What SIP Proxy should the gateway point to?

Set the SIP proxy for this service should point to sip.switch2voip.us UDP port 5060. Below is a sample configuration for a typical Asterisk gateway. The same parameters will apply for other types of gateways but the configuration may look slightly different. Please consult with the gateway vendor or software provider if unsure how to configure the below parameters.
On open source applications such as Asterisk, setup your SIP trunk as follows.
Please note that no SIP registration is required against Switch2Voip’s platform.

General
port=5060
dtmfmode=rfc2833
Progressinband=never
allow=g729
allow=g711
allow=g723

Outbound Calling
type=peer
username=USERNAME
secret=PASSWORD
host=sip.switch2voip.us
port=5060
canreinvite=no

Registration
register=>USERNAME:PASSWORD@sip.switch2voip.us/siptrunking

ADD CREDITS TO VOIP ACCOUNT