To display the call history table for fax transmissions, use the
showcallhistoryfaxcommand in user EXEC or privileged EXEC mode.
showcallhistoryfax
[ brief
[ ididentifier ] | compact
[ duration
{ less | more }
time ] | ididentifier | lastnumber ]
Syntax Description
brief
(Optional) Displays a truncated version of the call history table.
ididentifier
(Optional) Displays only the call with the specified identifier. Range is a hex value from 1 to FFFF.
compact
(Optional) Displays a compact version.
durationtime
(Optional) Displays history information for calls that are longer or shorter than a specified
time
value. The arguments and keywords are as follows:
less--Displays calls shorter than the value in the
time
argument.
more--Displays calls longer than the value in the
time
argument.
time --Elapsed time, in seconds. Range is from 1 to 2147483647.
lastnumber
(Optional) Displays the last calls connected, where the number of calls that appear is defined by the
numberargument. Range is from 1 to100 .
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series.
12.0(3)XG
This command was implemented for Voice over Frame Relay (VoFR) on the Cisco 2600 series and Cisco 3600 series.
12.0(4)XJ
This command was modified for store-and-forward fax.
12.0(4)T
This command was modified. The
brief keyword was added, and the command was implemented on the Cisco 7200 series.
12.0(7)XK
This command was modified. The
brief keyword was implemented on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(5)XM
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XA
This command was modified. The output of this command was modified to indicate whether the call in question has been established using Annex E.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 was not included in this release.
12.2(11)T
This command was implemented on the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
12.3(1)
This command was modified. The following fields were added: FaxRelayMaxJitterBufDepth, FaxRelayJitterBufOverFlow, FaxRelayHSmodulation, and FaxRelayNumberOfPages.
12.3(14)T
This command was modified. T.38 fax relay call statistics were made available to Call Detail Records (CDRs) through vendor-specific attributes (VSAs) and added to the call log.
12.4(15)T
This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.
12.4(16)
This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.
12.4(22)T
This command was modified. Command output was updated to show IPv6 information.
Usage Guidelines
This command displays a call-history table that contains a list of fax calls connected through the router in descending time order. The maximum number of calls contained in the table can be set to a number from 0 to 500 using the
dial-control-mib command in global configuration mode. The default maximum number of table entries is 50. Each call record is aged out of the table after a configurable number of minutes has elapsed, also specified by the
dial-control-mib command. The default timer value is 15 minutes.
You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword
last, and define the number of calls to be displayed with the number argument.
To display a truncated version of the call history table, use the
brief keyword.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following is sample output from theshowcallhistoryfaxcommand:
The table below provides an alphabetical listing of the fields displayed in the output of the
showcallhistoryfaxcommand and a description of each field.
Table 1 show call history fax Field Descriptions
Field
Description
ACOM Level
Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceler, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.
BearerChannel
Identification of the bearer channel carrying the call.
Buffer Drain Events
Total number of jitter buffer drain events.
Buffer Fill Events
Total number of jitter buffer fill events.
CallDuration
Length of the call, in hours, minutes, and seconds, hh:mm:ss.
CallerName
Voice port station name string.
CallOrigin
Call origin: answer or originate.
CallState
Current state of the call.
ChargedUnits
Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.
CodecBytes
Payload size, in bytes, for the codec used.
CoderTypeRate
Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.
ConnectionId
Global call identifier for this gateway call.
ConnectTime
Time, in milliseconds (ms), at which the call was connected.
Consecutive-packets-lost Events
Total number of consecutive (two or more) packet-loss events.
Corrected packet-loss Events
Total number of packet-loss events that were corrected using the RFC 2198 method.
Dial-Peer
Tag of the dial peer sending this call.
DisconnectCause
Cause code for the reason this call was disconnected.
DisconnectText
Descriptive text explaining the reason for the disconnect.
DisconnectTime
Time, in ms, when this call was disconnected.
EchoCancellerMaxReflector=64
The location of the largest reflector, in ms. The reflector size does not exceed the configured echo path capacity. For example, if 32 ms is configured, the reflector does not report beyond 32 ms.
