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"><<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</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"><a href="http://pastebin.freeswitch.org/16627" target="_blank">http://pastebin.freeswitch.org/16627</a><div>
I'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't seem to be getting any audio.</div>
<div><br></div><div>Is there some hidden parameter I can'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>
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