Text Call Detail Records
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Text call detail records (CDR) can be used on the TMG800, TMG3200 or TMG7800. The Call Detail Records are saved in a log file on a local disk. There is an entry done for each call leg, at the start of a call and at the end of the call. In addition, the system can be configured to update the CDRs periodically.
The format of the CDR traces is defined by configuration, using variables that are replaced by the Gateway application when writing to the log.
For example, the following variable will be replaced by the called number: @{CalledNumber}
In order to save disk space and simplify the archiving and backup of CDR log files, this log file is automatically archived (gzipped) and rotated every N seconds, as specified by the system configuration.
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Enabling text CDR
The enabling of Text CDR is described in Enabling CDR.
Available variables
The following variables exist and can be used to define the CDR log format.
Variables used for identifying call (or call leg) in general
*@{StatusType}: Indicates the type of record ("Start", "Update" or "End").
*@{LegId}: Unique Id for this leg (32 bits value, but may eventually be upgraded to 64 bits value).
*@{OtherLegId}: Id of the other call leg joined with current call leg.
*@{SessionId}: Unique call identifier for two joined and answered call legs, including failed outgoing call attempts (route retry)
Formatted as 4 values of 32 bits, printed as 4 blocks of 8 hexadecimal characters separated by a space
Ex: a939d169 299ffcd0 00000000 00000000
Note: Call Transfer Target legs have separate SessionId. If you need an id to correlate a transferred call
to the original call, use @{OriginalSessionId}.
*@{OriginalSessionId}: Refers to @{SessionId} of the original legs for this call, in case of call transfer.
In fact, Transfer Target leg has a different value for @{SessionId}, but can be linked with the original call
legs through @{OriginalSessionId}.
*@{LinkId}: Same meaning as @{OriginalSessionId}, but presented as a 32 bits integer value.
For all Ids above, please see Known Limitations below for notes about uniqueness of these values.
Timestamps
*@{AlertTime}: Time where the call has started ringing.
Note: Supported on release 2.7 and up only.
*@{ConnectedTime}: Time where the call was answered (and connected with another leg) (in number of seconds since epoch)
*@{ConnectedTime:format}: Same as @{ConnectedTime} but with custom print format (local time zone), using 'strftime' style, with added
support for @m replaced by milliseconds.
Example format: %Y-%m-%d %H:%M:%S.@m -> 2009-09-02 12:16:24.333
*@{ConnectedTimeUtc:format}:Same as @{ConnectedTime:format} but printed in UTC time, rather than local time zone.
*@{CallDuration}: Call duration in seconds (not available in the "start" CDR log)
Note: Supported on release 2.7 and up only.
*@{CallDurationMs}: Call duration in milliseconds (end - connected time) (not available in the "start" CDR log)
Note: Supported on release 2.7 and up only.
*@{EndTime}: Time where the call has started terminating (in number of seconds since epoch).
Note that slightly differs from the @{Timestamp}, since the 'End' CDR trace is printed once the call has
finished terminating,
while @{EndTime} reports the time where the call has started terminating (upon hangup for example).
*@{EndTime:format}: Same as @{EndTime} but with custom print format (local time zone), using 'strftime' style, with added
support for @m replaced by milliseconds.
Example format: %Y-%m-%d %H:%M:%S.@m -> 2009-09-02 12:16:24.333
*@{EndTimeUtc:format}: Same as @{EndTime:format} but printed in UTC time, rather than local time zone.
*@{RingingDuration}: Ringing duration in seconds (connected - start time)
Note: Supported on release 2.7 and up only.
*@{RingingDurationMs}: Ringing duration in milliseconds (connected - start time)
Note: Supported on release 2.7 and up only.
*@{StartTime}: Time where the call was created (in number of seconds since epoch)
*@{StartTime:format}: Same as @{StartTime} but with custom print format (local time zone), using 'strftime' style, with added
support for @m replaced by milliseconds.
Example format: %Y-%m-%d %H:%M:%S.@m -> 2009-09-02 12:16:24.333
*@{StartTimeUtc:format}: Same as @{StartTime:format} but printed in UTC time, rather than local time zone.
*@{Timestamp}: Time where this CDR log entry was written.
Should not be used for billing purposes. Use @{EndTime} for billing @{EndTime} reports the time where
the call has started terminating (hangup), while *@{Timestamp} the time where signaling confirmed the termination.
*@{Timestamp:format}: Same as @{Timestamp} but with custom print format (local time zone), using 'strftime' style, with added
support for @m replaced by milliseconds.
