Hello, I am new to FreeSWITCH and having a headache configuring it. My experiences to Asterisk do not work completely. I appreciate your suggestion. I am trying to setup a central SIP server which accepts registers from subsidiary SIP servers which is behind NAT and has no fixed IP address. Each subsidiary servers have extension number ranges. I have to route such extension ranges to each SIP servers. FS central server <--- (NAT) --- Server A (ext. 1000 - 1099) <--- (NAT) --- Server B (ext. 1100 - 1199) <--- (NAT) --- Phone ext. 1201 <--- (NAT) --- Phone ext. 1202 According to the wiki page http://wiki.freeswitch.org/wiki/Clarification:gateways , I created an XML file under "conf/directory/default" for a gateway setting inside a <user> block. In my configuration, the gateway setting is parsed in the profile "internal". My current problem is the dialplan "bridge" application does not work to bridge to the destination like "sofia/gateway/server-a/1001". The call terminates after printing a log message like "[NOTICE] sofia.c:5082 Hangup sofia/internal/1001 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]". What is my mistake? Is it wrong to use the "gateway" feature for such purpose? sgk