sipXcom / sipxecs
Unified Communications System
☆34Updated 4 months ago
Alternatives and similar repositories for sipxecs:
Users that are interested in sipxecs are comparing it to the libraries listed below
- SIP Phone Learning Tool☆36Updated 3 years ago
- Web Admin Interface for Kamailio☆102Updated 2 months ago
- A curated list of HEP / EEP enabled projects☆28Updated 6 years ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org☆161Updated 2 months ago
- SIP Express Media Server, very fast and flexible SIP media server☆62Updated this week
- sipXecs Unified Communication System http://www.sipfoundry.org☆25Updated 3 years ago
- Kamailio Command Line Interface Control Tool☆53Updated 3 months ago
- HEP-EEP: Extensible Encapsulation Protocol (Specs & Technical Docs)☆46Updated 2 years ago
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆102Updated 2 weeks ago
- This Project will provide the inbound sip using that we can route did call to customer ip or customer Phone number☆24Updated 7 years ago
- Nagios Plugin to check Call Quality in SIP VoIP (compatible checkmk, etc)☆31Updated 3 years ago
- SEMS core forked from https://github.com/sems-server/sems☆13Updated 2 weeks ago
- React SIP user agent☆53Updated 5 years ago
- Kamailio SIP Proxy with Sipwise patches☆61Updated last week
- REST API for sharing IP addresses sending unwanted SIP traffic☆61Updated last year
- Skype for Business/Lync PSTN gateway☆24Updated 6 years ago
- ☆35Updated 2 weeks ago
- Least cost route finder (for Asterisk)☆9Updated 9 years ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆81Updated 8 years ago
- Blox route configuration (opensips script)☆22Updated 4 years ago
- FreeSWITCH dialer program for VoIP performance tests☆44Updated 7 years ago
- VoIP signaling and media test automation☆119Updated 3 weeks ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆167Updated 11 months ago
- Monitor SIP server and Notify whenever downtime/latency detected.☆15Updated 4 years ago
- Documentation and Tutorials for Kamailio SIP Server☆28Updated this week
- 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
- Various installation scripts for Asterisk.☆29Updated 9 years ago
- HOMER 5: Back-End (API) DEPRICATED - use sipcapture/homer-app☆27Updated 5 years ago
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆50Updated 4 months ago
- Applications for FusionPBX☆41Updated last month