Hi,
I am VoIP developer and I have couple of year experience in VoIP/asterisk/FreeSWITCH/OpenSIPs/Kamailio/RTPproxy/WebRTC/RTPengine/ASTPP/A2billing. 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 Asterisk/FreeSWITCH, so I can setup VoIP platform as per your requirement and also ready to provide you support as per your need.
You can find my work on google or here on freelancer profile https://www.freelancer.in/u/prayantech
Let's discuss your detail requirements over chat messanger of freelancer.
Thanks,
Malay
PrayanTech Business Solutions.