Hmmmm seeing that port 127.0.0.1….9060
looks like to capture and forward to homer (I know I
installed it at one time to try to some triage).
Maybe homer built for additional checks.
I just disabled the module hosting that port and I no longer
see it in ngrep.
my path issue is wss:// webrtc over to uac.reg and FW it
off to PBX. So ill try and figure out where in kama.cfg
to hopefully get it to fw to pbx and make call.
From:
Johnny
Ritzer <sudoritz@gmail.com>
Date: Thursday, September 10, 2020 at 4:32 PM
To: Kamailio (SER) - Users Mailing List
<sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] sngrep nor tcpdump showing
5060 traffic but ngrep /kama logs show something ?
(outbound call)
Ya I have other scenarios that show up
like inbound call.
So I did this (made a test call) and saved
output ngrep to pcap.
And open in wireshark and I see it as
protocol UDP with SIP enclosed in data
ngrep -d any -qt -W byline port 5060 -O test.pcap
heres ngrep
2020/09/10
23:19:51.109093 10.111.0.4:5060 -> 127.0.0.1:9060 #1
.......[F...
...INVITE
sip:8000@24.xx.xx.xx SIP/2.0.
Via:
SIP/2.0/WSS 4d8asfsdfag3f.invalid;branch=z9hG4bK2797686.
Max-Forwards:
70.
To:
<sip:8000@24.xx.xx.xx>.
From:
"Ctx-WEBSIP-1138"
<sip:1138@24.xx.xx.xx>;tag=jd85ifa0f2.
Call-ID:
9pujji6vasdfasfrbn2.
CSeq: 906
INVITE.
Contact:
<sip:1138@24.xx.xx.xx;gr=urn:uuid:121211-a79d-4e76-a010-121211>.
Allow:
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER.
Supported:
gruu, outbound.
User-Agent:
SIP.js/0.7.8.
Content-Type:
application/sdp.
Content-Length:
2047.
You sure you’re using 5060 on kamailio?
No…
or not that I know of
This
is a temp GCP (as im building a POC to get this to
a working concept.) trying to fine-tune config
Are
you using vlans? In the past I had issues with
ngrep if the packets had vlan tags on them.
I
can see kama logs./ngrep when I try to call
out (WSS -> ) . But I cant see anything
in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im
not sure how/why or it could be something
simple but what puzzles me is I know im
sending data but its not captured.
Setup
is pbx at 24.xx.xx.xx but ext is uac.reg to
stay connected.
UA
client = sip.js / wss
Inbound works(but 1 way audio ill fix
after outbound issue)
Outbound
doesn’t dial out (UA = webRTC client) using
wss. Which led me to troubleshoot via
sngrep/tcpdump.
But
I seet his when it happens knowing I ame
getting something sent.
Sep 10 21:48:36
kamaProx /usr/sbin/kamailio[27480]:
WARNING: <script>:
----RouteLogic--SIP request received on
port 4443: from:1138 to:8040
srcip:70.xx.xx.xx
Sep 10 21:48:36
kamaProx /usr/sbin/kamailio[27480]: INFO:
<script>: START: INVITE from
sip:1138@24.xx.xx.xx (IP:
70.xx.xx.xx:55041)
Sep 10 21:48:36
kamaProx /usr/sbin/kamailio[27480]:
WARNING: <script>:
----RouteLogic--SIP request received on
port 4443: from:1138 to:8040
srcip:70.xx.xx.xx
Sep 10 21:48:36
kamaProx /usr/sbin/kamailio[27480]: INFO:
<script>: START: ACK from
sip:1138@24.xx.xx.xx (IP:
70.xx.xx.xx:55041)
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