ERLLevel
Current Echo Return Loss (ERL) level for this call.
FaxTxDuration
Duration of fax transmission from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.
FaxRelayJitterBufOverFlow
Count of number of network jitter buffer overflows (number of packets). These packets are equivalent to lost packets.
FaxRelayMaxJitterBufDepth
Maximum depth of jitter buffer (in ms).
FaxRelayHSmodulation
Most recent high-speed modulation used.
FaxRelayNumberOfPages
Number of pages transmitted.
GapFillWithInterpolation
Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithRedundancy
Duration of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithPrediction
Duration of the voice signal played out with signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1 compression algorithms.
GapFillWithSilence
Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.
GENERIC
Generic or common parameters, that is, parameters that are common for VoIP and telephony call legs.
Total H.323 call legs for which call records are available.
HiWaterPlayoutDelay
High-water-mark Voice Playout FIFO Delay during this call.
ImgPages
The fax pages that have been processed.
Incoming ConnectionId
The incoming_GUID. It can be different with ConnectionId (GUID) when there is a long_pound or blast_call feature involved. In those cases, incoming_GUID is unique for all the subcalls that have been generated, and GUID is different for each subcall.
Index
Dial peer identification number.
InfoActivity
Active information transfer activity state for this call.
InfoType
Information type for this call; for example, voice or fax.
InSignalLevel
Active input signal level from the telephony interface used by this call.
Last Buffer Drain/Fill Event
Elapsed time since the last jitter buffer drain or fill event, in seconds.
LogicalIfIndex
Index number of the logical interface for this call.
LoWaterPlayoutDelay
Low-water-mark Voice Playout FIFO Delay during this call.
LowerIFName
Physical lower interface information. Appears only if the medium is ATM, Frame Relay (FR), or High-Level Data Link Control (HDLC).
Media
Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry is TELE. If the call is carried over the voice network side, the entry is either ATM, FR, or HDLC.
Modem passthrough signaling method in use
Indicates that this is a modem pass-through call and that named signaling events (NSEs)--a Cisco-proprietary version of named telephone events in RFC 2833--are used for signaling codec upspeed. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions. This means that you might move to a faster codec when you have both voice and data calls and then slow down when there is only voice traffic.
NoiseLevel
Active noise level for this call.
OnTimeRvPlayout
Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.
Original call information regarding calling, called, and redirect numbers, as well as octet-3s. Octet-3s are information elements (IEs) of Q.931 that include type of number, numbering plan indicator, presentation indicator, and redirect reason information.
OutSignalLevel
Active output signal level to the telephony interface used by this call.
PeerAddress
Destination pattern or number associated with this peer.
PeerId
ID value of the peer table entry to which this call was made.
PeerIfIndex
Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.
PeerSubAddress
Subaddress when this call is connected.
Percent Packet Loss
Total percent packet loss.
Port
Identification of the voice port carrying the call.
ReceiveBytes
Number of bytes received by the peer during this call.
ReceiveDelay
Average Playout FIFO Delay plus the Decoder Delay during this voice call.
ReceivePackets
Number of packets received by this peer during this call.
ReleaseSource
Number value of the release source.
RemoteIPAddress
Remote system IP address for the VoIP call.
RemoteUDPPort
Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.
RoundTripDelay
Voice packet round-trip delay between the local and remote systems on the IP backbone for this call.
SelectedQoS
Selected Resource Reservation Protocol (RSVP) quality of service (QoS) for this call.
SessionProtocol
Session protocol used for an Internet call between the local and remote routers through the IP backbone.
SessionTarget
Session target of the peer used for this call.
SetupTime
Value of the system UpTime, in ms, when the call associated with this entry was started.
SignalingType
Signaling type for this call; for example, channel-associated signaling (CAS) or common-channel signaling (CCS).
SIP call-legs
Total SIP call legs for which call records are available.
Telephony call-legs
Total telephony call legs for which call records are available.
