dioris-moreno / ari-typescript-examples
Asterisk REST interface TypeScript Examples
☆14Updated last year
Related projects ⓘ
Alternatives and complementary repositories for ari-typescript-examples
- drachtio signaling resource framework☆171Updated this week
- ☆66Updated 5 months ago
- A SIP call processing server that can be controlled via nodejs applications☆248Updated last week
- An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine☆32Updated last year
- Asterisk speech driver for Google Dialogflow☆16Updated 4 years ago
- SIPREC recording server based on drachtio and rtpengine☆83Updated 4 months ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆28Updated 3 years ago
- An Asterisk REST Interface (ARI) websocket and API client library☆21Updated 10 months ago
- React based front-end demo for Asterisk's SFU☆30Updated last year
- Asterisk REST Interface proxy system written in Go☆32Updated 5 years ago
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆48Updated 2 months ago
- UDP implementation of the SIP.js library☆36Updated 11 months ago
- An ARI Bridge that enables spying on individual participants for STT purposes☆14Updated last year
- WebRTC SIP based VoIP client software (+chrome extension)☆99Updated last week
- Speech Recognition in Asterisk with Vosk Server☆105Updated 5 months ago
- ☆43Updated 5 months ago
- ☆34Updated 2 weeks ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆156Updated 4 months ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆35Updated 6 years ago
- Realtime configuration engine and CDR&CEL backend for Asterisk with MongoDB☆36Updated 6 years ago
- Asterisk Resource Interface for Trio☆25Updated last month
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆196Updated this week
- Node.js client and server for FreeSwitch Event Socket☆131Updated 2 months ago
- Asterisk PBX configuration syntax checker☆63Updated last year
- Asterisk external speech to text application☆49Updated last year
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆112Updated 2 years ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆37Updated 11 months ago
- Demo of scalable Asterisk on Kubernetes☆167Updated last year
- SIP Phone WebRTC for your browser☆57Updated 3 years ago