My thought at this stage is that the ATA is trying to send the ACK directly and not to FreeSWITCH using the Contact header. If I recall correctly, GS's can be configured with both a registrar and an outbound proxy. Try setting the outbound proxy to the FreeSWITCH box, see what happens.<br>
<br><div class="gmail_quote">On 20 March 2012 17:15, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
This is why I love Wireshark so much! Look at this purdee graph it makes:<br><br><img alt=""><br>
<br><br>See all those 200 OK's that your FS is sending to the Grandstream? Guess what your GS is sending in response to those: NADA! If you look at the BYE that FS sends to the GS you'll even see the reason:<br><br>
SIP;cause=408;text=\"ACK Timeout\"<br><br>FS never gets an ACK back from the GS. So the question is: why? I'm unfamiliar with the GS so I'll have to defer to those with more experience than I. However, I think you'll find that tcpdumps and analyzing w/ Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP calls" and then a new dialog opens. In this case it shows two calls - meaning two call legs. Click "Select All" then click "Flow" and you'll get the cool graph. Click around and see what other stuff does. :)<br>
<br>I'm thinking of doing a FreeSWITCH conference call presentation on the subject of collecting pcaps and doing Wireshark analysis. Let me know if you guys think that's a good presentation.<span class="HOEnZb"><font color="#888888"><br>
<br>-MC</font></span><div class="HOEnZb"><div class="h5"><br><br><br>
<div class="gmail_quote">On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster <span dir="ltr"><<a href="mailto:bdfoster@endigotech.com" target="_blank">bdfoster@endigotech.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Andrew, <div><br></div><div><div>root@homeserver:/usr/local/stund# ./client <a href="http://stunserver.org" target="_blank">stunserver.org</a></div><div>STUN client version 0.97</div><div>Primary: Independent Mapping, Independent Filter, preserves ports, will hairpin</div>
<div>Return value is 0x000003</div><div><br></div><div><a href="http://da1.endigovoip.com/dump.pcap" target="_blank">http://da1.endigovoip.com/dump.pcap</a></div><div><br></div><div>Kristian, </div><div><br></div><div><a href="http://pastebin.freeswitch.org/18708" target="_blank">http://pastebin.freeswitch.org/18708</a> <br>
</div><div><br></div><div>Michael,</div><div><br></div><div>I did replace the IP's for security purposes, but now I've realized that it's needed and it's not really that big of a deal. I'll end up changing the Flowroute creds after this is fixed up. The prior siptrace is exactly one call (two legs). I don't think it's a carrier issue, as I've tried calling a buddy's server direct sip with the same issues.</div>
<div><br></div><div>-BDF</div><br><div class="gmail_quote">On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
We have scores of machines behind NAT talking to Flowroute with no problems, so there's got to be something potentially non-obvious but easy that needs to be set/unset. I noticed in the SIP trace that there are several calls. It's hard to know what's what. I think your best bet is a pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I also noticed that you redacted IP addrs - you won't be able to do this with a pcap. If security is an issue then I'd say get the pcap and let us know here on the list, then those who can have a look will email you privately and you can send the pcap file to them.<br>
<font color="#888888">
<br>-MC</font><div><div></div><div><br><br><div class="gmail_quote">On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster <span dir="ltr"><<a href="mailto:bdfoster@endigotech.com" target="_blank">bdfoster@endigotech.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've only set up FS so far on machines with no NAT, so this is sort of a new experience for me.<div><br></div><div>I have a FreeSWITCH server located on the same local network as all of my phones here at the house. When I try to make a call to Flowroute, after about 30 seconds the call drops. It also does the exact same thing when I call a buddy's server directly via SIP.</div>
<div><br></div><div>Here's a siptrace of the call (I didn't think that the actual FS log would be much help):</div><div><a href="http://pastebin.freeswitch.org/18697" target="_blank">http://pastebin.freeswitch.org/18697</a></div>
<div><br></div><div>...and here's a paste of 'sofia status':</div><div><a href="http://pastebin.freeswitch.org/18698" style="font-family:Consolas,'Lucida Console',monospace;font-size:13px;background-color:rgb(240,247,255)" target="_blank">http://pastebin.freeswitch.org/18698</a>
</div><div><br></div><div>...and just for good measure, here's a paste of vars.xml:</div><div><a href="http://pastebin.freeswitch.org/18699" target="_blank">http://pastebin.freeswitch.org/18699</a><span><font color="#888888">
</font></span></div><span><font color="#888888"><div><br></div><div><br></div><div>-- <br>Brian D. Foster<br>Endigo Computer LLC<br>Email: <a href="mailto:bdfoster@endigotech.com" target="_blank">bdfoster@endigotech.com</a><br>
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