jpawlowski / freeswitch-sounds-tts
FreeSWITCH TTS Voice Prompt Generator
☆43Updated 6 years ago
Alternatives and similar repositories for freeswitch-sounds-tts:
Users that are interested in freeswitch-sounds-tts are comparing it to the libraries listed below
- SIP Express Media Server, very fast and flexible SIP media server☆64Updated last week
- Scripting toolkit for FreeSWITCH written in the Lua programming language☆47Updated last year
- FreeSWITCH socket client written in go (http://golang.org)☆73Updated 4 months ago
- Project to monitor FreeSwitch status and health☆42Updated 8 years ago
- sample sipp scenarios for testing freeswitch☆31Updated 2 years ago
- async FreeSWITCH cluster control☆74Updated last year
- FreeSWITCH dialer program for VoIP performance tests☆46Updated 7 years ago
- FreeSWITCH mod_amd☆41Updated last year
- SIPREC recording server based on drachtio and rtpengine☆89Updated 9 months ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- RTP Cluster is a front-end for multiple RTPproxies☆41Updated 11 months ago
- Sip Express Media Server☆167Updated last week
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆168Updated 11 months ago
- Load-balancing SIP proxy for Freeswitch☆24Updated 8 years ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆172Updated 3 years ago
- Freeswitch Kafka Plugin☆38Updated last year
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆50Updated 4 months ago
- Asterisk Manager Proxy☆28Updated 4 years ago
- VoIP signaling and media test automation☆119Updated this week
- Strategical outbound dialing module for Asterisk☆9Updated 8 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- Kamailio SIP Proxy with Sipwise patches☆61Updated last week
- Audio fingerprinting and recognition module for the Asterisk☆10Updated 6 years ago
- Documentation and Tutorials for Kamailio SIP Server☆28Updated last week
- Web Admin Interface for Kamailio☆102Updated 2 months ago
- HEP-EEP: Extensible Encapsulation Protocol (Specs & Technical Docs)☆46Updated 2 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆79Updated 10 years ago
- The GO port of the Sippy B2BUA☆66Updated last month
- SIPp Call Scenario for Performance Test☆17Updated 4 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…☆54Updated last year