SIP normally will not work over a simple NAT router like SME. The voice call will set up, but the media transfer will not take place.
This is why:
Media transfer is done using a protocol called RTP, a point to point, low latency UDP protocol. The information to set this up is sent in the data portion of the SIP packets. The far end is attempting to communicate with your internal device, using its non-routable, internal address.
The long and short of it is, you have to use a SIP-aware gateway that, using deep packet inspection, grabs the internal address in your SIP packets and subs the outside address of the gateway, and intercepts the incoming RTP packets, rips out the gateway address, and subs your internal address.
This is not a port issue at all. SIP session set-up works as expected over NAT, but RTP media, in this case voice, will fail.