<p>I'm going to go ahead and mark this issue as solved. The solution was to do a make current. There was a bug in FreeSWITCH that was fixed before Anthony went on vacation. </p>
<p>If anyone was on conference or if your name is Brian West, if there was a specific revision that this was fixed on please chime in for anyone else that may have this issue.</p>
<p>Fwiw I actually went ahead and started with a clean install (due to deleting the source folder and also not knowing what all I messed with to try and solve this issue). After I did the install, both the external and internal profile were getting the local ipv4 address. To fix this, I went to conf/sip_profiles/external.xml and changed the ext-sip-ip and ext-rtp-ip from $${local_ip_v4} to stun:<a href="http://stun.freeswitch.org">stun.freeswitch.org</a>. After restarting the profile, the external profile was getting my external IP and my internal profile was getting the local IP. </p>
<p>Thanks to everyone on the conference and on the thread for helping out. Brian, for repayment, I'll get you that pic you've always wanted lol.</p>
<p>-BDF</p>
<div class="gmail_quote">On Mar 21, 2012 8:43 AM, "Andrew Cassidy" <<a href="mailto:andrew@cassidywebservices.co.uk">andrew@cassidywebservices.co.uk</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I think we also need to check those variables on the internal profile. As said previously it looks like it's doing the tun on the internal profile which is why the ack is being send to the wrong place.<br><br><div class="gmail_quote">
On 21 March 2012 06:09, Fernando - NextBilling IP Solutions <span dir="ltr"><<a href="mailto:fernandojdk@gmail.com" target="_blank">fernandojdk@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<div>What you have in your external profile on variables ext-rtp-ip end ext-sip-ip?</div></div>
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<div style="FONT-FAMILY:Times New Roman"><font>Best Regards,</font></div>
<div style="FONT-FAMILY:Times New Roman"><font></font></div>
<div style="FONT-FAMILY:Times New Roman"><font><strong>Fernando da Silva Santos</strong></font></div>
<div style="FONT-FAMILY:Times New Roman"><font color="#004080">NextBilling IP Solutions LTDA</font></div>
<div style="FONT-FAMILY:Times New Roman"><font>Phone: <a href="tel:%2B55%2021%202143-9000" value="+552121439000" target="_blank">+55 21 2143-9000</a></font></div>
<div style="FONT-FAMILY:Times New Roman"><font>MSN: <a href="mailto:fernandojdk@gmail.com" target="_blank">fernandojdk@gmail.com</a></font></div>
<div style="FONT-FAMILY:Times New Roman"><font><a href="http://www.nextbilling.com.br/" target="_blank">www.nextbilling.com.br</a></font></div>
<div style="FONT-FAMILY:Times New Roman"><font>Rio de Janeiro, Brazil, BR</font></div>
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<div style="FONT-FAMILY:Times New Roman"><strong><font color="#666666"></font></strong></div>
<div style="FONT-FAMILY:Times New Roman"><strong><font color="#666666">Importante:</font></strong><span style="LINE-HEIGHT:normal;TEXT-TRANSFORM:none;FONT-VARIANT:normal;FONT-STYLE:normal;TEXT-INDENT:0px;LETTER-SPACING:normal;BORDER-COLLAPSE:separate;WHITE-SPACE:normal;WORD-SPACING:0px"><span style="LINE-HEIGHT:normal;TEXT-TRANSFORM:none;FONT-VARIANT:normal;FONT-STYLE:normal;TEXT-INDENT:0px;LETTER-SPACING:normal;BORDER-COLLAPSE:separate;WHITE-SPACE:normal;WORD-SPACING:0px"><span style="LINE-HEIGHT:normal;TEXT-TRANSFORM:none;FONT-VARIANT:normal;FONT-STYLE:normal;TEXT-INDENT:0px;LETTER-SPACING:normal;BORDER-COLLAPSE:separate;WHITE-SPACE:normal;WORD-SPACING:0px"><span style="LINE-HEIGHT:normal;TEXT-TRANSFORM:none;FONT-VARIANT:normal;FONT-STYLE:normal;TEXT-INDENT:0px;LETTER-SPACING:normal;BORDER-COLLAPSE:separate;WHITE-SPACE:normal;WORD-SPACING:0px"><span style="LINE-HEIGHT:normal;TEXT-TRANSFORM:none;FONT-VARIANT:normal;FONT-STYLE:normal;TEXT-INDENT:0px;LETTER-SPACING:normal;BORDER-COLLAPSE:separate;WHITE-SPACE:normal;WORD-SPACING:0px"><font size="4" face="Helvetica"> </font></span></span></span></span></span></div>
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<div style="font-size:medium;font-family:Helvetica;font-weight:normal"><span><span style="FONT-FAMILY:Tahoma"><span style="LINE-HEIGHT:27px;FONT-FAMILY:Webdings;COLOR:green;FONT-SIZE:18pt">P</span></span><span style="FONT-FAMILY:Tahoma"><strong><span style="LINE-HEIGHT:11px;FONT-FAMILY:Arial,sans-serif;COLOR:teal;FONT-SIZE:7.5pt"> "</span><span style="LINE-HEIGHT:11px;FONT-FAMILY:Arial,sans-serif;COLOR:green;FONT-SIZE:7.5pt">Antes de imprimir pense em seu compromisso com o Meio Ambiente."</span></strong></span></span></div>
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<div><i>-------Original Message-------</i></div>
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<div style="FONT-SIZE:11pt" dir="ltr"><i><b>From:</b></i> <a href="mailto:freeswitch@zoho.com" target="_blank">dingdong</a></div>
<div style="FONT-SIZE:11pt" dir="ltr"><i><b>Date:</b></i> 21/03/2012 02:57:02</div>
<div style="FONT-SIZE:11pt" dir="ltr"><i><b>To:</b></i> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">FreeSWITCH Users Help</a></div><div>
<div style="FONT-SIZE:11pt" dir="ltr"><i><b>Subject:</b></i> Re: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds</div></div></div>
<div> </div><div><div>
<div style="FONT-FAMILY:Verdana,Arial,Helvetica,sans-serif;FONT-SIZE:10pt">id like to know more bout wireshark and sip debugging,please include this on the conf call topics<br>
<div><br>---- On Tue, 20 Mar 2012 10:15:20 -0700 <b>Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></b> wrote ---- <br></div><br>
<div>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.<br><br>-MC<br><br><br>
<div>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>
<div>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>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>
<div>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>
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<div><br><br>
<div>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>
<div>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" 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">
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<div><br></div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Andrew Cassidy BSc (Hons) MBCS<div>Managing Director; Cassidy Web Services Ltd</div><div>T: 03300 100 960 F: 03300 100 961</div><div>E: <a href="mailto:andrew@cassidywebservices.co.uk" target="_blank">andrew@cassidywebservices.co.uk</a></div>
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