Example format: %Y-%m-%d %H:%M:%S.@m -> 2009-09-02 12:16:24.333
*@{TimestampUtc:format}: Same as @{Timestamp:format} but printed in UTC time, rather than local time zone.
*@{CalledNumber}: Called number
*@{CallingNumber}: Calling number
*@{CallingPresentation}: Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
Note: Supported on release 2.7 and up only.
*@{CallingSubscriberNumber}: Second calling number (ISDN) or Generic number of type "additional calling party number" (SS7)
Note: Supported on release 2.7.43 and up only.
*@{CallType}: Call type ("Telephony" or "VOIP")
*@{IncomingNAP}: Name of the NAP that originated this call (incoming call leg's NAP name).
Note: Supported on release 2.7 and up only.
*@{NAP}: Name of the NAP this call leg is from
*@{OriginatorName}: Direction of the call:
- "answer" (incoming call leg) - "originate" (outgoing call leg)
Note: In release 2.6 and earlier, the direction of the call is as follows:
- "originate" (incoming call leg) - "answer" (outgoing call leg)
In release 2.7, the CDR option "Reverse CDR call origin" in Gateway configuration provide the same values as in release 2.6.
*@{OriginalCalledNumber}: Original called number
Note: Supported on release 2.7 and up only.
*@{Protocol}: Type of protocol used ("SS7", "ISDN", or "SIP")
*@{RedirectingNumber}: Redirecting number
Note: Supported on release 2.7 and up only.
*@{TerminationCause}: Cause of the call termination, printed as an integer value (refering enum TBCMC_CALL_REASON_CODE)
*@{TerminationCauseString}: Cause of the call termination, printed as a string value
*@{OriginalCause}: Original cause of the call termination, printed as a string value (refering enum TBCMC_CALL_REASON_CODE)
*@{TerminationSource}: Identifies the cause of the leg termination:
- "TermInd": Terminating indication has been received on this leg
- "JoinedTermInd": Terminating indication has been received on the leg joined to current leg,
and has been forwarded to current leg
- "App": Call control application has asked to drop the call
- "Engine": Toolpack engine has decided to terminate this call
(generally due to local errors like disconnected TMedia)
*@{UserName}: Static value: 100.
*@{SipCallId}: Content of the "call-id" SIP header
*@{ChargeIndicator}: For TDM (SS7 and CAS R2); received charge indicator in Alert message.
*@{Codec}: Codec used for this call ("G711" for example)
*@{LocalMediaInfo}: Protocol type dependent information on the call leg (local information for SIP calls).
For TDM (Telephony) calls (SS7 or ISDN): "trunk_name:timeslot_nb".
For VOIP calls (SIP): "codec@ip:port" (IP and Port locally used for receiving RTP)
*@{LocalMediaIP}: For VOIP calls, RTP IP address locally used for this call leg.
*@{LocalMediaPort}: For VOIP calls, RTP UDP port locally used for this call leg.
*@{RemoteMediaInfo}: Protocol type dependent information on the call leg (remote information SIP calls).
For TDM (Telephony) calls (SS7 or ISDN): "trunk_name:timeslot_nb".
For VOIP calls (SIP): "codec@ip:port" (IP and Port TMedia is sending RTP to)
*@{RemoteMediaIP}: For VOIP calls, RTP IP address of the remote peer for this call leg.
*@{RemoteMediaPort}: For VOIP calls, RTP UDP port of the remote peer for this call leg.
*@{TimeslotNumber}: For TDM (Telephony) calls, timeslot number that this call was using for audio.
*@{TrunkName}: For TDM (Telephony) calls, name of the trunk that this call was using for audio.
Statistics
Note: Statistics are available only in the "end" CDR entry, not available during the call.
*@{Stat:stat_name}: Get the specified statistic's value. Example: @{Stat:Rtp:Rx:Packets}.