Time between Buffer Drain/Fills
Minimum and maximum durations between jitter buffer drain or fill events, in seconds.
Number of bytes sent by this peer during this call.
TransmitPackets
Number of packets sent by this peer during this call.
TxDuration
The length of the call. Appears only if the medium is TELE.
VAD
Whether voice activation detection (VAD) was enabled for this call.
VoiceTxDuration
Duration of voice transmission from this peer to the voice gateway for this call. Derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.
The following is sample output from the
showcallhistoryfaxbriefcommand:
The table below describes the fields not shown in the table above.
Table 2 show call history fax Field Descriptions
Field
Description
FaxRelayDirection
Direction of fax relay.
FaxRelayEcmStatus
Fax relay error correction mode status.
FaxRelayEncapProtocol
Fax relay encapsulation protocol.
FaxRelayFaxSuccess
Fax relay success.
FaxRelayInitHSmodulation
Fax relay initial high speed modulation.
FaxRelayMostRecentHSmodulation
Fax relay most recent high speed modulation.
FaxRelayNsfCountryCode
Fax relay Nonstandard Facilities (NSF) country code.
FaxRelayNsfManufCode
Fax relay NSF manufacturers code.
FaxRelayPktLossConceal
Fax relay packet loss conceal.
Related Commands
Command
Description
dial-control-mib
Specifies attributes for the call history table.
showcallactivefax
Displays call information for fax transmissions that are in progress.
showcallactivevoice
Displays call information for voice calls that are in progress.
showcallhistoryvoice
Displays the call history table for voice calls.
showdial-peervoice
Displays configuration information for dial peers.
shownum-exp
Displays how the number expansions are configured in VoIP.
showvoiceport
Displays configuration information about a specific voice port.
show call history media
To display the call history table for media calls, use the
showcallhistorymediacommand in user EXEC or privileged EXEC mode.
show call history media
[ [ brief ]
[ ididentifier ] | compact
[ duration
{ less | more }
seconds ] | lastnumber ]
Syntax Description
brief
(Optional) Displays a truncated version of the call history table.
ididentifier
(Optional) Displays only the call with the specified
identifier. The range is from 1 to FFFF.
compact
(Optional) Displays a compact version of the call history table.
duration
(Optional) Displays the call history for the specified time duration.
less
Displays the call history for shorter duration calls.
more
Displays the call history for longer duration calls.
seconds
Time, in seconds. The range is from 1 to 2147483647.
lastnumber
(Optional) Displays the last calls connected, where the number of calls that appear is defined by the
numberargument. The range is from 1 to100 .
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Release
Modification
12.4(15)T
This command was introduced.
Usage Guidelines
This command displays a call-history table that contains a list of media calls connected through the router in descending time order. The maximum number of calls contained in the table can be set to a number from 0 to 500 using the
dial-control-mib command in global configuration mode. The default maximum number of table entries is 50. Each call record is aged out of the table after a configurable number of minutes has elapsed, also specified by the
dial-control-mib command. The default timer value is 15 minutes.
You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the
last keyword, and define the number of calls to be displayed with the
number argument.
To display a truncated version of the call history table, use the
brief keyword.
When a media call is active, you can display its statistics by using the
showcallactivemediacommand.
