2600hz / monster-ui-callflows
Monster UI Application: Advanced configuration of KAZOO callflows using a drag-and-drop interface
☆12Updated last week
Alternatives and similar repositories for monster-ui-callflows:
Users that are interested in monster-ui-callflows are comparing it to the libraries listed below
- Monster UI Application: Simplified PBX configuration for end-users☆31Updated last month
- The JavaScript framework to leverages the power of Kazoo☆72Updated last week
- Docs for System Admins☆17Updated 8 months ago
- ☆12Updated 2 weeks ago
- Monitor VoIP call quality☆17Updated 11 years ago
- [DEPRECATED] SIP Swiss army knife☆18Updated 5 years ago
- SEMS core forked from https://github.com/sems-server/sems☆13Updated 2 weeks ago
- Voice quality probe for end-to-end measurement of jitter and packet loss☆41Updated 10 years ago
- Scripts from the 2600hz Community to help manage, enhance and debug the KAZOO platform☆42Updated last year
- 3GPP Home Subscriber Server (HSS)☆27Updated last year
- sipXecs Unified Communication System http://www.sipfoundry.org☆25Updated 3 years ago
- The Great Big Book Of Kazoo☆37Updated 8 years ago
- FreeSWITCH dialer program for VoIP performance tests☆44Updated 7 years ago
- Erlang RADIUS server framework☆57Updated 6 months ago
- Kazoo Ansible Playbooks☆13Updated 6 years ago
- The original KAZOO UI initially developed as a reference application for utilizing the KAZOO APIs☆54Updated 2 years ago
- Erlang & FreeSWITCH based Automated Contact Distribution (ACD) system☆137Updated 12 years ago
- automatic device provision in monster-ui-voip☆19Updated 8 years ago
- Unified Communications System☆34Updated 3 months ago
- Configurations for Kamailio, FreeSWITCH and all other components of KAZOO☆19Updated 7 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆62Updated last week
- Forked from http://sysadminman.net/blog/2013/asterisk-outbound-call-status-page-5600 and modified for FreePBX☆19Updated 9 years ago
- Web application to faciliate benchmarking and testing SIP based services☆20Updated 8 years ago
- HOMER 5: Back-End (API) DEPRICATED - use sipcapture/homer-app☆27Updated 5 years ago
- SIP Voice Quality reports Collector☆14Updated 8 years ago
- Asterisk module that provides the "Flite" dialplan application, which allows you to use the Flite text to speech engine with Asterisk..Wo…☆28Updated last year
- SIPP-Scenarios for IMS☆10Updated 7 years ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆167Updated 10 months ago
- erGW - Erlang implementations of GGSN or P-GW☆84Updated 3 years ago
- SIP Phone Learning Tool☆36Updated 3 years ago