<div dir="ltr"><div>Hi,<br><br><br></div><div>Strange part that my LB receives invite with Route header:<br><br><pre>U 2015/07/22 10:46:19.531380 <a href="http://10.101.241.55:5060">10.101.241.55:5060</a> -> <a href="http://10.101.237.133:5060">10.101.237.133:5060</a>
INVITE sip:<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">11 at 10.50.210.9</a>:50365 SIP/2.0.
Record-Route: <sip:10.101.241.55;lr;nat=yes>.
Record-Route: <sip:10.101.237.133;lr;nat=yes>.
Via: SIP/2.0/UDP 10.101.241.55;branch=z9hG4bK9f82.ae0b52341a6aa41d8f28e7f65ed9fd55.0.
<b>Route: <sip:<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">loadbalancer at 10.101.237.133</a>;lr;received=sip:<a href="http://10.50.210.9:50365">10.50.210.9:50365</a>></b>.
Via: SIP/2.0/UDP 10.101.237.133;rport=5060;branch=z9hG4bK9f82.68623c2d6329f689d3ab43fdd3cf83c5.0.
Via: SIP/2.0/UDP 192.168.5.148:55529;received=10.50.210.9;branch=z9hG4bK.jWzhtRBsW;rport=6574.
From: <sip:<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">22 at voipdev.mydomain.net</a>>;tag=AZWk8e87s.
To: sip:<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">11 at voipdev.mydomain.net.</a>
CSeq: 21 INVITE.
Call-ID: LJoKbYM0YY.
Max-Forwards: 69.
Supported: outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 577.
Contact: <sip:<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">22 at 10.50.210.9</a>:6574>;+sip.instance="<urn:uuid:c1c7dbe5-7dac-4d3f-a496-af9fde38a0e7>".
User-Agent: LinphoneAndroid/2.4.1 (belle-sip/1.4.1).
Path: <sip:<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">loadbalancer at 10.101.237.133</a>;lr;received=sip:<a href="http://10.50.210.9:6574">10.50.210.9:6574</a>>.
.
v=0.
o=22 4026 1869 IN IP4 10.101.254.33.
s=Talk.
c=IN IP4 10.101.254.33.
b=AS:380.
t=0 0.
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics.
m=audio 46696 RTP/AVP 96 97 98 99 0 8 101 100 102.
a=rtpmap:96 opus/48000/2.
a=fmtp:96 useinbandfec=1; stereo=0; sprop-stereo=0.
a=rtpmap:97 SILK/16000.
a=rtpmap:98 speex/16000.
a=fmtp:98 vbr=on.
a=rtpmap:99 speex/8000.
a=fmtp:99 vbr=on.
a=rtpmap:101 telephone-event/48000.
a=rtpm</pre>But instead of sending this Invite to: received=sip:<a href="http://10.50.210.9:50365">10.50.210.9:50365</a> it sends back to Kamailio Proxy(10.101.241.55), even if I have following line in my config.<br></div><div><br>modparam("path", "use_received", 1)<br><br></div><div>Maybe somebody have any idea why?<br><br></div><div>Thank you in advance!<br></div><div><br></div><div>With kind regards,<br><br></div><div>Jurijs<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-07-21 19:18 GMT+03:00 Daniel Tryba <span dir="ltr"><<a href="mailto:d.tryba@pocos.nl" target="_blank">d.tryba@pocos.nl</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On Tuesday 21 July 2015 17:59:53 Jurijs Ivolga wrote:<br>
> So basically extension 100 sends invite to LB, LB then send invite to<br>
> Kamailio server and Kamailio server replies with 407 error.<br>
><br>
> Do you have any ideas why?<br>
<br>
No, you'll have to debug this the usual way. Run kamailio at a highe debug<br>
level to see what it is doing and log queries to the database. You'll have to<br>
make captures to see how the flow op sip messages is and who is requiring<br>
authentication.<br>
<br>
> Maybe you can give me more details about config on Kamailio server behind<br>
> LB?<br>
<br>
The config is equal to standard in this regard. Show us your config for the<br>
backend regarding authentication and registration.<br>
<br>
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</blockquote></div><br></div>