Admin Guide - VoIP software provider

4.1.13. CDR RecordsAfter every call (and SMS, etc), a new CDR is stored in the database tb_cdrs table (and in tb_cdrresellers when the reseller option is used).CDR records can be filtered, analyzed, exported and lots of vital statistics are based on this records.CDR records will contain the following fields:Id: database identifier. Auto incrementDatum: the date-time when the CDR were inserted into the database (call end time)Callstartdate: call start time (first INVITE sent or received)Callenddate: first disconnect code or CANCEL/BYE received or sentConnectdate: first 200 OK received or ACK for 200 OK sentConnecttime: time elapsed until call fail or call pickup (routing+ringing time)Workenddate: used for callcenters and represents the time when the operator have finished to work with the current client (CRM updates, etc)Realduration: speech lengthDiscparty: disconnect origination. 1=called or gsm, 2=caller or h323, 3=router (server)Discreason: disconnect reason code. Explanations in tb_reasoncodesCallerid: caller database id from tb_usersCallerip: the origination ipCallernumber: caller phone number (or sip username)Calledid: called database id from tb_usersSimid: called simid (if any)Calledline: Engine (phone line) or the called proxy authorization id (from tb_proxyauth)Calledip: the ip address of the called partyOrigCalledNumber: received called party number (not modified)Callednumber: techprefix and the normalized called number. If the server will block the call too early, than you may have the “origcallednumber” here (no techprefix and normalization)DialedNumber (calleddialed): the forwarded called number (sometimes only the “callednumber” will be insterted here)Rtpsent: rtp packets from caller to called. 0 if no rtp routing. At least 1 if routed. If remains 1, then routing has failedIn case of sip this means rtp packets received from the called and sent to caller successfullyRtprec: rtp packets from called to caller. 0 if no rtp routing. At least 1 if routed. If remains 1, then routing has failedIn case of sip this means rtp packets received from the caller and sent to called successfullyRtplost: lost rtp packetsRtpcodec: voice codec nameRtpname: used for gatewaysRtpframes: rtp payload framed in one udp packetSignalin: audio signal strength into the playback device