USAN / res_speech_gdfeLinks
Asterisk speech driver for Google Dialogflow
☆16Updated 5 years ago
Alternatives and similar repositories for res_speech_gdfe
Users that are interested in res_speech_gdfe are comparing it to the libraries listed below
Sorting:
- ☆88Updated last week
- ☆37Updated 5 years ago
- Stream Asterisk audio over Websockets☆186Updated 3 months ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- ☆63Updated 3 years ago
- AGI-server voice recognizer for #Asterisk☆101Updated 2 years ago
- Speech Recognition in Asterisk with Vosk Server☆127Updated last year
- Register Asterisk on consul☆29Updated 6 months ago
- Kubernetes dynamic configuration engine for Asterisk☆71Updated 3 years ago
- Simple bidirectional audio protocol☆102Updated last month
- Asterisk Streaming Connection with google speech API☆14Updated 9 years ago
- Classes of Telephony for the Python's diagrams package☆11Updated 4 years ago
- Load-balancing SIP proxy for Freeswitch☆27Updated 9 years ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆43Updated 11 years ago
- ☆16Updated 9 years ago
- asterisk json utilities☆45Updated 2 years ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- Node.js client and server for FreeSwitch Event Socket☆140Updated 11 months ago
- UniMRCP modules for Asterisk☆51Updated last year
- Library to help write Asterisk Dialplans.☆27Updated 6 months ago
- Asterisk AGI script that makes use of Microsoft Azure Cognitive Services for text to speech synthesis☆19Updated 3 years ago
- Realtime configuration engine and CDR&CEL backend for Asterisk with MongoDB☆36Updated 7 years ago
- React SIP user agent☆56Updated 6 years ago
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆44Updated 2 years ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆183Updated last year
- An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine☆33Updated 2 years ago
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆112Updated last year
- Asterisk Management Interface (AMI) to Web-socket proxy☆90Updated 2 years ago
- Simple RTPEngine Speech-to-Text Recording Spooler☆16Updated last year
- SIP provisioning server / Auto configuration system (ACS)☆30Updated 2 years ago