[Freeswitch-users] Change invite header on inbound call
phenix at vfemail.net
Sun Jun 14 15:28:59 MSD 2015
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
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 )
<action application="transfer" data="1000 XML default" />
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
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