<div>Although I can't answer your question from direct know, here are some relevant links;</div><div><br></div>NDLB-force-rport (taken from <a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-force-rport">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-force-rport</a> )<br>
This will force FreeSWITCH to send SIP responses to the network port from which they were received. Use at your own risk! For more information see NAT Traversal.<br><param name="NDLB-force-rport" value="true|safe"/><br>
safe = param that does force-rport behavior only on endpoints we know are safe to do so on. This is a dirty hack to try to work with certain endpoints behind sonicwall which does not use the same port when it does nat, when the devices do not support rport, while not breaking devices that acutally use different ports that force-rport will break<div>
<br></div><div>Also found these;</div><div><a href="http://freeswitch-users.2379917.n2.nabble.com/NDLB-force-rport-safe-td5911932.html">http://freeswitch-users.2379917.n2.nabble.com/NDLB-force-rport-safe-td5911932.html</a></div>
<div><a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2011-November/077600.html">http://lists.freeswitch.org/pipermail/freeswitch-users/2011-November/077600.html</a></div><div><a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085658.html">http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085658.html</a></div>
<div><br></div><div>The last update on this was Brian West was ( <a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/034336.html">http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/034336.html</a> )</div>
<br>Try adding this param to your sofia profile. It will break cisco <br>phones or any other phone that follows the sip spec. This explicitly <br>breaks RFC to accommodate broken phones.<br><param name="NDLB-force-rport" value="true"/> in your sofia profile.<div>
<br></div><div>This information is somewhat out of date and there were some changes to the way NAT works since then, so I'm not 100% sure.</div><div><br></div><div>If anyone else can chime in to confirm (possibly even Brian West himself) that'd be good.</div>
<div><br></div><div>Hope this helps</div><div><br></div><div>Cal<br><div><br></div><div><div><div class="gmail_quote">On Sat, Feb 16, 2013 at 6:48 PM, George Cooper <span dir="ltr"><<a href="mailto:geocooper@gmail.com" target="_blank">geocooper@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">Hi,<div><br></div><div>I just started trying to work with freeswitch using the FS 1.06 book( although I am running FS ver 1.2.6+git.) , FS cookbook,, and wiki. </div>
<div><br></div><div>I and have a question regarding NAT traversal. </div><div><br></div><div>I have been stuck trying to get phones to register and be able to call each other using ext 1000 to 1002 for example. I couldn't get the phones to register until I read an archive regarding polycom NAT issues.</div>
<div><br></div><div>The freeswitch server I'm connecting to is hosted on a remote public IP. I'm using a polycom 335, grandstream xpx 2020, and an x-lite client on a Local 192 network. </div><div><br></div><div>
The below settings seem to have fixed my registration issue.</div><div><br></div><div>Is there a reason that I would NOT set the parameter settings in sip_profiles/internal.xml to true:</div><div><br></div><div><div><!-- use at your own risk or if you know what this does.--></div>
<div> <param name="NDLB-force-rport" value="true"/></div></div><div><br></div><div><br></div><div><br></div><div>and to add the variable "sip-force-contact" for each user in directory/default/1000.xml - 1019.xml ? </div>
<div><br></div><div></div><div><div><include></div><div> <user id="1000"></div><div> <params></div><div> <param name="password" value="$${default_password}"/></div>
<div> <param name="vm-password" value="1000"/></div><div> </params></div><div> <variables></div><div> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/></div>
<div> <variable name="toll_allow" value="domestic,international,local"/></div><div> <variable name="accountcode" value="1000"/></div><div> <variable name="user_context" value="default"/></div>
<div> <variable name="effective_caller_id_name" value="Extension 1000"/></div><div> <variable name="effective_caller_id_number" value="1000"/></div><div> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/></div>
<div> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/></div><div> <variable name="callgroup" value="techsupport"/></div><div> </variables></div>
<div> </user></div><div></include></div></div><div><br></div><div>Sorry if this has been gone over, but I'm not sure if there is something I'm missing because the wiki ( <a href="http://wiki.freeswitch.org/wiki/NAT" target="_blank">http://wiki.freeswitch.org/wiki/NAT</a> ) says "NAT just works" and the internal.xml file says "<!-- use at your own risk or if you know what this does.-->"...</div>
<div><br></div><div>Thanks for any clarity you can provide,</div><div><br></div><div>Geo</div></div><br>_________________________________________________________________________<br>
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