Get a console log with a sip trace. Connect to your server with fs_cli and then issue this command:<div><br></div><div>sofia global siptrace on</div><div><br></div><div>Then make a test call capturing all the output. Repeat for each soft phone. Put this info on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a> and use &quot;FreeSWITCH Log&quot; as the syntax highlighting. Put the pastebin (pb) URL in this email thread and hopefully the gang here will be able to help you diagnose what&#39;s going on.</div>
<div><br></div><div>In the meantime your homework is to read up on a few things:</div><div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Reporting_Bugs">http://wiki.freeswitch.org/wiki/Reporting_Bugs</a></div><div>
<a href="http://wiki.freeswitch.org/wiki/Packet_Capture">http://wiki.freeswitch.org/wiki/Packet_Capture</a></div><div><a href="http://wiki.freeswitch.org/wiki/Nat">http://wiki.freeswitch.org/wiki/Nat</a></div><div><br></div>
<div>The information in those wiki pages will serve you well in your FreeSWITCH hacking. :)</div><div><br></div><div>-MC<br><br><div class="gmail_quote">On Mon, Oct 31, 2011 at 11:44 AM, Julien Chavanton <span dir="ltr">&lt;<a href="mailto:jchavanton@gmail.com">jchavanton@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br><div class="gmail_quote"><span style="font-family:verdana,sans-serif">Hi, I have audio problems when bridging calls from different sofia profiles.</span><br style="font-family:verdana,sans-serif">
<span style="font-family:verdana,sans-serif">I use 2 profiles in order to connect softphone through a VPN to avoid NAT problem</span><br style="font-family:verdana,sans-serif">

<span style="font-family:verdana,sans-serif">One profile is sending/listening on a private IP (for softphone) and the other profile is using a public IP (for external PSTN gateways)</span><br style="font-family:verdana,sans-serif">


<span style="font-family:verdana,sans-serif"></span><br style="font-family:verdana,sans-serif"><span style="font-family:verdana,sans-serif">I have tested with X-lite and Ekiga and I face different one way audio with both:</span><br style="font-family:verdana,sans-serif">


<span style="font-family:verdana,sans-serif">With X-lite the audio received from the private IP is not forwarded </span><br style="font-family:verdana,sans-serif"><span style="font-family:verdana,sans-serif">While with Ekiga the audio from the public IP profile is not forwarded</span><br style="font-family:verdana,sans-serif">


<br style="font-family:verdana,sans-serif"><span style="font-family:verdana,sans-serif"><span style="font-family:verdana,sans-serif">I have traced SIP and RTP and the signaling looks good. RTP is received by FS on the correct socket but not forwarded.<br>


<br>I understand I do not provide enough information to diagnostic, but maybe I could get input on how to  diagnostic RTP socket etc. in this scenario ?<br style="font-family:verdana,sans-serif"></span><br><br></span>
</div><br>
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