Wrong RTP port in SDP

[PV22892]

Hello Everyone,
I'm not sure if this is the best place to ask, but here goes.
My Scenario:
Softphone -> FreeSWITCH -> Internet -> Asterisk
FreeSWITCH Domains:

Internal: 172.17.220.201
External: 83.240.xxx.xxx
Global

The issue occurs on outbound calls: My router or FreeSWITCH isn't updating the SDP for the RTP port, resulting in an RTP port mismatch. This leads to one-way or no audio, as the actual RTP packets are sent from a different source port than what's advertised in the SDP (likely due to NAT port remapping).

I have this configured in two outbound routes in my dialplan:

<extension name="call_direction-outbound" continue="true" uuid="aa37cf43-6efc-483f-8ff9-e3feb80fd4eb">
  <condition field="${user_exists}" expression="false"/>
  <condition field="${call_direction}" expression="^$"/>
  <condition field="destination_number" expression="^(100)$">
    <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets" inline="true"/>
    <action application="export" data="call_direction=outbound" inline="true"/>
  </condition>
</extension>

<extension name="Devscope.number" continue="" uuid="fdc47941-9d66-4086-85cb-ffbb1c3c2552">
  <condition field="${user_exists}" expression="false"/>
  <condition field="destination_number" expression="^(100)$">
    <action application="export" data="call_direction=outbound" inline="true"/>
    <action application="unset" data="call_timeout"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
    <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
    <action application="set" data="inherit_codec=true"/>
    <action application="set" data="ignore_display_updates=true"/>
    <action application="set" data="callee_id_number=$1"/>
    <action application="set" data="continue_on_fail=1,2,3,6,18,21,27,28,31,34,38,41,42,44,58,88,111,403,501,602,607,809"/>
    <action application="bridge" data="sofia/gateway/07143571-a044-4cb3-9650-d52267267b09/$1"/>
  </condition>
</extension>

And here's my gateway configuration (I can't check the exact XML right now, but this should be close):

<include>
  <gateway name="Devscope">
    <param name="from-domain" value="135.236.xxx.xxx" />
    <param name="proxy" value="135.236.xxx.xxx" />
    <param name="realm" value="135.236.xxx.xxx" />
    <param name="expire-seconds" value="800" />
    <param name="register" value="false" />
    <param name="retry-seconds" value="30" />
    <param name="context" value="83.240.xxx.xxx" />
    <param name="distinct-to" value="false" />
    <param name="extension-in-contact" value="false" />
    <param name="enabled" value="true" />
    <!-- Description: Devscope SIP Trunk -->
    <!-- Domain (tenant): 172.17.220.201 -->
  </gateway>
</include>

SIP Log:
From FreeSWITCH SDP:
c=IN IP4 83.240.xxx.xxx
t=0 0
m=audio 20432 RTP/AVP 9 0 8 101

From Asterisk SDP:
c=IN IP4 135.236.108.143
t=0 0
m=audio 13868 RTP/AVP 9 0 8 101

I've attached a screenshot of the relevant packet capture/log for more details:

Any help or suggestions would be greatly appreciated! I've tried settings like disable_rtp_auto_adjust and symmetric RTP on the Asterisk side, but no luck so far.

Thank you.