Hi, 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. I and have a question regarding NAT traversal. 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. 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. The below settings seem to have fixed my registration issue. Is there a reason that I would NOT set the parameter settings in sip_profiles/internal.xml to true: <!-- use at your own risk or if you know what this does.--> <param name="NDLB-force-rport" value="true"/> and to add the variable "sip-force-contact" for each user in directory/default/1000.xml - 1019.xml ? <include> <user id="1000"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> Sorry if this has been gone over, but I'm not sure if there is something I'm missing because the wiki ( http://wiki.freeswitch.org/wiki/NAT ) says "NAT just works" and the internal.xml file says "<!-- use at your own risk or if you know what this does.-->"... Thanks for any clarity you can provide, Geo