If you're using Asterisk, create a new context in your sip.conf as follows:

[siptermination]
type = peer
insecure = very
host = outbound1.wholesale.siptermination.net
dtmfmode = rfc2833
canreinvite = no
sendrpid = yes

and add any codec restrictions that you need (we recommend sticking with g.711u and g.729a for maximum quality).

You can point your outbound calls to this new context in your extensions.conf such as in this example:

[outdial]
exten =>_1NXX-NXX-XXXX,1,Set(CALLERID(number)=3023512250)
exten =>_1NXX-NXX-XXXX,2,Dial(SIP/sipidv/${EXTEN})



Calls should be sent to our SIP gateway in one of the 2 following formats:
sip:1NPANXXXXXX@outbound1.wholesale.siptermination.net for domestic (NANPA) dialed numbers
sip:011CCXXXXXXX@outbound1.wholesale.siptermination.net for International