<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#FFFFFF">
Hello<br>
<br>
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"<br>
<br>
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.<br>
<br>
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 )<br>
<br>
<b><tt><condition> <br>
<action application="transfer" data="1000 XML default"
/><br>
</condition></tt></b><br>
<br>
When i call one of my public numbers the 1000@default extension is
ringing but i don't like the SIP invite header<br>
<br>
<b><tt>INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:1000@192.168.0.14:5070;transport=udp;user=phone">sip:1000@192.168.0.14:5070;transport=udp;user=phone</a>
SIP/2.0</tt></b><br>
<br>
I would like something like this to distinguish the destination
number on the remote servers.<br>
<br>
<tt><b>INVITE sip:<public_destination_number>@ip_adresss:5060
SIP/2.0</b></tt><br>
<br>
As you can see, my headers may simulate the telco headers.<br>
<br>
I did not find how to change theses headers on the dialplan<br>
<br>
Best regards<br>
<br>
<br>
<br>
</body>
</html>