Hi,
Thanks for the reply.
Manually insert header field Record-Route by
insert_hf(Record-Route: sip:106.x.x.x\r\n);
And, change IP in SDP
if (has_body(application/sdp)) {
replace_all(IN IP4 [0-9]\.[0-9]\.[0-9]\.[0-9], 106.x.x.x);
}
Seems work with WIFI behind one level NAT, not test 3G/4G.
Nice ngrep invocation. ;)
On 26/06/2015 19:26, Terrance Devor wrote:
ngrep -d eth0 -qt -W byline portrange 5060
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Attach SIP signalling pls.
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ngrep -d eth0 -qt -W byline portrange 5060
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Hi everyone,
I have a public IP(106.x.x.x), a OpenSIPS(192.168.1.x) and two SIP
UAC(192.168.2.x/192.168.3.x), the OpenSIPS and UACs are all behind NAT but
not the same NAT.
I configure my route to use the public IP, and forward all packets to my
OpenSIPS.
1. The UACs register to 106.x.x.x, and
in last days i change opensips port to 5090
now
i want change 5090 port to default port 5060
i try it , so zoiper ip phone can not translate voice and X_light eyebeam
worked ok
please help me
--
View this message in context:
Dear,
By the both of the last versions of opensips 1.8 and 1.10, there is a
same problem,
the clients behind of NAT don't have media if the port of opensips is 5060.
by changing the port to 5090 or any other ports everything is ok.
Could you please let me where my fault is?
--
Best
Hi,
I've configured OpenSIPs using Nathelper module and rtpproxy. the problem
I'm facing is when I try to register my softphone, it got registered but as
I issue the command opensipsctl ul show, in contact header the IP is private
not public. The configuration of OpenSIPs is listed down below;
Hi Ahmed,
check the following things:
1) you do fix_nated_register() before save(location)
2) the received_avp param has the same value in registrar and nathelper
module
3) you configured the nat_bflag param in usrloc module and you are
setting it before save(location)
Regards,
Bogdan
Problem using Nat helper (Laszlo)
--
Message: 1
Date: Fri, 30 Apr 2010 08:35:00 +0200
From: Laszlo las...@voipfreak.net
Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
To: OpenSIPS users mailling list users
...@voipfreak.net
Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
To: OpenSIPS users mailling list users@lists.opensips.org
mailto:users@lists.opensips.org
Message-ID:
r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
Hi,
Thanks for supporting me, really appreciated your help.
Date: Mon, 03 May 2010 12:39:55 +0300
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4bde99eb.9090
, 29 Apr 2010 19:34:16 -0300
From: Antonio Anderson Souza anto...@voicetechnology.com.br
Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID:
s2o285c24cc1004291534m1deec8c4zb6c4ddb003311...@mail.gmail.com
Content
: [OpenSIPS-Users] NAT Problem using Nat helper
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID:
s2o285c24cc1004291534m1deec8c4zb6c4ddb003311...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Ahmed,
Could you send an wireshark trace to the list
Hi,
I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm using
is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
sofphone, they got authenticated and authorized by radius and got
registered sucessfully. Even I made calls between two softphone
sucessfully(Can
Ahmed,
Could you send an wireshark trace to the list? It will be easier to check
what's going wrong.
Besta regards,
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:
Hi,
I've configured OpenSIPs with
Hi Khan,
You can start with 2 simple checks:
1) be sure your force_rtp_proxy() functions are triggred both for
request and reply - put some xlog to see if you get there in the script
2) check the messages with SDP (on the outgoing part) if they have the
rtpproxy indication in SDP
Regards,
Hey everyone,
I have been trying to work this for a long time, this mailing list is
my last resort. I have applied NAT traversal using RTP proxy. My
scenario is as follows:
UAC1 (behind NAT) --- UAC2 (behind NAT)
The UAC's get authenticated fine, call establishes but there is no
voice, neither i
Hi Bogdan
Thank you for your help.
The nated client does register to opensips. It is set to register every
3600 sec, min time is 20 s and max time is 1800 s. It is default xLite
setting.
Here is the 200OK I captured from my nated client box:
!'DVVEGTeEd=3SIP/2.0 200 OK
Via: SIP/2.0/UDP
Hi Juan,
I need to see the request part also to figure out if the flow through
the NAT is ok or not.
As a side note - could you check if the device behind the nat is
actually receiving the 200 OK?. Because a typical reason for a missing
ACK is a missing 200 OK.
Another question - the device
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