Hello I would like to use a freeswitch server as gateway to share my inbound trunks between several other servers ( asterisk and freeswitch ). My trunks will be connected to the "gateway freeswitch server" On inbound call: some DID numbers should ring the production asterisk server ( or the backup asterisk server if the production peer is not registered ) Some others DID numbers will ring a another test freeswitch server. I created several sip account on internal profile for each remote server 1000( asterisk ) , 1001(asterisk-backup), 1002(freeswitch testing server) and i tried to transfer a call like this ( i did not implemented DID number filtering yet ) *<condition> <action application="transfer" data="1000 XML default" /> </condition>* When i call one of my public numbers the 1000@default extension is ringing but i don't like the SIP invite header *INVITE sip:1000@192.168.0.14:5070;transport=udp;user=phone SIP/2.0* I would like something like this to distinguish the destination number on the remote servers. *INVITE sip:<public_destination_number>@ip_adresss:5060 SIP/2.0* As you can see, my headers may simulate the telco headers. I did not find how to change theses headers on the dialplan Best regards