<div dir="ltr">After checking everything for the umpteenth time, I discovered I was missing a route back to <a href="http://10.1.254.0/24">10.1.254.0/24</a><br><br>Once I put that in, bingo, audio!<br></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Thu, Aug 1, 2013 at 12:33 PM, Peter <span dir="ltr"><<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div><div><div><div><div><div><div><div><div><div>I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.<br><br></div>The set up is as follows:<br>
<br><br><br></div>Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)<br><br><br></div>Using Freeswitch 1.2.12 on CentOS (installed via RPM)<br><br></div>
Freeswitch has two interfaces:<br><br></div>external - 10.1.1.206<br></div>internal - 10.10.10.206<br><br></div>Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.<br>
<br></div>The SIP negotiation is working as I can call either Linphone client. <br><br></div>I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.<br>
<br></div>I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio. <br><br></div>Any pointers on how to track down the issue?<br><br>Thanks<span class="HOEnZb"><font color="#888888"><br>
<br>Peter<br><br>
</font></span></div>
</blockquote></div><br></div>