In Asterisk, I have a "SIP Proxy", with 2 sip trunk. An external Asterisk dials in SIP-1 and asterisk, thru the dialplan, sends the call to SIP-2 (an ISTP) In Freeswitch, I don't know if I should use directory/user for "SIP-1" or if it is allowed to use a gateway inside internal sip_profile. Is it possible authenticate user by IP instead of user/password? Thanks, Valter