One way audio problem

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Asterisk is making a call to a peer. In 200 ok, peer is sending its
application server ip in contact field, so asterisk sends ACK to that IP.
RTP starts flowing between endpoints and peer plays an IVR and asks for
destination number. After entering destination number peer’s application
server sends INVITE again with different media IP and asterisk accepts with
200 ok. RTP from peer comes from new media IP but asterisk keep sending to
their old media IP that came in their 200 ok before and don’t send to new
one. Hence, I can hear their voice but they can’t.

Does anyone know how to make asterisk send RTP to new media IP that came in
new INVITE within the call?