated this week
- Load-balancing SIP proxy for Freeswitch☆26Updated 9 years ago
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆305Updated 3 years ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated 3 months ago
- Web Admin Interface for Kamailio☆103Updated 6 months ago
- Create SIP load test scenarios the easy way☆225Updated 5 years ago
- This project can be used to deploy a FreeSWITCH server inside a Docker container. The container currently uses the latest stable release …☆249Updated 8 years ago
- FreeSWITCH mod_amd☆42Updated last year
- FreeSWITCH dialer program for VoIP performance tests☆47Updated 7 years ago
- SIPREC recording server based on drachtio and rtpengine☆94Updated last year
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- async FreeSWITCH cluster control☆74Updated last year
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆104Updated 5 months ago
- 100% Open-Source Packet Capture Agent for HEP☆173Updated 4 months ago
- SIP/XMPP/WebRTC Application Server☆206Updated 2 months ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆175Updated 2 months ago
- 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 …☆327Updated last year
- Project to monitor FreeSwitch status and health☆43Updated 8 years ago
- UniMRCP modules for Asterisk☆51Updated last year
- SIPp scenarios I use for testing SIP stuff☆300Updated last year
- The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.☆450Updated this week
- Linphone.org mirror for bcg729 (git://git.linphone.org/bcg729.git)☆129Updated last year
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆166Updated 3 months ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆176Updated last year
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆111Updated last year