Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:<a href="http://pastebin.freeswitch.org/12632"></a><br><br>Calling an IVR dialplan:<br><a href="http://pastebin.freeswitch.org/12634">http://pastebin.freeswitch.org/12634</a><br>
<br>Calling from one extn to another.<br><a href="http://pastebin.freeswitch.org/12633">http://pastebin.freeswitch.org/12633</a><br>(With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)<br>
<br>For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.<br><br>Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)<br>
<br clear="all">Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote">On Mon, Apr 5, 2010 at 5:21 PM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div style="word-wrap: break-word;">Most likely its in the RTP stream as RFC2833 which is the reason you're not getting anything plus I need a FULL sip trace not this abbreviated trace.<div><br></div><font color="#888888"><div>
/b</div></font><div class="im"><div><br><div><div>On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:</div><br><blockquote type="cite"><span style="border-collapse: separate; font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;">I've taken another stab at this one way audio problem today.<br>
<br>I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all.<br><br>Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing.<br>
<br>Any suggestions of what I should try now?<br><br>Would the dtmf's be sent in the sip packets, or in the rtp?<br><br>To preempt the easy answers and save some time:<br>- ports are open on EC2 config,<span> </span><br>
- iptables turned off for the test,<br>- RTP port range uncommented in switch.conf.xml,<br>- softphone stun was set to<span> </span><a href="http://stun.freeswitch.org/" target="_blank">stun.freeswitch.org</a><br><br clear="all">
Cheers,<br>Fraser</span></blockquote></div><br></div></div></div><br>_______________________________________________<br>
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