This article explains binding an address for Asterisk/FreePBX with more than one NIC.
In one example, a DID (phone number) was directed to an IVR (interactive voice response), and when the caller would call in, there would be no audio for the caller. If you dialed the extension, the call would go in queue and eventually the phone would ring; however, there would be one-way audio and only the person receiving the call would hear the caller. What made this more interesting to troubleshoot was changing the inbound route from the IVR to a ring group and there would be two-way audio. After support carefully reviewed PCAP files from the router, it was determined audio and signaling were being done on different private IP addresses than the phone provisioning.
At that time FPBX-2.10.0 with Asterisk 22.214.171.124 was running.
The easiest way to edit the bind address is to go in to the settings menu and look for Asterisk SIP Settings. Before continuing down, ensure your NAT settings are correct since a misconfiguration here can also cause an issue.
You will now want to look for the Advanced general settings towards the end of the page. You may enter only one IP address in the bind address field. In my case, it's 192.168.8.22 as this is the IP I use to provision the phones with. Save and apply the changes and you should now have two-way audio.
If you want to do this in the CLI, it appears that FreePBX indicates you will need to edit sip_custom_post.conf. It might be possible to edit sip_additional.conf but there will be a message indicating NOT to.