[Freeswitch-users] Change invite header on inbound call

Tanguy 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 
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 )

<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;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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150614/0e516229/attachment.html 

Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list