zaf / asterisk-speech-recogLinks
Speech recognition script for Asterisk that uses google's speech engine.
☆252Updated 7 years ago
Alternatives and similar repositories for asterisk-speech-recog
Users that are interested in asterisk-speech-recog are comparing it to the libraries listed below
Sorting:
- Asterisk AGI script that uses Google's translate text to speech service.☆219Updated last year
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆304Updated 2 years ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated 2 months ago
- Asterisk module that provides the "Flite" dialplan application, which allows you to use the Flite text to speech engine with Asterisk..Wo…☆28Updated 2 years ago
- FreeSWITCH dialer program for VoIP performance tests☆46Updated 7 years ago
- 100% Open-Source Packet Capture Agent for HEP☆173Updated 3 months ago
- Dashboard for Queues in Asterisk and FreeSWITCH. app_queue panel for Asterisk and mod_callcenter in FreeSWITCH. Get news -> http://eepu…☆185Updated last year
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆211Updated this week
- AGI-server voice recognizer for #Asterisk☆100Updated 2 years ago
- Asterisk AGI script that makes use of Microsoft Azure Cognitive Services for text to speech synthesis☆19Updated 3 years 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 …☆325Updated last year
- Stream Asterisk audio over Websockets☆182Updated 11 months ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆175Updated last year
- Asterisk Management Interface (AMI) to Web-socket proxy☆87Updated 2 years ago
- Asterisk PBX configuration syntax checker☆66Updated 2 years ago
- Create SIP load test scenarios the easy way☆224Updated 5 years ago
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆207Updated this week
- Web status monitor for FreeSWITCH's mod_callcenter queues and agents☆38Updated 11 years ago
- Speech Recognition in Asterisk with Vosk Server☆119Updated last year
- A SIP call processing server that can be controlled via nodejs applications☆286Updated 3 weeks ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- Barebone Opensource Powered SBC☆110Updated 3 weeks ago
- A2Billing is a telecom switch and billing system capable of providing and billing a range of telecom products and services to customers s…☆188Updated 2 years ago
- ☆83Updated 4 months ago
- WebRTC SIP based VoIP client software (+chrome extension)☆108Updated 8 months ago
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆188Updated last month
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆297Updated last year
- ☆35Updated 2 weeks ago
- asterisk json utilities☆43Updated 2 years ago