Examples
The following is sample output from theshowcallhistorymediacommand:
Router# show call history media
Telephony call-legs: 0
SIP call-legs: 0
H323 call-legs: 0
Call agent controlled call-legs: 0
Media call-legs: 4
Total call-legs: 4
GENERIC:
SetupTime=308530 ms
Index=4
PeerAddress=sip:mrcpv2ASRServer@10.5.18.224:5060
PeerSubAddress=
PeerId=2234
PeerIfIndex=184
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing (16)
ConnectTime=309440 ms
DisconnectTime=320100 ms
CallDuration=00:00:10 sec
CallOrigin=1
ReleaseSource=7
ChargedUnits=0
InfoType=speech
TransmitPackets=237
TransmitBytes=37920
ReceivePackets=0
ReceiveBytes=0
VOIP:
ConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]
IncomingConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]
CallID=14
RemoteIPAddress=10.5.18.224
RemoteUDPPort=10002
RemoteSignallingIPAddress=10.5.18.224
RemoteSignallingPort=5060
RemoteMediaIPAddress=10.5.18.224
RemoteMediaPort=10002
SRTP = off
TextRelay = off
Fallback Icpif=0
Fallback Loss=0
Fallback Delay=0
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=rtp-nte
FastConnect=FALSE
AnnexE=FALSE
Separate H245 Connection=FALSE
H245 Tunneling=FALSE
SessionProtocol=sipv2
ProtocolCallId=2FBDA670-C3511DB-8015C48C-6A894889@10.5.14.2
SessionTarget=10.5.18.224
OnTimeRvPlayout=3000
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=2740 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=100 ms
LoWaterPlayoutDelay=40 ms
Source tg label=test5
ReceiveDelay=90 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
VAD = disabled
CoderTypeRate=g711ulaw
CodecBytes=160
cvVoIPCallHistoryIcpif=16
MediaSetting=flow-around
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=
OriginalCallingOctet=0x0
OriginalCalledNumber=
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x0
TranslatedCallingNumber=555-0100
TranslatedCallingOctet=0x21
TranslatedCalledNumber=
TranslatedCalledOctet=0xC1
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwOutpulsedCallingNumber=555-0101
GwOutpulsedCallingOctet3=0x21
GwOutpulsedCallingOctet3a=0x81
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurationCallTimerStamp=
LongDurationCallDuration=
Username=
GENERIC:
SetupTime=308520 ms
Index=5
PeerAddress=sip:mrcpv2TTSServer@10.5.18.224:5060
PeerSubAddress=
PeerId=2235
PeerIfIndex=185
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing (16)
ConnectTime=309370 ms
DisconnectTime=320100 ms
CallDuration=00:00:10 sec
CallOrigin=1
ReleaseSource=7
ChargedUnits=0
InfoType=speech
TransmitPackets=0
TransmitBytes=0
ReceivePackets=551
ReceiveBytes=88160
VOIP:
ConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]
IncomingConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]
CallID=13
RemoteIPAddress=10.5.18.224
RemoteUDPPort=10000
RemoteSignallingIPAddress=10.5.18.224
RemoteSignallingPort=5060
RemoteMediaIPAddress=10.5.18.224
RemoteMediaPort=10000
SRTP = off
TextRelay = off
Fallback Icpif=0
Fallback Loss=0
Fallback Delay=0
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=rtp-nte
FastConnect=FALSE
AnnexE=FALSE
Separate H245 Connection=FALSE
H245 Tunneling=FALSE
SessionProtocol=sipv2
ProtocolCallId=2FBC6E20-C3511DB-8013C48C-6A894889@10.5.14.2
SessionTarget=10.5.18.224
OnTimeRvPlayout=7000
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=2740 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=100 ms
LoWaterPlayoutDelay=40 ms
Source tg label=test5
ReceiveDelay=95 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
VAD = disabled
CoderTypeRate=g711ulaw
CodecBytes=160
cvVoIPCallHistoryIcpif=16
MediaSetting=flow-around
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=
OriginalCallingOctet=0x0
OriginalCalledNumber=
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x0
TranslatedCallingNumber=555-0102
TranslatedCallingOctet=0x21
TranslatedCalledNumber=
TranslatedCalledOctet=0xC1
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwOutpulsedCallingNumber=555-0103
GwOutpulsedCallingOctet3=0x21
GwOutpulsedCallingOctet3a=0x81
