Why is it that FROMUSER & FROMDOMAIN settings in a peer definition in SIP.CONF are IGNORED in SIP INVITE Messages?
This I have in the SIP.CONF:
[sip-out]
type=peer
fromuser=
fromdomain=
username=
secret=
host=
port=5060
outgoingproxy=
call-limit=10
context=
qualify=yes
insecure=invite,port
nat=no
disallow=all;
allow=alaw;
allow=ulaw;
When looking at the SIP INVITE Asterisk is sending, the users Caller ID is getting used in the FROM field, instead of the FROMUSER / FROMDOMAIN.
The SIP REGISTER is correctly building the FROM header using the values, the SIP INVITE not any more.
I went throught the source code of the SIP channel to understand what may overwrite FROMUSER / FROMDOMAIN and was unable to find the solution.
In the dialplans there is no single CallerID being set.
What may be the reason for this?
Any input appreciated, as this behaviour keeps me away from using certain SIP Services.
Asterisk 1.4.13 on CentOS 5