I believe all you need to do are to configure your account on your asterisk to register to the VoSP. Then, craft an incoming dial-plan context to provide the IVR.
For sip signalling open and redirect the udp port 5060, for the audio open and redirect udp ports range 10000:20000, check rtp.conf; if you have few concurrent calls change the value in rtp.conf to ease the router configuration, two rtp ports per call are used.