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Is there a console for troubleshooting Kamailio?

I would like to see real time call routing from within Kamailio like the Asterisk Console provides.  Is there a feature for this built in?
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jer2eydevil88
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jer2eydevil88
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1 Solution
 
feptiasCommented:
I just use ngrep to capture the SIP packets. It is available as an RPM. Search for an ngrep RPM here:
http://rpm.pbone.net/

Once installed, you can run it direct from the command line:
ngrep -q <search_string> -W byline port 5060 -d any

... or you can install sipgrep which acts as a wrapper to ngrep, with slightly easier command line options and it displays the SIP packets with some elements highlighted in colour so they are easier to read.
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/utils/sipgrep/

It is also possible to capture SIP packets using ethereal/wireshark:
http://w.e-e.com/WGwRNz

The only other option I know of (and this is opensips so kamailio may differ) is the monitor utility
opensipsctl moni

..but it only shows summary statistics for the number of SIP requests handled.
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jer2eydevil88Author Commented:
In my instance I was able to see traffic passing through the system by using TCPDump as an alternative to the Asterisk console.

tcpdump -i venet0 -vv -nn host INSERT_IP_ADDRES_OF_HOST and udp port 5060
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feptiasCommented:
I'm glad that you found a solution, but not so pleased to be called "no one". Please bear in mind that the experts who contribute to EE do so voluntarily. The answers we post usually require a reasonable amount of time and effort.
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jer2eydevil88Author Commented:
I apologize... I don't know how I didn't notice the solution posted on this question but I did, also I agree that they posted an adequate solution and should be awarded their points.
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feptiasCommented:
Don't worry. If you really want to award points, then you'll have to cancel the close request as the process is automated. I've already got my required minimum quota of points for September so I'm happy.

I don't see many questions about opensips or kamailio on EE. If you have any more questions I'll try to help, but post them in the Asterisk topic area (in addition to the VoIP area) as that is where I tend to look most often.
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