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 "FreeSWITCH Log" 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'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"><<a href="mailto:jchavanton@gmail.com">jchavanton@gmail.com</a>></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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