Available statistics names:
RTP Rx (received IP packets):
- Rtp:Rx:Packets: Number of packets received from the IP network (voice + comfort noise + signaling)
- Rtp:Rx:VoicePackets: Number of voice packets received from the IP network
- Rtp:Rx:VoiceBytes: Number of payload bytes from voice packets received from the IP network
- Rtp:Rx:CNPackets: Number of comfort noise packets received from the IP network
- Rtp:Rx:SignalingPackets: Number of signaling packets (like "Telephony events") received from the IP network
- Rtp:Rx:RFC2833Tones: Number of RFC2833 tones received from the IP network, and played back to TDM
- Rtp:Rx:RedundantPackets: Number of redundant packets received from the IP network (same sequence number received more than once)
- Rtp:Rx:MeanPower: Mean audio power (from IP to TDM direction), in dBov (0 the loudest, negative is softer level)
- Rtp:Rx:VoiceDuration: Duration (in seconds) of voice (from IP to TDM direction)
- Rtp:Rx:PlayoutDelay: Current IP to TDM audio delay in millisecond (adaptive jitter buffer delay)
- Rtp:Rx:MaxPlayoutDelay: Maximum (for the whole call) IP to TDM audio delay in millisecond (adaptive jitter buffer delay)
- Rtp:Rx:SsrcChanges: Number of changes of "SSRC" RTP header value (counting initial SSRC received as one)
- Rtp:Rx:Concealments: Number of times decoder made audio concealment to preserve audio continuity
(frequent upon first packet of a call, or after RFC2833 tones)
- Rtp:Rx:Echo:ERLLevel: Attenuation of a signal from the received-out port to the send-in port of an echo canceller
- Rtp:Rx:Error:BufOverflows: Number of packets dropped due to jitter buffer adaptation (to keep play-out delay low)
- Rtp:Rx:Error:SeqErrors: Number of sequence errors detected among packets received from the IP network
- Rtp:Rx:Error:BadPackets: Number of packets received with invalid header, or other error preventing proper audio decoding
- Rtp:Rx:Error:RFC2833BadPackets: Number of RFC2833 signaling packets that could not be properly decoded
- Rtp:Rx:Error:LatePackets: Number of packets that arrived after the moment they should have started to be played-out to TDM
- Rtp:Rx:Error:LostPackets: Number of packets considered "lost" (never received for play-out to TDM)
- Rtp:Rx:Errors: Total of all "Rtp:Rx:Error:" statistics above
RTP Tx (sent IP packets):
- Rtp:Tx:Packets: Number of packets sent to the IP network (voice + comfort noise + signaling)
- Rtp:Tx:VoicePackets: Number of voice packets sent to the IP network
- Rtp:Tx:CNPackets: Number of comfort noise packets sent to the IP network
- Rtp:Tx:SignalingPackets: Number of signaling packets (like "Telephony events") sent to the IP network
- Rtp:Tx:RFC2833Tones: Number of RFC2833 tones sent to the IP network (relaying corresponding detected TDM tones)
- Rtp:Tx:MeanPower: Mean audio power (from TDM to IP direction), in dBov (0 the loudest, negative is softer level)
- Rtp:Tx:VoiceDuration: Duration (in seconds) of voice (from TDM to IP direction)
- Rtp:Tx:Echo:ACOMLevel: Attenuation of a signal from the receive-out port to the send-out port of an echo canceller
- Rtp:Tx:Error:ArpFailure: Number of failures to ARP remote IP address to send RTP packets to (preventing sending any packets)
- Rtp:Tx:Errors: Total of all "Rtp:Tx:Error:" statistics above
RTCP (only if RTCP is enabled on the profile):
- Rtcp:Packets: Number of packets reported as received by remote equipment
- Rtcp:LostPackets: Number of packets reported as lost by remote equipment
- Rtcp:Jitter: Inter-packet interval jitter, in millisecond.
T.38 Rx (from IP to TDM direction):
- T38:StateChanges: Number of T.38 state changes, through T.30 negotiation between FAX machines
- T38:Rx:Pages: Number of FAX pages received from T.38 packets from the IP network
- T38:Rx:Packets: Number of T.38 packets received from the IP network
- T38:Rx:Bytes: Number of T.38 payload bytes from T.38 packets received from the IP network
- T38:Rx:Duration: Duration of the T.38 session from IP to TDM
- T38:Rx:PlayoutDelay: Current IP to TDM audio delay in millisecond (re-modulation and jitter buffer delay)
- T38:Rx:MaxPlayoutDelay: Maximum IP to TDM audio delay in millisecond (re-modulation and jitter buffer delay)
- T38:Rx:Error:BufOverflows: Number of packets dropped because "from IP" jitter buffer is full
- T38:Rx:Error:BufUnderflows: Number of times T.38 re-modulation to TDM had no data (received from IP) to re-modulate
- T38:Rx:Error:SeqErrors: Number of sequence errors detected among packets received from the IP network
- T38:Rx:Error:BadPackets: Number of packets received with invalid header, or other error preventing proper audio decoding
- T38:Rx:Errors: Total of all "T38:Rx:Error:" statistics above
T.38 Tx (to IP, from TDM direction):
- T38:Tx:Pages: Number of FAX pages received from TDM, sent to T.38 packets to the IP network
- T38:Tx:Packets: Number of T.38 packets sent to the IP network
- T38:Tx:Bytes: Number of T.38 payload bytes from T.38 packets sent to the IP network
- T38:Tx:Duration: Duration of the T.38 session from TDM to IP
- T38:FromTdm:MeanPower: Mean audio power received from TDM side, in dBov (0 the loudest, negative is softer level)
*@{ApplicationName}: Name of the application that has written this log ("Gateway")
*@{RouteAttribute:attr}: Replaced by the value of a custom route attribute, for the selected
route. This will apply only to outgoing call legs that were made from routing.
