Hello,

the request has public IP address in Via and Contact, matching the source IP. If you look at the tests you do in the readme of the nathelper module, then the request appears as not being natted, see:

- http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.f.nat_uac_test

Test with 3 is 1 + 2 and test with 19 is 1 + 2 + 16

Cheers,
Daniel

On 14/07/15 11:56, Alberto Sagredo wrote:

I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on this phones

Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind nat so NATMANAGE is not called

Here its some trace. ANy clue how to handle this phones if they activate STUN for example

U 80.26.x.x:52768 -> 192.168.0.170:8002

INVITE sip:2@x.x.x.x:8002 SIP/2.0.

Via: SIP/2.0/UDP 80.26.x.x:52768;branch=z9hG4bK358742535;rport.

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1478444456.

To: <sip:2@x.x.x.x:8002>.

Call-ID: 244257786-52768-56@IA.CG.BIE.BCH.

CSeq: 550 INVITE.

Contact: "Anonymous" <sip:212@80.26.x.x:52768>.

X-Grandstream-PBX: true.

Max-Forwards: 70.

User-Agent: Grandstream GXP2140 1.0.4.23.

Privacy: id.

P-Preferred-Identity: <sip:212@x.x.x.x:8002>.

Supported: replaces, path, timer.

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.

Content-Type: application/sdp.

Accept: application/sdp, application/dtmf-relay.

Content-Length: 335.

.

v=0.

o=212 8000 8000 IN IP4 80.26.x.x.

s=SIP Call.

c=IN IP4 80.26.x.x.

t=0 0.

m=audio 55422 RTP/AVP 0 8 18 9 2 101.

a=sendrecv.

a=rtpmap:0 PCMU/8000.

a=ptime:20.

a=rtpmap:8 PCMA/8000.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=no.

a=rtpmap:9 G722/8000.

a=rtpmap:2 G726-32/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-15.

# Caller NAT detection route

route[NATDETECT] {

#!ifdef WITH_NAT

force_rport();

if (nat_uac_test("19)) {

if (is_method("REGISTER")) {

fix_nated_register();

} else {

fix_nated_contact();

}

setflag(FLT_NATS);

}

#!endif

return;

}
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