Hi Daniel,
I added xlog and $mb here is the debug:
I still cant figure out how to overwrite the From field.
Call flow:
SP1 -> SIP call --> 5061 Kamailio --> DNS resolution -> Remote client, I dont see pstn.parzee.io I see sip.sp1.com DOESNT WORK
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
if($fd=~"sip\.xxxxxx\.com") {
xlog("L_DBG","$mb \n| Masking call from: $fu");
$fd = "pstn.parzee.io";
$fu = "sip:" + $fU + "@pstn.parzee.io";
}
xdbg("New branch [$T_branch_idx] to: $ru from: $fu $fd\n");
xlog("L_DBG","$mb \n| New branch \n");
route(NATMANAGE);
}
Hello,
for the case when it doesn't work, do you see the xlog message printed?
Cheers,
Daniel
On 21/12/2016 08:37, Gonzalo Gasca Meza wrote:
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/k0j
Z3aDE Doesnt work: http://pastebin.com/p19r
wcrn
On Tue, Dec 13, 2016 at 4:50 AM, Daniel-Constantin Mierla <miconda@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" <sip:+18888888888@sip.sp1.com> to
From: "+18888888888" <sip:+18888888888@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 list sr-users@lists.sip-router.org http://lists.sip-router.org/cg i-bin/mailman/listinfo/sr-user s -- Daniel-Constantin Mierla www.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@lists.sip-router.org http://lists.sip-router.org/cg i-bin/mailman/listinfo/sr-user s -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com