Hi,
Should I try something like
nat_uac_test(diff-ip-src-contact || private-contact || diff-ip-src-via ||
diff-port-src-via)?
Any ideas?
Regards,
Ronald
December 15, 2023 at 4:38 PM, r...@rvgeerligs.nl wrote:
>
> Hi
>
> I use opensips 3.4 and I have NAT problems with 2 devices
Hi
I use opensips 3.4 and I have NAT problems with 2 devices behind the same NAT
(called party hears nothing).
The A party is softphone on iPhone (linphone) the B (called)party is
Polycom310. The other way around works (Polycom calls linphone).
Actually I tested this in two different NAT
Thanks for the responses. They helped me exclude some things. I've managed
to make progress and pinned down the lack of audio to a misconfiguration of
Mediaproxy. Two-way audio through double-nat / firewall is working but goes
silent after about 60 seconds connected and Asterisk kills the
Check out what IPs are offered in the SDPs in asterisk. Make sure they’re
both public IPs.
If you only have 1 asterisk, forwarding the rtp port range configured in
asterisk from the firewall to asterisk should do it.
On Thu, 14 Jan 2021 at 08:23, Mark Allen wrote:
> Thanks Adrian
>
> The
Thanks Adrian
The firewall has SIP-ALG disabled and just forwards ports from externally
to where they need to be internally - so ports 5060 and 1 - 65535 of
46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box)
On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu wrote:
> Google search for SIP
Google search for SIP ALG problem to see if this is relevant for your case.
Regards,
Adrian
> On 13 Jan 2021, at 13:08, Mark Allen wrote:
>
> Hi all - I've been banging my head against this but not succeeding.
>
> Our setup...
>
> UAC 192.168.x.x
> |
> Router5.x.x.x
>
Firewall is not sip aware, rtprelay via box in dmz
Outlook voor iOS<https://aka.ms/o0ukef> downloaden
Van: Users namens Mark Allen
Verzonden: Wednesday, January 13, 2021 5:08:27 PM
Aan: OpenSIPS users mailling list
Onderwerp: [OpenSIPS-Users] OpenSI
Hi all - I've been banging my head against this but not succeeding.
Our setup...
UAC 192.168.x.x
|
Router5.x.x.x
|
(internet)
|
Firewall 46.x.x.x maps
| directly to
OpenSIPS 192.168.x.x Mid-registrar
|
Asterisk 192.168.x.x
Current
Hi All,
Can anyone send a sample configuration of OpenSIPS which behind NAT and
doing connects to clients on TLS and forwards to Softswitch on UDP?
--
___
Users mailing list
Users@lists.opensips.org
I didnt troxlinux, I forced all my traffic through rtpproxy
Atenciosamente.
Eng.° Rodrigo Ferreira
ITIL v3 Certified
http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901
2013/10/23 troxlinux xserverli...@gmail.com
Hi, I have the same problem, did you solve?
2013/10/4 Rodrigo Ferreira
You can set a flag on usr_preferences to force the nat to that customer.
2013/10/23 troxlinux xserverli...@gmail.com
Hi, I have the same problem, did you solve?
2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com
I think I found the problem ..
Looking at my SIP Messages, the VIA and the
Hi, I have the same problem, did you solve?
2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com
I think I found the problem ..
Looking at my SIP Messages, the VIA and the Contact headers doesnt have my
INVALID IP, it shows me my VALID IP.
But I dont know how to set that, im doing the fixes
You can set a flag on usr_preferences to force the nat to that customer so
you can manage this on your dialplan
if the user cannot be recognized over nat help you can force.
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Users mailing list
Users@lists.opensips.org
Thnk mike, an example where I can watch this?
2013/11/5 Mike Tesliuk m...@ultra.net.br
You can set a flag on usr_preferences to force the nat to that customer so
you can manage this on your dialplan
if the user cannot be recognized over nat help you can force.
