Some providers do retain call data for diagnostic purposes and to to aid in troubleshooting. Why not politely ask them if they could provide you with a sip trace themselves or forward along the evidence that supported their conclusion. They should be willing to help you solve a problem that may potentially be of benefit to their other customers that report similar issues.
Otherwise, as others suggest, you could simply capture the signaling and media traffic from the FS box itself using "tcpdump" (e.g. tcpdump -i eth0 -s 0 -w debug.pcap host 127.0.0.1 ) or ngrep (-d eth0 -W byline -O /tmp/debug.pcap host 127.0.0.1) and analyze the resulting file in Wirehark (Statistics->Voip Calls or Telephony->Voip Calls in the current version). If your provider is using a session border controller or does not have a distributed architecture, then you can replace 127.0.0.1 with the appropriate address. If not, then simply don't use the host filter at all (it will result in a larger capture file). I would just keep in mind that if an upstream device (NAT router, firewall, etc.) is wreaking havoc with session refreshes by dropping re-INVITEs or UPDATEs (associated with session refreshing), you may not see them because of your vantage point. The reason I typically recommend using the "-i" (tcpdump) and "-d" (ngrep) switch is to avoid linux 'cooked' captures (more of a personal preference since I occasionally do have to convert or merge captures). If you only have SSH access to your FS box, you may want to use tcpdump or ngrep along with "screen". "tshark" (tty/cli vesion of Wireshark) and "sipgrep" are also extremely useful. The later requires ngrep and a couple perl modules but I believe it is included with FS in the contrib or scripts directory--I forget which). -metik Frank @ Impact 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? > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
