[OpenSIPS-Users] Rewriting From header using Contact information

[Julian Yap]

I have a scenario where the PSTN to SIP gateway (AudioCodes) I am
using sets the From header to 'anonymous' when it does not receive a
Calling Name from the PSTN side.

The modified INVITE from the gateway then looks like this (changed
some numbers and IP's):
From: "anonymous" <sip:anonymous at anonymous.invalid>;tag=1c49690767.
To: <sip:+18085557005 at 192.168.178.50;user=phone>.
CSeq: 1 INVITE.
Contact: <sip:8084446503 at 192.168.178.60>.

When the gateway does receive the Calling Name from the PSTN, it looks
like this:
From: "HONOLULU     HI" <sip:8083330048 at 192.168.178.60>;tag=1c1248847826.
To: <sip:+18085557005 at 192.168.178.50;user=phone>.
CSeq: 1 INVITE.
Contact: <sip:8083330048 at 192.168.178.60>.

In the first instance, I want to re-write the From header because I do
in fact have the calling number from the Contact header.  In theory,
the PSTN gateway should sort this out for me and not send me the
'anonymous' From header but I've searched the manuals and it doesn't.
Anyone else encountered this?

This is the IF statement that satisfies the criteria:
if($fu=='sip:anonymous at anonymous.invalid' &&
$ct=~"^<sip:[2-9][0-9]{2}[2-9][0-9]{6}@")
{
    xlog("L_INFO", "fix anonymous\n");
}

The final From would be:
From: $ct;$ft
 --> That is Contact header variable;From tag

I had a look at the UAC module but using the function
uac_replace_from(), I don't know how to strip the '<' and '>' of the
Contact header to satisfy the arguments of the function.  Is it
possible?  I'm thinking that using the UAC is preferable to what I'm
proposing below.

Does this method sound sane?  Or is it dangerous?:
remove_hf("From");
append_hf("From: $ct;$ft\r\n");

Thanks,
Julian