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