[Spce-user] Need help for Sip trunk setup

Sat Sep 21 07:10:46 EDT 2019

Hello,

for Debugging here is a short cheatlist 😊

/var/log/ngcp/api.log

API logs providing type and content of API requests and responses as well as potential errors

/var/log/ngcp/panel.log /var/log/ngcp/panel-debug.log

Admin Web UI logs when performing operational tasks on the ngcp-panel

/var/log/ngcp/cdr.log

mediation and rating logs, e.g. how many CDRs have been generated and potential errors in case of CDR generation or rating fails for particular accounting data

/var/log/ngcp/kamailio-proxy.log

Overview of SIP requests and replies between lb, proxy and sems processes. It’s the main log file for SIP overview

/var/log/ngcp/kamailio-lb.log

Overview of SIP requests and replies along with network source and destination information flowing through the platform

/var/log/ngcp/sems.log

Overview of SIP requests and replies between lb, proxy and sems processes

/var/log/ngcp/rtp.log

rtpengine related log, showing information about RTP communication

Example of kamailio-proxy.log line:

Nov 19 00:35:56 sp1 proxy[7475]: NOTICE: <script>: New request on proxy - M=REGISTER R=sip:sipwise.local

F=sip:jdoe at sipwise.local T=sip:jdoe at sipwise.local IP=10.10.1.10:5060 (127.0.0.1:5060) ID=364e4676776621034977934e055d19ea at 127.0.0.1 UA='SIP-UA 1.2.3.4'

The above line shows the SIP information you can find in a general line contained in /var/log/ngcp/kamailio-*:

*        M=REGISTER : The SIP Method

*        R=sip:sipwise.local : The SIP Request URI

*        F=sip:jdoe at sipwise.local : The SIP From header

*        T=sip:jdoe at sipwise.local : The SIP To header

*        IP=10.10.1.10:5060 (127.0.0.1:5060) : The source IP where the message is coming from. Between brackets it is shown the local internal IP where the message come from (in this case Load Balancer)

*        ID= <mailto:364e4676776621034977934e055d19ea at 127.0.0.1> 364e4676776621034977934e055d19ea at 127.0.0.1 : The SIP CallID.

*        UAIP=10.10.1.10 : The User Agent source IP

*        UA=SIP-UA 1.2.3.4 : The SIP User Agent header

In order to collect the full log related to a single call, it’s necessary to "grep" the /var/log/ngcp/kamailio-proxy.log using the ID= string, for example:

# cat /var/log/ngcp/kamailio-proxy.log

# cat /var/log/ngcp/kamailio-lb.log

# tail -f /var/log/ngcp/kamailio-proxy.log | zgrep --line-buffered 'SEARCHTERM '

# tail -f /var/log/ngcp/kamailio-lb.log | zgrep --line-buffered 'SEARCHTERM '

# cat  /var/log/ngcp/kamailio-proxy.log | zgrep -i 'SEARCHTERM'

# cat  /var/log/ngcp/kamailio-lb.log | zgrep -i 'SEARCHTERM '

Increase Debug Level:

Kamailio LB

# ngcp-kamctl lb fifo corex.debug 1

# ngcp-kamctl lb fifo corex.debug 2

# ngcp-kamctl lb fifo corex.debug 3

Kamailio Proxy

# ngcp-kamctl proxy fifo corex.debug 1

# ngcp-kamctl proxy fifo corex.debug 2

# ngcp-kamctl proxy fifo corex.debug 3

Show FiFo Commands:

Kamailio LB

# ngcp-kamctl lb fifo system.listMethods

Kamailio Proxy

# ngcp-kamctl proxy fifo system.listMethods

SIP Package Inspection:

# ngrep-sip

# ngrep -s0 -Whyline -d any -O /root/filename.pcap port 5062 or port 5060

btw: for SIP call debugging also check out Sipcapture Homer 7: 

https://github.com/sipcapture/homer-installer/blob/master/homer_installer.sh

I think this will be enough for the beginning 😉

Von: Spce-user <spce-user-bounces at lists.sipwise.com> Im Auftrag von Tomi Hakkarainen
Gesendet: Samstag, 21. September 2019 12:29
An: Kamran Asghar <kamran.asghar at live.com>
Cc: spce-user at lists.sipwise.com
Betreff: Re: [Spce-user] Need help for Sip trunk setup

Hi Kamran,

I normally use sngrep on sipwise host to debug all kind of sip related issues.

Also would suggest looking proxy and lb logs.

Just ssh to the sipwise host and use sngrep and tail or less...

Let us know if you would need more help though...

Tomi

On 21 Sep 2019, at 10.35, Kamran Asghar <kamran.asghar at live.com <mailto:kamran.asghar at live.com> > wrote:

Hi guys,

I am new to sipwise and recently installed sipwise latest version. I am configured the subscribers and able to dial within the extensions. Now stepped further and created peering with one of the sip trunk. Configured it like what is mentioned in the document but i have tow challenges;

*	Unable to check sip of trunks
*	Call rejected and i have tested it through call verification and it is giving following;

<image.png>

Can someone advise me how to check the status of sip trunks? and How we call troubleshoot or trace the call records to find the issue?

Your help is really appreciated.

Regards

Kamran

Sent from Outlook <http://aka.ms/weboutlook> 

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