[Freeswitch-users] Change invite header on inbound call

[Tanguy]

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 at default extension is 
ringing but i don't like the SIP invite header

*INVITE sip:1000 at 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



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