<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">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><div>/b</div><div><br><div><div>On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-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; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 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 class="Apple-converted-space"> </span><br>- iptables turned off for the test,<br>- RTP port range uncommented in switch.conf.xml,<br>- softphone stun was set to<span class="Apple-converted-space"> </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></body></html>