you might want to include the sip-trace and the whole call setup .<div>Your log shows its getting audio from the phone and autodetects the correct rtp port.</div><div>Do you have both on at once? maybe one already owns your soundcard so it breaks the other, try turning them both off and only start the one.</div>
<div>while the call is up, get a pcap on both sides and see if audio is going to the right place.</div><div><br><br><div class="gmail_quote">On Thu, Jun 30, 2011 at 1:23 AM, Avi Marcus <span dir="ltr">&lt;<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div dir="ltr"><a href="http://pastebin.freeswitch.org/16627" target="_blank">http://pastebin.freeswitch.org/16627</a><div>
I&#39;ve got a Linksys ATA behind NAT and a softphone behind NAT. Both seem to register with the same UDP-NAT string and both have the same type of contact string.. but on one FS is rewriting the RTP IP to work properly - the 1102 works, but on 1000 I don&#39;t seem to be getting any audio.</div>


<div><br></div><div>Is there some hidden parameter I can&#39;t see affecting this? Both are dialing the same extension 9664 default MOH stuff...</div><div>This softphone actually seems to have a responsive gui in linux. If I can get the audio to actually work, that would great!</div>


<div>(Oh, it has tls/srtp too, it seems)</div><div><br></div><div>Thanks!</div><div><a href="http://pastebin.freeswitch.org/16627" target="_blank"></a><div dir="ltr"><span style="font-family:Verdana, Arial, Helvetica, sans-serif"><span style="font-size:small">-Avi Marcus</span><br>


<br></span><div><div style="font-size:13.3px;font-family:Verdana, Arial, Helvetica, sans-serif"><img width="1" height="1"></div>

</div></div><br>
</div></div>
<br>_______________________________________________<br>
Join us at ClueCon 2011, Aug 9-11, Chicago<br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a> 877-7-4ACLUE<br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
Twitter: <a href="http://twitter.com/FreeSWITCH_wire">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:+19193869900<br>
</div>