CUCM - FreePbx SIP trunk media bypass not working

Hello,

I have a SIP trunk between CUCM and FreePBX server. My phone registers to CUCM and CUCM routes the calls out the SIP trunk to FreePBX, which has SIP trunks with external voice provider. I can make inbound and outbound calls successfully. However, the RTP/media traffic is asymmetric. RTP from phones goes directly to external voice provider, but RTP from external voice provider comes to FreePBX server first and does not go to phone directly.

I see CUCM is sending phone IP address in the SIP/SDP to FreePBX, but FreePBX does not forward the phone IP address to external voice provider in SIP/SDP as connection address. Instead it sends it own address. How do I make Asterisk forward the phone IP address that it receives from CUCM to external voice provider, so that the RTP is between phone and the external voice provider and does not have to go through FreePBX?

My CUCM SIP trunk configuration on FreePBX is below:

type=peer
trustrpid=yes
session-timers=refuse
rfc2833compensate=yes
qualifyfreq=60
qualify=yes
port=5060
nat=no
insecure=invite,port
host=192.168.25.13
dtmfmode=rfc2833
context=from-internal
canrevinvite=yes
allow=all

Thanks!!

I’m reasonably certain both of those are wrong. I could be wrong - I’m getting old and my memory isn’t what it used to be.