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