I would like to get the cdr accounting messages instantly and syslog works nice for this. However, I need to know what phone that answered shared dn's. The syslog message for gw-accounting does not include this. Is there some way to fix that? If not - what would be the best way to get such information instantly?
This is gw-accounting to file example (with phone id/tag = 5):
1347975330,41182,0,1,"2E4995E1 CC11E2 B38B0013 803E911E","","",".15:32:52.544 gmt Tue Sep 18 2012",".15:32:52.554 gmt Tue Sep 18 2012",".15:32:59.724 gmt Tue Sep 18 2012",".15:35:30.394 gmt Tue Sep 18 2012","","","originate",0,"",0,0,7527,1204320,"94822580","094822580","73197457","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","","",151,"Tariff:Unknown","","0","","","","","","","","","","","","","","Ventelo-Cust","","","","","","","","","ton:0,npi:0,#:73197457","ton:0,npi:0,pi:0,si:0,#:94822580","","","","","ton:0,npi:0,pi:0,si:0,#:094822580","","","","","","","Unknown","","","","","","TWC","09/18/2012 15:32:52.528","094822580","73197457",0,62507,2E4995E1 CC11E2 B38B0013 803E911E,A0DE,"","","","","dn:shared,usr:57,tag:5","cme","","","",""
This is the same gw-accounting syslog message (no phone information):
<189>2246405: 2244998: .Sep 18 15:35:30.414 gmt: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 2E4995E1CC11E2B38B0013803E911E, SetupTime .15:32:52.544 gmt Tue Sep 18 2012, PeerAddress 73197457, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime .15:32:59.724 gmt Tue Sep 18 2012, DisconnectTime .15:35:30.394 gmt Tue Sep 18 2012, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 7527, ReceiveBytes 1204320
Please take a llok on this:
You'll find all the available values that CDR File Accounting can generate:
System time stamp when CDR is captured.
Value of the Call-ID header.
Call leg type:
Unique call identifier generated by the gateway. Used to identify the separate billable events (calls) within a single calling session.
Number that this call was connected to in E.164 format.
Subaddress configured under a dial peer.
Setup time in Network Time Protocol (NTP) format: hour, minutes, seconds, microseconds, time_zone, day, month, day_of_month, year.
Time at which call is alerting.
Connect time in NTP format: hour, minutes, seconds, microseconds, time_zone, day, month, day_of_month, year.
Disconnect time in NTP format: hour, minutes, seconds, microseconds, time_zone, day, month, day_of_month, year.
Q.931 disconnect cause code retrieved from Cisco IOS call-control application programming interface (Cisco IOS CCAPI).
ASCII text describing the reason for call termination.
Gateway's behavior in relation to the connection that is active for this leg.
Number of charged units for this connection. For incoming calls or if charging information is not supplied by the switch, the value is zero.
Type of information carried by media.
1=Other 9 not described
Total number of transmitted packets.
Total number of transmitted bytes.
Total number of packets received.
Total number of bytes received.
Username for authentication. Usually this is the same as the calling number.
Originating carrier identification code, used in routing to identify the network.
Terminating carrier identification code.
Duration, in ms, of transmit path open from this peer to the voice gateway for the call.
ID value of the peer table entry to which this call was made. If a peer table entry for this call does not exist, the value of this object is zero.
ifIndex value of the peer table entry to which this call was made. If a peer table entry for this call does not exist, the value of this object is zero.
ifIndex value of the logical interface through which this call was made. For ISDN media, this is the ifIndex of the B channel that was used for this call.
Average ACOM level, in dB, for the call (ACOM is the combined loss achieved by the echo canceler). 1 indicates that the level cannot be determined or level detection is disabled.
Average noise level for the call, in dBm.
Duration, in ms, for this call.
Account code entered using the Acct soft key during call setup or when connected to an active call.
Payload size of the voice packet.
Negotiated coder rate. Transmit rate of voice/fax compression to its associated call leg for the call.
Duration, in ms, of voice playout from data received on time for this call.
Remote system UDP listener port to which voice packets are transmitted.
Remote-media gateway UDP port.
Whether or not voice-activity detection (VAD) is enabled for the voice call.
Average playout FIFO delay plus the decoder delay during the voice call.
Voice-packet round-trip delay, in ms, between local and remote devices on the IP backbone during a call.
High-water mark voice playout FIFO delay during the voice call.
Low-water mark voice playout FIFO delay during the voice call.
Duration, in ms, of the voice signal played out with the signal synthesized from parameters or samples of data preceding and following in time because of voice data not received on time (or lost) from the voice gateway for this call.
Duration, in ms, of the voice signal played out with signal synthesized from redundancy parameters available because of voice data not received on time (or lost) from the voice gateway for this call.
Duration, in ms, of the voice signal replaced with the signal played out during silence because of voice data not received on time (or lost) from the voice gateway for this call
Duration, in ms, of voice signal played out with signal synthesized from parameters or samples of data preceding in time because of voice data not received on time (or lost) from voice gateway for this call.
