Hello I'd like to set things up, where... - FS and local SIP extensions are on a private LAN behind a NAT firewall, - remote employees are allowed to register with Freeswitch so they're part of Freeswitch, - none of the routers support UPnP or NAT-PMP, and - to lower load on the Freeswitch server, I'd like to have RTP packets flow directly between extensions instead of going through Freeswitch, like Asterisk does with its "canreinvite=no" option. However, Freeswitch should remain in the loop somehow so that users can access PBX features during their call. Here's the picture: http://img577.imageshack.us/img577/1795/freeswitchnat.png Apparently, changes to Freeswitch made http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios this page obsolete. If Freeswitch can do this, does someone have a basic, working configuration that I could use to get started? Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Working-configuration-for-FS-b... Sent from the freeswitch-users mailing list archive at Nabble.com.