Hi Bogdan,
Could you confirm what i have done here is correct please?
route[5] {
if (search("From:.*<sip:090[0-9]{5}")) {
uac_replace_from("anonymous","sip:anonymous@1.1.1.1");
remove_hf("Remote-Party-ID");
append_hf("Remote-Party-ID: \"Anonymous\"
<sip:anonymous@anonymous.invalid>;party=calling;screen=yes;privacy=full\r\n");
rewritehostport("gatewayip:5060");
route(1);
}
I have used uac_replace_from just incase sip header not contain rpid.
Regards,
Howard
On 2/8/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
Hi Howard,
in this case is simple. If RPID hdr is used you don not have to change
the FROM hdr anymore.
To replace the RPID hdr, use remove_hf() see textops module to remove
the hdr and append_hf() to add a new one.
regards,
bogdan
Howard Tang wrote:
Hi Bogdan,
After I check the sip message, I think the problem is
Remote-Party-ID: 1234
<sip:1234@111.111.111.111>;screen=yes;party=calling
I can see the uac_replace_from has changed the From Header, but i
think the problem was caused by the Remote-Party-ID on the Linksys ATA
Is there a way to change this Remote-Party-ID as well?
Regards,
Howard
On 2/7/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
> yes :)
>
> regards,
> bogdan
>
> Howard Tang wrote:
> > HI Bogdan,
> >
> > do you means using " ngrep -t -W byline port 5060 " ?
> >
> > regards,
> > Howard
> >
> >
> > On 2/7/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
> >> Hi Howard,
> >>
> >> could please post the network trace of a non-working call?
> >>
> >> regards,
> >> bogdan
> >>
> >> Howard Tang wrote:
> >> > HI All,
> >> >
> >> > I have a problem with uac_replace_from() function not working
for
> >> > Linksys Unit.
> >> >
> >> > This is what i have in the route[1],
> >> >
> >> > if (search("From:.*<sip:900*")) {
> >> >
> uac_replace_from("anonymous","sip:anonymous@x.x.x.x");
> >> > }
> >> >
> >> > I have 0900xx as internal sip account, then I have to remove the
> >> > 0900xx callerid before i send the call to PSTN gateway. Because
> 0900xx
> >> > is not a valid number, Mobile phone providers block the call
> because
> >> > of that, as a result, I need to replace the callerid to
anonymous.
> >> >
> >> > I have tested with many other ATA and X-Lite without problem.
The
>>
> problem only ocurrs when i use Linksys ata. I have monitored the
SIP
>> > message, the first request will change to anonymous then back to
the
>> > original username.
>> >
>> > Anyone have an idea on how i can fix this ?
>> >
>> >
>>
>>
>