Create an account with SIP.US


https://www.sip.us/get-started/


Once an account is established for use, login to the SIP.US customer portal and navigate to SIP Trunking > SIP Trunks:




Click the + to expand the Trunk Registration Information 



These username, password, and server fields translate directly into the WIN-911 Voice Gateway settings:




In the WIN-911 Workspace > Voice Gateway settings - Check the "Registration Required" box, enter the Trunk # as the User ID and Registration ID, enter the password from the SIP.US trunk "show password" field.


Note: the password can be changed by the SIP.US account holder at any time by clicking the "Modify trunk #xxxxxxxxxx" link


Enter "gw1.sip.us" as the Server Address.




To ensure the actual RTP Voice traffic (the speaking on the call) is not lost through NAT at the firewall, it is recommended to use the SIP.US built-in STUN server.


A STUN (Session Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. IP Phones behind a firewall) to setup phone calls to a VoIP provider hosted outside of the local network.


stun server

The STUN protocol is defined in RFC 3489.



To utilize the STUN server, simply change the NAT type in WIN-911 to STUN and enter "stun.sip.us" in the server field:




Important! - If you have more than 1 IP address assigned to your machine running WIN-911, you MUST change the "Binding Address" setting to "Specify" and put in the IP address that is on the correct network to communicate with the internet.


All other settings can remain default.


Save the configuration and test SIP settings to call a valid phone number.