Ok. I read the OpenSips doc of force_rport again:
Force_rport() adds the rport parameter to the first Via header. Thus, OpenSIPS will add the received IP port to the top most via header in the SIP
message, even if the client does not indicate support for rport. This enables subsequent SIP messages to return to the proper port later on in a SIP
transaction.
I misunderstand th "received IP port" maybe ;-)
But we didn't have this problems before changing the platform.
The Re-Invite enters here, and as I saw in syslog, the topology_hiding matches
and also the uac_test is be done:
if (has_totag())
{
if (topology_hiding_match())
{
if (nat_uac_test("127"))
{
xlog("L_INFO", "Topology hided, Contact fixed - LF_BASE");
fix_nated_contact();
}
t_relay();
exit;
}
Am 29.04.2016 um 15:56 schrieb Johan De Clercq:
I don;t think so : force_rport just adds the port on which you receive to the
first via header.
2016-04-29 15:36 GMT+02:00 Julian Santer <julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>>:
We are using 2.1.2
So we made a "huge" version update and also changed the kind of working
Am 29.04.2016 um 14:27 schrieb Johan De Clercq:
What version do you use in your new install ?
2016-04-29 13:12 GMT+02:00 Julian Santer <julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>
<mailto:julian.san...@rolmail.net <mailto:julian.san...@rolmail.net>>>:
Hi Johan,
we changed our platform 2 weeks ago from 1 OpenSips Proxy 1.6.4 to
3 OpenSips instances with topology hiding.
Since we didn't have this problems earlier, I think it worked with
the old platform.
As we call force_rport() in the main_route, the private IP in the
VIA header should be ignored, right?
At the moment we call force_rport() in all our instances.
I think, we should call force_rport() only on the edge server
where we make the nat_handling, right?
Kind regards,
Julian Santer
Raiffeisen OnLIne
Am 29.04.2016 um 10:10 schrieb Johan De Clercq:
Indeed.
2016-04-29 9:49 GMT+02:00 Julian Santer <julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>
<mailto:julian.san...@rolmail.net <mailto:julian.san...@rolmail.net>>
<mailto:julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net> <mailto:julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>>>>:
Hi Johan,
as the asterisk is not administrated by us, I have to ask
the customer.
As I understand you think the problem should be the
private IP in the VIA header
and this should be fixed with STUN, right?
Kind regards,
Julian Santer
Raiffeisen OnLine
Am 29.04.2016 um 09:17 schrieb Johan De Clercq:
do you use stun on the phones ?
2016-04-29 9:14 GMT+02:00 Julian Santer
<julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net> <mailto:julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>>
<mailto:julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net> <mailto:julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>>>
<mailto:julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>
<mailto:julian.san...@rolmail.net <mailto:julian.san...@rolmail.net>>
<mailto:julian.san...@rolmail.net <mailto:julian.san...@rolmail.net>
<mailto:julian.san...@rolmail.net
<mailto:julian.san...@rolmail.net>>>>>:
Hi guys,
we are using OpenSips 2.1.2 with topology_hiding.
Now we got trouble with Re-Invites, which are routed to
the private IP from the "from".
The topology_hiding matches, but the Re-Invite is sent to
the private IP from the "from".
Have we to call lookup again on the
registrar/core, when we got a Re-Invite?
You can find the trace under
http://siptrace.rolbox.net/siptrace.html
Kind regards,
Julian Santer
Raiffeisen OnLine
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