<div dir="ltr"><div><div><div><div><div><div><div><div><div>Here's the Freeswitch log showing the SIP traffic:<br><br><a href="http://pastebin.freeswitch.org/21253">http://pastebin.freeswitch.org/21253</a><br><br></div>
The summary of IP addresses are:<br><br></div>10.1.1.19 - External Kamailio<br></div>10.1.1.206 - External FS interface<br></div>10.10.10.206 - Internal FS interface<br></div>10.10.10.207 - Internal Kamailio<br><br></div>
Call was made from 2005 (10.1.254.25) to 1008 (10.10.10.165).<br><br></div>Note that <a href="http://10.1.254.0/24">10.1.254.0/24</a> is routable to <a href="http://10.1.1.0/24">10.1.1.0/24</a><br><br><br><br></div>Here's the tcpdump sample taken on both interfaces on Freeswitch<br>
<br><a href="http://pastebin.freeswitch.org/21254">http://pastebin.freeswitch.org/21254</a><br><br></div>Let me know if anymore logs/debugs are required.<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Thu, Aug 1, 2013 at 1:42 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</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>I'd start by getting a debug log of the call and posting it on <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a>. Hopefully that will yield some clues.<br></div>-MC<br>
</div><div class="gmail_extra">
<br><br><div class="gmail_quote"><div><div class="h5">On Wed, Jul 31, 2013 at 7:33 PM, Peter <span dir="ltr"><<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div class="h5">
<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><font color="#888888"><br>
<br>Peter<br><br>
</font></span></div>
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