Hi I've stumbled upon the following links: http://freeswitch-users.2379917.n2.nabble.com/effective-callee-id-name-behav... http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-November/005355.ht... Am I correct to say that these set the callee_id_name to be passed in the SIP INVITE when configured on the extension? We require the following INVITE format to be sent over a specific gateway: INVITE sip:1234567890@192.168.200.153;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.130:5070;branch=z9hG4bK75b28f842f76d9c9e299a80c0;rport;X-DispCookie=1013;X-HwTRC=259446;X-DptMsg=1405 Route: <sip:192.168.200.153:5060;transport=udp;lr> Call-ID: bb4bcb34ec7041237fbfc1e63f687744@10.18.5.64 From: "0987654321"<sip:0987654321@192.168.200.130;transport=udp;user=phone>;tag=fd64e901-CC-1013-TRC-259446 To:"1234567890"<sip:1234567890@192.168.200.153;transport=udp;user=phone> How could one go about to achieve the above? Thanks for any help. Kind Regards Deon Vermeulen