[SR-Users] Rewrite From domain

Gonzalo Gasca Meza gascagonzalo at gmail.com
Wed Dec 21 08:37:26 CET 2016


Hi Daniel,

Thanks for the advise, I'm using the following configuration,

# Manage outgoing branches

branch_route[MANAGE_BRANCH] {


        if($fd=~"sip\.sp1\.com") {

           xlog("L_INFO","|Masking SP1 call from: $fu");

           $fd = "pstn.parzee.io";

        }

        xdbg("new branch [$T_branch_idx] to: $ru from: $fu\n");

        route(NATMANAGE);

}

I have two scenarios which are very similar:

1) PSTN call -> SP1 --> VXML -> Call forward to --> 5061 Kamailio --> DNS
resolution -> Remote client, I do see pstn.parzee.io WORKS

2)  SP1 -> SIP call --> 5061 Kamailio --> DNS resolution -> Remote client,
I dont see pstn.parzee.io I see sip.sp1.com DOESNT WORK

I have attached the traces, destination is the same, all calls are SIP TLS
in both call legs, any suggestion tu turn on higher debug level to see SIP
messages.

Traces:

Works: http://pastebin.com/k0jZ3aDE

Doesnt work: http://pastebin.com/p19rwcrn





On Tue, Dec 13, 2016 at 4:50 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

> Hello,
>
> as alternative to assigning to $fd, you can use uac_replace_from()
> exported by uac module.
>
> The best place to do updates to headers for outgoing traffic is in a
> branch_route block.
>
> Cheers,
> Daniel
>
> On 13/12/2016 12:05, Gonzalo Gasca Meza wrote:
>
> Hi all,
>
> I'm using Kamailio to forward calls between 2 Service Providers and I need
> to rewrite the From header "domain" URI.
>
> Example:
>
> From: "+18888888888 <(888)%20888-8888>" <sip:+18888888888 at sip.sp1.com> to
>
> From: "+18888888888 <(888)%20888-8888>" <sip:+18888888888@*sip.sp2.com
> <http://sip.sp2.com>*>
>
> *Call flow:*
>
> Phone A --- > SP1 ---> sip ----> (kamailio) SP2 --(LOCATION)-> Phone B
>
> When Phone A calls SP2 PhoneB, it contains original sip domain from sp1. (
> sip.sp1.com) hence user in SP2 can see call comes from SP1. I would like
> to rewrite the From domain field in this conditions:
>
> a) Calls comes from "sip.sp1.com" AND
>
> b) Call is being routed to PhoneB.
>
> Right now Im using the following code to find user and send call to B.
>
> #!ifdef WITH_ALIASDB
>
>         # search in DB-based aliases
>
>         xlog("L_INFO","alias_db_lookup: Call received. $rU\n");
>
>         if(alias_db_lookup("dbaliases")) {
>
>                 route(SIPOUT);
>
>         }
>
> #!endif
>
>
> I found this in documentation:
>
> $fd - From URI domain
>
> if($hdr(From)=~"sip.sp1\.com") {
>  ...}
>
> But not sure where is the best place to overwrite the From URI domain
> header.
>
> Thanks
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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