yeti-switch / sems
SEMS core forked from https://github.com/sems-server/sems
☆12Updated 2 weeks ago
Related projects: ⓘ
- Skype for Business/Lync PSTN gateway☆24Updated 5 years ago
- Applications for FusionPBX☆39Updated last month
- Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS☆13Updated 9 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆60Updated last month
- Nagios Plugin to check Call Quality in SIP VoIP (compatible checkmk, etc)☆31Updated 3 years ago
- Multitenant PBX☆29Updated 8 months ago
- Documentation and Tutorials for Kamailio SIP Server☆24Updated 2 weeks ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆36Updated 9 months ago
- SIPp Call Scenario for Performance Test☆14Updated 3 years ago
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆46Updated 2 months ago
- Firewall for VoIP systems☆9Updated 4 years ago
- SEET: Orchestrating multiple user-agents for complex SIP testing scenarios.☆20Updated last week
- Unified Communications System☆34Updated last year
- A curated list of HEP / EEP enabled projects☆28Updated 5 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- Kamailio Command Line Interface Control Tool☆51Updated 3 weeks ago
- Black Box SIP Tester☆30Updated 2 months ago
- SIPP-Scenarios for IMS☆9Updated 6 years ago
- HOMER 5: Back-End (API) DEPRICATED - use sipcapture/homer-app☆27Updated 4 years ago
- Unofficial armhf build of wazo-platform☆10Updated 4 years ago
- Least cost route finder (for Asterisk)☆9Updated 9 years ago
- (obsolete) Debian/Ubuntu package for the Bcg729 G.729 codec library☆14Updated last year
- Audio fingerprinting and recognition module for the Asterisk☆10Updated 6 years ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆106Updated 2 years ago
- Block unwanted registrations from SIP Scan attack using Asterisk AMI and iptables☆13Updated 4 months ago
- Simple dialer is a system for automatic dialing of subscribers using an asterisk.☆12Updated 3 years ago
- ICTCore: Unified Communications Framework for web developers. Communications APIs for voice calls, SMS messaging, Fax communications an…☆22Updated 3 months ago
- MagnusBilling, OpenSource SoftWitch and billing to Asterisk. Developed with EXTJS and YiiFramework☆12Updated 7 years ago
- SIPCheck is a tool that watch the authentication of users of Asterisk and bans automatically if some user (or bot) try to register o make…☆25Updated 4 years ago
- Barebone Opensource Powered SBC☆106Updated 3 weeks ago