I don't have the luck to put the freeswitch on a public IP, and I need the phones behind NAT registering with default context on port 5060 while in the meantime remote phones registered as remote SIP extensions to my freeswitch on port 5080 by public context. In the public context, I will have the outbound and inbound SIP trunks to my service providers as well.<br>
<br>I will try your solution to see if it fits my needs.<br>Thanks,<br>Chris<br> <br><br><br><div class="gmail_quote">On Sun, Apr 27, 2008 at 11:32 AM, Brian West <<a href="mailto:brian@freeswitch.org">brian@freeswitch.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="">The double nat scenario is going to be a fun one to solve. You're better off putting FreeSWITCH on a public IP or fully understanding what is going on. <div>
<br></div><div>The outbound profile isn't for inbound registrations by default. Its almost what you need but you'll need to turn auth-calls=true and you might need to force the registration domain. Do you have phones behind that nat with FreeSWITCH registering to port 5060?</div>
<div><br></div><div><param name="force-register-domain" value="<a href="http://cluecon.com" target="_blank">cluecon.com</a>"/> <-- force the domain on a sip profile.</div><div><br></div><div>
/b</div><div><br></div><div><br></div><div><br><div><div><div></div><div class="Wj3C7c">On Apr 27, 2008, at 10:25 AM, Chris Chen wrote:<br></div></div><blockquote type="cite"><div><div></div><div class="Wj3C7c">Could any of you share your experience with remote SIP extensions behind another NAT?<br>
<br>Basically my setup would be like this:<br><br>[ Remote SIP extensions]------>(private IP@) NAT Router B---(public IP @B)-------(Internet Cloud)-----------(public IP @A) NAT Router A---(private IP @A)------(private IP@C)[Freeswitch]<br>
<br>I have the domain registered as "<a href="http://sip.mydomain.com" target="_blank">sip.mydomain.com</a>"<br><br><br>let's see, I am trying to config the remote SIP extension as 1018 using port 5080 hitting the public context, but I got the error on the freeswitch console<br>
<br>"user <a href="mailto:1018@sip.mydomain.com" target="_blank">1018@sip.mydomain.com</a> not found"<br><br>Could you guys guide me to the right direction for remote SIP extensions' deployment?<br><br>Your helps are appreciate.<br>
Thanks,<br><br>Chris<br><br><br></div></div> _______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org" target="_blank">Freeswitch-users@lists.freeswitch.org</a><br>
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