Requirements
-Deep knowledge with leading unified collaboration systems like Asterisk, Free Switch
-Good knowledge of SIP-Servers such as Kamailio
-Good knowledge of WebRTC
-knowledge of AGI, Ami, FastAGI, Dialplan, FS Scripts
-Good knowledge of programming languages like c++, c, Javascript
-Good knowledge of REST-API
-knowledge about Git
-Hands-on experience with Session Border Controllers and Voice Gateways
-Expert level knowledge of VOIP protocols such as SIP, RTP/SRTP including TLS
-High Responsibility
-Partnership and Teamwork skills
-Systematic and Creative Thinking
-Result Oriented
