zaf / Asterisk-eSpeak
Asterisk module that provides the "eSpeak" dialplan application. It allows you to use the eSpeak text to speech synthesizer. Works with asterisk 1.6 or newer.
☆41Updated 11 months ago
Alternatives and similar repositories for Asterisk-eSpeak:
Users that are interested in Asterisk-eSpeak are comparing it to the libraries listed below
- 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
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆77Updated 9 years ago
- asterisk json utilities☆42Updated 2 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 8 years ago
- RTP Cluster is a front-end for multiple RTPproxies☆40Updated 8 months ago
- FreeSWITCH dialer program for VoIP performance tests☆43Updated 7 years ago
- Web status monitor for FreeSWITCH's mod_callcenter queues and agents☆38Updated 10 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆61Updated this week
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆168Updated 8 months ago
- Blox route configuration (opensips script)☆22Updated 4 years ago
- A reverse proxy for the FastAGI protocol☆27Updated 11 months ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆40Updated 10 years ago
- ☆16Updated 8 years ago
- Fork of the Asterisk VOIP software with support for the Opus codec☆14Updated 12 years ago
- Dockerfiles to easily build kamailio on different Debian releases☆14Updated 8 years ago
- React SIP user agent☆52Updated 5 years ago
- CDR mediation and rating engine for Call Details Records.☆16Updated last month
- AGI script that uses Google Translate API v2 for text translation and language detection.☆15Updated 8 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆43Updated 2 years ago
- Asterisk AGI script that makes use of Microsoft Azure Cognitive Services for text to speech synthesis☆19Updated 2 years ago
- Web application to faciliate benchmarking and testing SIP based services☆20Updated 8 years ago
- Pipe arbitrary data rows (logs, events, cdrs, esl, etc) to HEP Server (HOMER)☆25Updated 7 months ago
- RTPProxy - application for RTP packets relaying - additional patches☆29Updated 10 years ago
- Audio fingerprinting and recognition module for the Asterisk☆10Updated 6 years ago
- Asterisk 13 transcoding module: Opus☆32Updated 2 years ago
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆40Updated last year
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆301Updated 2 years ago
- A2Billing is a telecom switch and billing system capable of providing and billing a range of telecom products and services to customers s…☆182Updated 2 years ago
- ☆61Updated 2 years ago
- Voice quality probe for end-to-end measurement of jitter and packet loss☆41Updated 10 years ago