@BraswellJay said in What determines the interface that SIP RTP streams over ...:

@BraswellJay

I finally figured out what was causing my issue.

In FreePBX under Settings->Asterisk SIP Settings there is a parameter to set the External Address used. I had this still set to the public IP of the original interface. Once I changed that parameter to the public IP of the new interface then all started to work over the new interface as expected.

Glad you sorted it out. Good work!

Made no configuration changes to the firewall tonight. Shutdown FreePBX VM, made a new one, and stuff seems to be working as it should. I'll do a few more tests tomorrow to make sure all is well.

Edit: I lied. I made a new DHCP reservation for my new FreePBX server.

@RamblingBiped said in SIP Phone registers no audio...:

Here is the guide I followed to get the Yealink T23G to work with OpenVPN: http://www.sunstatetechnology.com/docs/YealinkOpenVPNGuide.pdf

When I disable NAT on the extension the phone is unable to register.

Something is wrong there, that sounds like the VPN is not working as there is no NAT when there is a VPN.

We use an international blocking trunk provider. It's great peace of mind.