[Freeswitch-users] sip message logging and analysis

Kristian Kielhofner kristian.kielhofner at gmail.com
Thu Dec 17 16:27:16 PST 2009


  Probably the cleanest (albeit non-FreeSWITCH) way to implement this
would be to use OpenSIPS/SER/etc between you and the carrier with the
siptrace module.

  But that's probably more work than you want.  There's always tcpdump
with a decent filter (udp port 5060 and host x.x.x.x) and then
something like http://www.badpenguin.co.uk/files/pcap-util2

  Both will allow you to search for BYEs and who is sending them.

  Also keep in mind that they (or you) may just be dropping the RTP
without ever sending a BYE.  Setting the various RTP timeouts in
FreeSWITCH can help with that.  You can then look for logs/events (are
there any for RTP timeout?) to see who's dropping RTP.

On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact <frank at impactfax.com> wrote:
> I bit off topic but…
> Using FS to send calls sip to the LD carrier.
> Some calls have problems where they drop the call or audio drops or
> whatever.
> The carrier’s first response is that we dropped the call.  But this is  a
> day later after the trouble has been reported.
> I am looking for guidance on how to log all sip message traffic and then be
> able to easily retrieve to find a call and look at what sip messages really
> were being based and by whom.  Maybe store them in a database or some other
> file that might be opened by an analysis tool.
> Any suggestions on how to log this information and then what tool to use for
> later analysis?
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Kristian Kielhofner

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