Thanks everyone for your help! I think I have it working at this point. Adding --cc "rtp-peer 192.168.3.225" did the trick and I could call into my pbx and listen to a recording. I still need to make sure incoming calls work and ring the amps phone (which I'll attempt this week) but I have no reason to believe it wouldn't work as expected.
Andreas noted correctly that there is no NAT in between my machine and the asterisk server. If we were to suppose that there was a NAT and the asterisk server lived somewhere on the Internet, would I need to take any other considerations aside from using my external IP address in the osmo-cc-sip-endpoint arguments in place of the internal IP? I'm just thinking about a future scenario if I want to take the phone somewhere else for testing.
Thanks,
Mike.