Hi, Problem I want to solve: Allow internal registrations to FreeSWITCH behind NAT AND for external authenticated registrations to FreeSWITCH. I think this means using two different profiles. I'm using the default one and one I've called "outside". There's a page that touches on the second half but I need a bit more help with it. https://freeswitch.org/confluence/display/FREESWITCH/External+Profile There's basically three changes I've made: created a new sip_profile; created a new directory called outside.xml; created a new dialplan called outside.xml sip_profile changes for external5090.xml (like suggested on the page): 1. Deleted gateway registrations 2. updated sip-port to 5090 3. Updated the context to a new one "outside" directory/outside.xml: 1. Use domain sip.example.com 2. Use group name outside 3. Create a user and put it at directory/outside/ with the user_context of outside dialplan/outside.xml: 1. Use context name ouside The problem: User 2602 can register to sip.example.com:5090 and dial 3500 user 2600 can register on the LAN and call 3500. Both have bidirectional audio. If user 2602 calls user 2600, the phone immediately disconnects on answer on 2600 end, and no audio is heard. A 481 Call Does Not Exist is generated and I don't know why. SIP trace here: https://pastiebin.com/5f38e341e3fd8 Another thing I noticed is sofia status says sip.example.com is an alias for internal. Is that right? https://pastiebin.com/5f38e3b0276d3 Thanks for any pointers, Sean