AlexPereverzyev / sip_telLinks
The projects demonstrates how tenants softphones and FreeSWITCH PBXs can be connected to each other using Kamailio SIP router to enable advanced calls routing, logging, measurements and NAT traversal.
☆16Updated 4 months ago
Alternatives and similar repositories for sip_tel
Users that are interested in sip_tel are comparing it to the libraries listed below
Sorting:
- Simple config file of Kamailio as Loadblancer for calls and registrations☆32Updated 3 years ago
- Kamailio Command Line Interface Control Tool☆53Updated 5 months ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆124Updated 3 years ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆172Updated last year
- Kamailio config for public/private proxy with rtpengine☆12Updated last year
- Barebone Opensource Powered SBC☆109Updated 10 months ago
- Web Admin Interface for Kamailio☆103Updated 4 months ago
- REST API for sharing IP addresses sending unwanted SIP traffic☆62Updated last year
- ☆62Updated 3 years ago
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆104Updated 2 months ago
- Asterisk Queues Dashboard with amiws☆54Updated last year
- HOMER 7 Docker Images☆96Updated 10 months ago
- Freeswitch setup, profiles , dial-plans and vars for various use-cases☆17Updated 3 years ago
- Kamailio Wiki with content in markdown format☆39Updated this week
- SIP performance test tool☆39Updated 2 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆208Updated this week
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- OpenSIPS AI Voice Connector Community Edition Platform☆49Updated 2 weeks ago
- OpenSIPS CLI tool - an interactive command line tool that can be used to control and monitor OpenSIPS servers.☆95Updated last month
- Asterisk Management Interface (AMI) to Web-socket proxy☆87Updated 2 years ago
- VoIP signaling and media test automation☆119Updated last month
- Voip Open Linear Testing Suite☆43Updated last month
- 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
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆210Updated this week
- Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH☆54Updated 2 years ago
- Freeswitch installation guides and config files☆55Updated 5 months ago
- ☆35Updated 2 weeks ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 7 years ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆163Updated 3 weeks ago