You can use the avpops module
avp_db_load($fu,$avp(678));
This will load the preferences from the user on avp(678) , so you
check the value and force the use of rtpproxy/mediaproxy as you do
when the user is behind proxy.
you can use the avp_db_query also
avp_db_query(select value from
I'm running out of ideas ..
My rtpproxy is fine
Oct 4 09:10:02 opensips /sbin/opensips[5019]: INFO:rtpproxy:rtpp_test: rtp
proxy udp:127.0.0.1:7890 found, support for it enabled
Oct 4 09:10:02 opensips /sbin/opensips[5020]: INFO:rtpproxy:rtpp_test: rtp
proxy udp:127.0.0.1:7890 found, support
I think I found the problem ..
Looking at my SIP Messages, the VIA and the Contact headers doesnt have my
INVALID IP, it shows me my VALID IP.
But I dont know how to set that, im doing the fixes for nated contacts.
Atenciosamente.
Eng.° Rodrigo Ferreira
ITIL v3 Certified
Yes I did Mike,
and my SIP messages are ok.
I will take a look at your tutorial.
tks
Atenciosamente.
Eng.° Rodrigo Ferreira
ITIL v3 Certified
http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901
2013/10/3 Mike Tesliuk m...@ultra.net.br
Did you try to made some debug rodrigo ? maybe
That howto is just a sample (with a lot of comments) to better understand
of nat configuration (over my understand offcourse), so, you can check and
compare with your configuration to identify about something missing
2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com
Yes I did Mike,
and my
I did that Mike ..
my nat_uac_client isnt passing in any verification ...
I did this ..
if ( nat_uac_test(1) ) xlog(UAC TEST = 1);
if ( nat_uac_test(2) ) xlog(UAC TEST = 2);
if ( nat_uac_test(4) ) xlog(UAC TEST = 4);
if ( nat_uac_test(8) ) xlog(UAC TEST = 8);
well, probably you softphone/ip phone, is using some kind of stun or other
kind of nat features, so, nothing come to be detected, this can happen, so,
if you will be ever using nat, you can force the rtpproxy without nat
detection, this will solve your problem, if you read the documentation (
Forcing the traffic through RTPPROXY worked, but why isnt working the
nat_uac_test?
Kinda weird
Atenciosamente.
Eng.° Rodrigo Ferreira
ITIL v3 Certified
http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901
2013/10/4 Mike Tesliuk m...@ultra.net.br
well, probably you softphone/ip phone,
probably if the UA's are on the same network, when you send the package the
package is going with the external ip on the SDP , when the call is
stablished probably you router is not allowing to open the second lag
because the UA's are trying to stablish from inside using the outside ip
addressl,
Hi guys,
After a long time without using Opensips (almost a year) I tried to install
the opensips 1.10 and everything went well BUT when I make a call, there's
no audio, I know that is something because of NAT, but I have the nathelper
and rtpproxy configuration on my opensips.cfg.
There's
Did you try to made some debug rodrigo ? maybe some rule is missing on your
route script
i made a tutorial over version 1.9 that you can check
[portugues]
http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy
[english]
Have you installed and started rtpproxy? if not just scroll through this
website http://www.rtpproxy.org/.
Regards,
Qasim
On Fri, Aug 3, 2012 at 2:27 AM, Ashish Kundu kash...@gmail.com wrote:
Opensips is a great product, but I have been having problem in configuring
the nat traversal +
Opensips is a great product, but I have been having problem in configuring
the nat traversal + rtpproxy with opensips and have spent about a week on
this. I am a novice in this... when opensips runs with the following
opensips.cfg relevant portions -- it raises the following rtpproxy problem:
Hello all,
I have some problems with OpenSIPS and OpenIMS due to NAT configuration.
My setup is a follow:
UA -- HGW (embedded both OpenSIPS and NAT stuff) -- P-CSCF (OpenIMS)
I used different UA (mainly Twinkle and X-Lite). The HGW (Home GateWay)
is running under OpenWRT on which I compile and
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