No RTP from asterisk?

I’m running 1.4.30-RC2, freshly buit and reinstalled with asterisk-gui.

I can register lines, I’m using u-law (G711) as the codec, and I’ve got 2 SIP trunks and PRI connected.

This is a lab installation, so I can play with it at will.

I can terminate calls outbound, in order to terminate inbound, I need to do some call
routing on the switch * is connected to via SIP trunk, but that’s not really the problem.

When I place a call, the call comes in. No audio (and RTP setup shows up in debug), but it seems like no packets pass.

dumpcap (a component of current wireshark distributions) verified, no RTP at all.

Initially I thought this was related to the fact that G729 isn’t licensed,
so I removed G729 as an option in both the Trunks and user configuration,
while this gave me progress in the right direction according to debug on *,
I’m still not getting RTP or an audio stream.

Can anyone help me troubleshoot this?