[OpenSIPS-Users] Rewriting From header using Contact information

[Julian Yap]

Yeah, I've tried modifying the RPID header...

It looks like that's working for me.  I had to make up a hard coded
display name...

This is slightly off topic but in my current T1's I need to add in an
Asterisk 'wait' command for the facility IE with the Calling name to
be sent through.

ie. --> http://www.voipinfo.org/wiki/view/CallerID

Is there something similar for OpenSIPS?

- Julian

On Wed, Mar 18, 2009 at 9:49 AM, Brett Nemeroff <brett at nemeroff.com> wrote:
> It's not really standards compliant, but you can do it. I'm not sure why you
> want to take stuff out of the contact header and stick it in from. From
> shouldn't ever be changed, if you can help it. If your really trying to
> change the resultant display name (ie: caller id) then go for manipulating /
> adding RPID headers instead.
> http://www.opensips.org/html/docs/modules/1.4.x/auth.html#append-rpid-hf-no-params
>
> I've had to rewrite "From" headers because non-compliant carriers have
> insisted on using data in the from header for E911 call routing.. idiots..
> :P
> If you insist on changing the From header; this may help.. you won't be able
> to do it the way you are presently trying..
> take a look here:
> http://www.opensips.org/index.php?n=Resources.DocsTipsFaqs
> and here:
> http://www.opensips.org/html/docs/modules/1.4.x/uac.html#id227417
>
> On Wed, Mar 18, 2009 at 2:37 PM, Julian Yap <julianokyap at gmail.com> wrote:
>>
>> I just tested and this does not work:
>> remove_hf("From");
>> append_hf("From: $ct;$ft\r\n");
>>
>> So basically I want to rewrite the From header by using the details
>> from the Contact header.
>>
>> Any suggestions?
>>
>> - Julian
>>
>> On Wed, Mar 18, 2009 at 1:15 AM, Julian Yap <julianokyap at gmail.com> wrote:
>> > 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
>> >
>>
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