[Serusers] 30s calls

Hello, I have configured SER and asterisk to allow me to make calls to the PSTN network, however on my voip phone (behind NAT) I am having issues with the voip tx audio dropping out after 30 seconds. Now I'm guessing it's a nat issue but even that doesn't really make sense!! Why, well because the only the TX of the voip phone drops out (ie the PSTN phone cannot hear what is said on the voip phone). The PSTN phone can still transmit audio to the voip phone (through the nat). Anyway in SER, I have set the natping_interval to 5 seconds, and this still doesn't resolve the issue. Strangely at the time that the audio disconnects Asterisk is sending my phone an INVITE message. Why would it do this mid call? I'm using SER0.9.0+Asterisk as my platform. Any pointers?? JB