you only need the params on profiles that accept inbound registration.<br>you can also try using a dedicated profile for outbound reg and disable all the nat options on it.<br><br>The nat mode will cause all sip messages to be delivered to the return port that the invite came in on regardless of what the sip message may say we should do.<br>
<br><br><div class="gmail_quote">On Fri, Jul 11, 2008 at 2:53 PM, Ivan C Myrvold <<a href="mailto:ivan@myrvold.org">ivan@myrvold.org</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style="">Yes, there is nat involved. And I have those options enabled, in the internal profile, as the calls are coming in on port 5060.<div>Maybe I should set up the server outside the firewall, to avoid this.</div>
<div><br></div><div>Ivan</div><div><br><div><div>Den 11. juli. 2008 kl. 20:51 skrev Anthony Minessale:</div><br><blockquote type="cite"><div><div></div><div class="Wj3C7c">it's sending it to the wrong place and giving up.<br>
<br>is there nat involved? <br><br>if not, edit your sip profile and comment <br>aggressive-nat-detection and apply-nat-acl both from your config<br><br>otherwise if you don't have those options set, try enabling them instead.<br>
The same problem happens if you are missing the options when you need them as when you have the options and you don't need them so try both.<br><br><br><br><div class="gmail_quote">On Fri, Jul 11, 2008 at 1:30 PM, Ivan C Myrvold <<a href="mailto:ivan@myrvold.org" target="_blank">ivan@myrvold.org</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div>I updated to latest svn, but exactly the same is happening now. This is a SIP trace from before I restart FreeSWITCH:<div>
<a href="http://pastebin.freeswitch.org/4796" target="_blank">http://pastebin.freeswitch.org/4796</a> </div><div><br></div><div>And here after I restart it:</div><div><a href="http://pastebin.freeswitch.org/4797" target="_blank">http://pastebin.freeswitch.org/4797</a></div>
<div><br></div><div>I have done this several times now, and I see that FreeSWITCH always sends several <span style="color: rgb(51, 51, 51); font-size: 11px;">SIP/<span style="color: rgb(255, 0, 0);">2.0</span> <span style="color: rgb(255, 0, 0);">200</span> OK<span style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px;"> </span></span></div>
<div>and 0.5 seconds after the last one, FreeSWITCH sends a <span style="color: rgb(51, 51, 51); font-size: 11px;">BYE <a>sip:Unknown@</a><span style="color: rgb(255, 0, 0);">81.201</span><span style="color: rgb(255, 0, 0);">.82</span><span style="color: rgb(255, 0, 0);">.28</span> SIP/<span style="color: rgb(255, 0, 0);">2.0<span style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px;"> message.</span></span></span></div>
<div><br></div><div>I was wrong in my earlier message in that it was the other end who sends the bye message, it is in fact FreeSWITCH who sends it.</div><div>You can see that from the traces.</div><div><br></div><div>This is a subscribed DID from Voxbone, and I can redirect it to another URL if you want, and have my wife call the did number (she is on holiday in Romania, where the DID number is subscribed) and see if the same happens (maybe bridge to some music, so my wife doesn't have to say anything). </div>
<div><br></div><div>Ivan</div><div><br></div><div><br><div><div>Den 9. juli. 2008 kl. 23:39 skrev Brian West:</div><div><div></div><div><br><blockquote type="cite"><div>Can you update to the latest your may rev's behind?<div>
<br><div>On Jul 9, 2008, at 4:29 PM, Ivan C Myrvold wrote:<br><blockquote type="cite"><div>I should add that after 1 day, the same happens again. So I have to restart FreeSWITCH at least once a day.<div><br></div> <div>Ivan</div>
<div><br><div><div>Den 9. juli. 2008 kl. 23:16 skrev Ivan C Myrvold:</div><br><blockquote type="cite"><div>My FreeSWITCH is behind nat, and I have calls coming in from internet on port 5060. This works great after I got the IP range set in acl.conf.xml.<div>
<br></div><div>But the calls get hung up after 40-50 seconds into the call for no reason at all. But after I restart FreeSWITCH, this annoying behaviour doesn't happen.</div><div>I traced both cases, when the call is hung up here:</div>
<div><a href="http://pastebin.freeswitch.org/4784" target="_blank">http://pastebin.freeswitch.org/4784</a> </div><div>It looks like FreeSWITCH is resending <span style="color: rgb(51, 51, 51); font-size: 11px;">SIP/<span style="color: rgb(255, 0, 0);">2.0</span> <span style="color: rgb(255, 0, 0);">200</span> OK<span style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px;"> for some reason, until the other end gives up and sends a <span style="color: rgb(51, 51, 51); font-size: 11px;">BYE <a>sip:Unknown@</a><span style="color: rgb(255, 0, 0);">81.201</span><span style="color: rgb(255, 0, 0);">.82</span><span style="color: rgb(255, 0, 0);">.28</span> SIP/<span style="color: rgb(255, 0, 0);">2.0</span></span></span></span></div>
<div><br></div><div>But after a restart, FreeSWITCH behaves much better, and the call goes on until one of the parties hang up normally:</div><div><a href="http://pastebin.freeswitch.org/4785" target="_blank">http://pastebin.freeswitch.org/4785</a> </div>
<div><br></div><div>I am not sure if this is a bug, or a misconfiguration of FreeSWITCH</div><div><br></div><div><span style="color: rgb(51, 51, 51); font-size: 11px;">FreeSWITCH Version <span style="color: rgb(255, 0, 0);">1.0</span>.trunk <span style="color: rgb(102, 204, 102);">(</span><span style="color: rgb(255, 0, 0);">8900</span><span style="color: rgb(102, 204, 102);">)</span></span></div>
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