[Serusers] 30s calls

[Jerlique Ban]

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