Howard,
it looks fine, but you really do not have to alter the from if your GW supports RPID hdr. FROM mangling is something very expensive from processing point of view and you should avoid it if possible.
you can get rid of the uac_replace_from() - the RPID will be added all the time (even if it was not originally present).
regards, bogdan
Howard Tang wrote:
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@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@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@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 ? > >