hnimminh / sip-healthcheck
Monitor SIP server and Notify whenever downtime/latency detected.
☆15Updated 5 years ago
Alternatives and similar repositories for sip-healthcheck
Users that are interested in sip-healthcheck are comparing it to the libraries listed below
Sorting:
- Wazo Platform SBC Kamailio configuration for the C4 infrastructure☆13Updated 2 years ago
- Kamailio in Kubernetes configuration manager☆14Updated 6 years ago
- ☆10Updated 2 years ago
- ☆35Updated this week
- Simple config file of Kamailio as Loadblancer for calls and registrations☆32Updated 3 years ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- Kamailio+rtpengine failover demo infrastructure for kamailio-ru-2020 conference☆15Updated 3 years ago
- SIPp Call Scenario for Performance Test☆17Updated 4 years ago
- Prometheus exporter for Kamailio SIP server☆60Updated last year
- SIP provisioning server / Auto configuration system (ACS)☆28Updated 2 years ago
- REST API for sharing IP addresses sending unwanted SIP traffic☆61Updated last year
- SIP outbound proxy based on drachtio and freeswitch that includes siprec client functionality☆19Updated 10 months ago
- Classes of Telephony for the Python's diagrams package☆11Updated 3 years ago
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆50Updated 5 months ago
- 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
- ☆16Updated 8 years ago
- Docker files to easily build Kamailio on different Debian/Ubuntu releases☆17Updated 4 months ago
- Nagios Plugin to check Call Quality in SIP VoIP (compatible checkmk, etc)☆31Updated 4 years ago
- Register Asterisk on consul☆29Updated last week
- sample sipp scenarios for testing freeswitch☆33Updated 2 years ago
- Firewall for VoIP systems☆11Updated 4 years ago
- Kamailio Command Line Interface Control Tool☆53Updated 4 months 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
- Meta-repository to test interop between latest versions of opensips, kamailio, sippy b2bua and rtpproxy☆43Updated this week
- ☆24Updated last year
- Asterisk Metrics sent to Statsd☆17Updated 7 years ago
- Least cost route finder (for Asterisk)☆9Updated 9 years ago
- Freeswitch setup, profiles , dial-plans and vars for various use-cases☆17Updated 3 years ago
- Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS☆13Updated 10 years ago
- Simple Queue Monitor for Asterisk using AsterNET framework☆19Updated 3 years ago