Number of received voice packets that arrived too early to store in the jitter buffer during the call.
Number of received voice packets that arrived too late to play out with the codec during the call.
Number of lost voice packets during the call.
Maximum bandwidth used by the video call.
Fax start time in a call. Multiple fax start/stop time stamps can exist in one call. Recorded for both VoIP and telephony call legs.
Fax stop time in a call. Multiple fax start/stop time stamps can exist in one call. Recorded for both VoIP and telephony call legs.
Depth of the jitter buffer, in ms.
Number of jitter buffer overflow events during the call.
Initial high-speed modulation and baud rate negotiated at the time the call is connected.
Most recent high-speed modulation and baud rate.
Total number of transmitted and received fax pages.
Number of packets transmitted.
Number of packets received.
Whether a fax was originated (transmitted) or terminated (received) by this gateway.
Packet loss concealment; number of white scan lines inserted (nonzero for outbound gateway only).
Whether error correction mode is enabled.
Encapsulation protocol used for fax transfer.
NSF country code of the local fax device; country name per T.35, Annex A.
NSF manufacturer code of the local fax device.
Whether fax transfer was successful, the transfer failed, or indeterminate.
Override session time.
AV pairs sent from the voice gateway to the RADIUS server that you can define. You can set (write) the value with a customized Tcl IVR script.
Cause of failed calls. For more information, see the "Internal Error Codes" section on page 91.
Value representing impairment/calculated planning impairment factor (ICPIF) of the voice quality on the connection provided by lower-layer drivers (such as the digital-signal-processor). Low numbers represent better quality.
Remote-media gateway IP address.
Remote-media gateway DNS name.
ISUP carrier ID.
Best-fit calling party category value extracted from the Generic Transparency Descriptor (GTD). Sent in start and stop accounting messages for call legs 1 and 4. Optionally, this field also contains:
•3-character country code representing the country variant extracted from the GTD Protocol Name (PRN) country field.
•National value extracted from the GTD Field Compatibility Information (FDC) data field.
Sent in start and stop accounting messages for call legs 1 and 4.
Charge number used for call.
Sent in start and stop accounting records for call legs 1 and 4.
Gatekeeper identifier, or the destination zone or area, of the outgoing VoIP call.
Gatekeeper identifier, or the source zone or area, of the incoming VoIP call.
Trunk-group label associated with the group of voice ports from which the outgoing TDM call leaves the gateway.
Carrier ID of the trunk group through which the call leaves the gateway or the partnering voice services provider identifier of the outgoing VoIP call.
DSP ID used for the current call.
Trunk group label associated with the group of voice ports from which the incoming TDM call arrived on the gateway.
Carrier ID of the trunk group through which the call arrived or the partnering voice service provider identifier of the incoming VoIP call.
SIP business group ID.
Transferor information in the REFER/BYE/ALSO of SIP call. Used only in SIP call transfer.
Type of feature:
BXFER = Blind transfer
Success (0) or failure (1).
Feature operation time. Time stamp of the operation start and stop time, if applicable for a given feature.
Feature ID of the invocation. Identifies a unique instance of a feature attribute within a gateway. This number is incremented for each added feature attribute.
Called number received in the incoming signaling message before any translation rules are applied.
Calling number received in the incoming signaling message before any translation rules are applied.
Original calling number received by the gateway.
GTD connected number.
Redirection number received by the gateway.
Called number to be sent out of the gateway.
Calling number to be sent out of the gateway.
Final translated received number.
Called number presented by the gatekeeper in the ACF RAS message. GK/GKTMP could modify the called number by appending a prefix or leave it unchanged.
Calling number presented by the gatekeeper in the ACF RAS message. The GK/GKTMP could modify the calling number which is carried in the ACF nonstandard parameter.
Destination number collected by the gateway (application) that is used to route the call. Only applicable for 2-stage calls.
Maximum number of hops in the SIP invite message.
Redirecting number extracted from the redirect number parameter. Sent in start accounting messages for all call legs.
noa=Nature of address
T1/channel associated signaling (CAS) or E1/R2 signal information about a subscriber.
Description assigned to the voice port of the incoming call.
Description assigned to the voice port of the outgoing call.
Session protocol used for calls between the local and remote router through the IP backbone. Always equal to "sip" for SIP or "Cisco" for H.323.
Local hostname that would be accessed or used by the SNMP MIBs.
Sent in stop accounting messages for call legs 1 and 4. Also included in interim-update packets.
Feature name. Two-Way Call (TWC), Call Forward All (CFA), Call Forward Busy (CFBY), Call Forward No Answer (CFNA), Blind Transfer (BXFER), Consultive Transfer (CXFER), Hold (HOLD), Resume (RESUME).
Feature invocation time.
CFA, CFNA, CFBY
feature status (frs)
feature ID (fid)