Hi Joseph, I use the below for similar use case, on an inbound route, call comes in, gets bridged direct to pbx. {sip_invite_to_uri=<sip:${destination_number}@${domain}>}user/12341234@${domain} 12341234 being the ext you register pbx as on FS I have used another that sends call direct to IP, but found if the pbx was behind NAT this way didn’t use the NDLB type NAT fixes, this one still takes them into consideration. From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Joseph Waite Sent: Wednesday, 27 June 2018 10:34 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge Hi Guys Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box. We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this. Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead of what is currently INVITE sip:user@ Im hoping there is something like sip_invite_to_uri, however google is failing me. Regards