I'm new to freeswitch, so I'm sure this is just a newbie configuration error.  Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages.  I've already followed the confluence page on configuring NAT.

My Setup:

Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.

SIP Provider: Flowroute

NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute.  I configured external sip and rtp to use stun entries.  External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.  

I have one DID configured to go directly to my one extension.  My extension can register just fine.  My extension can dial out.  When I call my cell, the call connects and I have 2 way audio.  When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).

When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072

Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

Sofia Status says:

If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue?  I feel like it's just me missing something, I just can't determine what that might be.

Thanks,

  -Brian