Eligible route attributes are:
Route name: Use attribute "route_name". Ex: @{RouteAttribute:route_name}
Route set name: Use attribute "routeset_name". Ex: @{RouteAttribute:routeset_name}
Custom route attribute column: Use the custom attribute name. Ex: @{RouteAttribute:priority}
Routing script parameters: Use the route attribute name provided by routing script.
Ex: If script provides: route[:my_param]="myval"
Then it's included in CDR with @{RouteAttribute:my_param}]
***** Important note: This parameter cannot be retrieved after a switchover of the active to
the standby Toolpack host.
It's thus recommended to insert this information in the "Start" CDR entry,
rather than the "End" CDR entry.
*@{ScriptAttribute:attr}: Replaced by the value of a routing script attribute stored in params[:bridge].
This will apply for incoming and outgoing call legs.
Eligible script attributes are:
Custom CDR value: Use attribute "CustomCdrValue". Ex: @{ScriptAttribute:CustomCdrValue}
Script parameters: Use the attribute name provided by routing script.
Ex: If script provides: params[:bridge][:my_param]="myval"
Then it's included in CDR with @{ScriptAttribute:my_param}
***** Important note: This parameter cannot be retrieved after a switchover of the active to
the standby Toolpack host.
It's thus recommended to insert this information in the "Start" CDR entry,
rather than the "End" CDR entry.
Deprecated values:
*@{MediaInfo}: Same as @{RemoteMediaInfo}
*@{RemoteIP}: Same as @{RemoteMediaIP}
*@{RemotePort}: Same as @{RemoteMediaPort}
CDR sample
Text CDR sample
2013-02-07 10:19:04.640-0500,BEG,SessionId='f2685706 00000000 00000000 00000000',LegId='0xF2685706',StartTime='1360250341',ConnectedTime='1360250344',Calling=,Called='123',NAP='NAP_SIP',Protocol='SIP',Direction='answer' 2013-02-07 10:19:04.640-0500,BEG,SessionId='f2685706 00000000 00000000 00000000',LegId='0x72685C1F',StartTime='1360250344',ConnectedTime='1360250344',Calling=,Called='123',NAP='NAP_SIP2',Protocol='SIP',Direction='originate' 2013-02-07 10:19:07.562-0500,END,SessionId='f2685706 00000000 00000000 00000000',LegId='0x72685C1F',StartTime='1360250344',ConnectedTime='1360250344',EndTime='1360250347',FreedTime='1360250347',TerminationCause='TOOLPACK_NORMAL',TerminationSource='App',Calling=,Called='123',NAP='NAP_SIP2',Direction='originate' 2013-02-07 10:19:07.656-0500,END,SessionId='f2685706 00000000 00000000 00000000',LegId='0xF2685706',StartTime='1360250341',ConnectedTime='1360250344',EndTime='1360250347',FreedTime='1360250347',TerminationCause='TOOLPACK_NORMAL',Termination Source='App',Calling=,Called='123',NAP='NAP_SIP',Direction='answer'
Radius AAA sample
Redundancy
In a TMG7800 configuration, with redundant TMG-Control servers, only the active server writes the CDR logs. Since the active server may change over time, CDR parsing must take into account that CDR logs may be found in files from the two servers.
The analysis of the logs for the purpose of extracting billing information must be done after combining the two logs, from both TMG-Control servers, sorting the entries by timestamp for example. See how to Retrieve Text CDR here.
CDR entry loss due to switchover
In some situations (during HA switchover for example), some CDR entries may be lost.
The following guide lines provide information on how to deal with these corner cases:
Retrieving Text CDRs
There are 2 ways to retrieve the text CDR manually or automatically. The procedures are described in Retrieve Text CDR.
Notes
- This mode of operation is not recommended for TMG800 or TMG3200 with a flash disk (last shipment January 2011). However, the TMG800 or TMG3200 with SATA or SSD disk option is OK (use command tbproduct: if the command can be executed, the disk is SATA or SSD).
- TelcoBridges does not recommend storing CDR logs via network file systems (NFS or other). We highly recommend writing to a local hard drive and have developed a background script that moves the gzipped log segments for backup or analysis.
Related actions
Refer to the appropriate Toolpack release:
