**resending as I am not sure if this made it out the first time, I do not believe it did, if this is a duplicate -- my apologies.**<br><br>Hello,<br><br>   As always, thank you for all / any help and input you may provide in advance.<br>
<br>Call Scenario:<br><br>UA1
-&gt; REGISTRAR-01 -&gt; Kamailio-01 -&gt; Asterisk (New Call-ID +
Asterisk in Media Path) -&gt; Kamailio-01 -&gt; REGISTRAR-02 -&gt; UA2<br>
<br>UA1 is behind NAT<br>UA2 is behind NAT<br><br>The purpose of this
is when using a shared &quot;USRLOC&quot; database to simulate calls from &quot;PSTN&quot;
to generate both legs of the call, i.e. incoming and outgoing, and also
allow for easier / cleaner &quot;traversal&quot;<br>
<br>This aids from scenario&#39;s happening where UA1 calls UA2 (while UA1
exists on P1 and UA2 exists on P2) this prevents P1 -&gt; UA2, and
forces P2 -&gt; UA2<br><br>We determine that this is a call from P1 to P2 (internal call) and thus create this &quot;bridge / interconnection&quot;<br>
<br>We are running into a problem it seems with one way audio, i.e. the
CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE.<br><br>REGISTRAR-01 AND REGISTRAR-02 are both &quot;proxying&quot; RTP<br><br>
As well as the initial Asterisk in &quot;the middle&quot; SDP.<br><br>Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong.<br><br>Perhaps there are better ways to go about this, let me know if I am way off course, thank you!<br>
<br>Sincerely,<br>Brandon Armstead<br>