Hi,
I am VoIP developer and I have couple of year experience in VoIP/asterisk/FreeSWITCH/OpenSIPs/Kamailio/RTPproxy/WebRTC/RTPengine/ASTPP. I have developed Click to call, Class4 & Class5 Soft Switches, SIP Proxy Server, SIP Redirect Server, SIP Load Balance & Fail-Over, IVR Application, Fax over IP (FoIP), T38 Supported FAX Server, FAX to Email, SHOUTcast Server, SIP Messaging Server, High scalable Least Cost Routing(LCR) with Fail-Over, Session Border Controller(SBC), WebRTC client, RESTfull API , Web Services for Mobile Application, Web Applications, Mobile Applications.
As I am experienced person in FreeSWITCH, so I can setup FreeSWITCH platform as per your requirement and also ready to provide you support as per your need.
So I would like to apply for your given job.
I am looking forward for your favorable response.
Thanks.