In that case you would need a sip proxy in place to rewrite the packets for the nat issue.<br>There&#39;s nothing else we can really do.&nbsp; We have a way to do what you want but you are using it under<br>circumstances we can&#39;t control.&nbsp; <br>
<br><br><br><div class="gmail_quote">On Wed, Feb 18, 2009 at 1:46 PM, Justin Miller <span dir="ltr">&lt;<a href="mailto:fax@virgintechnologies.com">fax@virgintechnologies.com</a>&gt;</span> 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><font color="#0000ff" face="Verdana" size="2">Anyone else have any ideas on this?</font></div><div><div></div><div class="Wj3C7c">
<blockquote style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px; margin-right: 0px;"><font face="Tahoma" size="2">-----Original Message-----<br><b>From:</b> <a href="mailto:fax@virgintechnologies.com" target="_blank">fax@virgintechnologies.com</a> [mailto:<a href="mailto:fax@virgintechnologies.com" target="_blank">fax@virgintechnologies.com</a>]<br>
<b>Sent:</b> Tuesday, February 17, 2009 05:45 PM<br><b>To:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br><b>Subject:</b> Re: [Freeswitch-users] Sending media streams to a media gateway<br>
<br></font>
<div><font color="#0000ff" face="Verdana" size="2">I looked at that, but I think that will cause issues with the NAT traversal.&nbsp; Our phones will all be in external networks.&nbsp; I forgot to mention that.</font></div>
<blockquote style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px; margin-right: 0px;"><font face="Tahoma" size="2">-----Original Message-----<br><b>From:</b> Anthony Minessale [mailto:<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>]<br>
<b>Sent:</b> Tuesday, February 17, 2009 05:34 PM<br><b>To:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br><b>Subject:</b> Re: [Freeswitch-users] Sending media streams to a media gateway<br>
<br></font>you could set the variable bypass_media to true before you call bridge<br><br>&lt;action application=&quot;set&quot; data=&quot;bypass_media=true/&gt;<br>&lt;action application=&quot;bridge&quot; data=&quot;sofia/internal/someuser@somehost&quot;/&gt;<br>
<br>that will negotiate a point to point media connection between the caller and callee<br><br><br>
<div class="gmail_quote">On Tue, Feb 17, 2009 at 5:51 PM, <span dir="ltr">&lt;<a href="mailto:fax@virgintechnologies.com" target="_blank">fax@virgintechnologies.com</a>&gt;</span> 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 bgcolor="#ffffff">
<div><font face="Verdana" size="2">I have Freeswitch running successfully with a fairly basic config.&nbsp; Nat traversal is working well on both the client and server side.&nbsp; I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only.</font></div>

<div><font size="2">I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts.&nbsp; Where can I start with this?&nbsp; Does anyone have any example configs?</font></div>

<div><font size="2">Justin</font></div></div><br>_______________________________________________<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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