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurationCallTimerStamp=
LongDurationCallDuration=
Username=
GENERIC:
SetupTime=408050 ms
Index=7
PeerAddress=sip:mrcpv2ASRServer@10.5.18.224:5060
PeerSubAddress=
PeerId=2234
PeerIfIndex=184
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing (16)
ConnectTime=408160 ms
DisconnectTime=426260 ms
CallDuration=00:00:18 sec
CallOrigin=1
ReleaseSource=7
ChargedUnits=0
InfoType=speech
TransmitPackets=598
TransmitBytes=95680
ReceivePackets=0
ReceiveBytes=0
VOIP:
ConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]
IncomingConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]
CallID=19
RemoteIPAddress=10.5.18.224
RemoteUDPPort=10002
RemoteSignallingIPAddress=10.5.18.224
RemoteSignallingPort=5060
RemoteMediaIPAddress=10.5.18.224
RemoteMediaPort=10002
SRTP = off
TextRelay = off
Fallback Icpif=0
Fallback Loss=0
Fallback Delay=0
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=rtp-nte
FastConnect=FALSE
AnnexE=FALSE
Separate H245 Connection=FALSE
H245 Tunneling=FALSE
SessionProtocol=sipv2
ProtocolCallId=6B0E94CD-C3511DB-801DC48C-6A894889@10.5.14.2
SessionTarget=10.5.18.224
OnTimeRvPlayout=11000
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=9560 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=100 ms
LoWaterPlayoutDelay=55 ms
Source tg label=test5
ReceiveDelay=100 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
VAD = disabled
CoderTypeRate=g711ulaw
CodecBytes=160
cvVoIPCallHistoryIcpif=16
MediaSetting=flow-around
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=
OriginalCallingOctet=0x0
OriginalCalledNumber=
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x0
TranslatedCallingNumber=555-0100
TranslatedCallingOctet=0x21
TranslatedCalledNumber=
TranslatedCalledOctet=0xC1
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwOutpulsedCallingNumber=555-0101
GwOutpulsedCallingOctet3=0x21
GwOutpulsedCallingOctet3a=0x81
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurationCallTimerStamp=
LongDurationCallDuration=
Username=
GENERIC:
SetupTime=408040 ms
Index=8
PeerAddress=sip:mrcpv2TTSServer@10.5.18.224:5060
PeerSubAddress=
PeerId=2235
PeerIfIndex=185
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing (16)
ConnectTime=408130 ms
DisconnectTime=426260 ms
CallDuration=00:00:18 sec
CallOrigin=1
ReleaseSource=7
ChargedUnits=0
InfoType=speech
TransmitPackets=0
TransmitBytes=0
ReceivePackets=911
ReceiveBytes=145760
VOIP:
ConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]
IncomingConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]
CallID=18
RemoteIPAddress=10.5.18.224
RemoteUDPPort=10000
RemoteSignallingIPAddress=10.5.18.224
RemoteSignallingPort=5060
RemoteMediaIPAddress=10.5.18.224
RemoteMediaPort=10000
SRTP = off
TextRelay = off
Fallback Icpif=0
Fallback Loss=0
Fallback Delay=0
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=rtp-nte
FastConnect=FALSE
AnnexE=FALSE
Separate H245 Connection=FALSE
H245 Tunneling=FALSE
SessionProtocol=sipv2
ProtocolCallId=6B0CC055-C3511DB-801BC48C-6A894889@10.5.14.2
SessionTarget=10.5.18.224
OnTimeRvPlayout=9000
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=9560 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=100 ms
LoWaterPlayoutDelay=55 ms
Source tg label=test5
ReceiveDelay=100 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
VAD = disabled
CoderTypeRate=g711ulaw
CodecBytes=160
cvVoIPCallHistoryIcpif=16
MediaSetting=flow-around
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=
OriginalCallingOctet=0x0
OriginalCalledNumber=
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x0
TranslatedCallingNumber=555-0100
TranslatedCallingOctet=0x21
TranslatedCalledNumber=
TranslatedCalledOctet=0xC1
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwOutpulsedCallingNumber=555-0101
GwOutpulsedCallingOctet3=0x21
GwOutpulsedCallingOctet3a=0x81
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurationCallTimerStamp=
LongDurationCallDuration=
Username=
The table below describes the significant fields shown in the display, in alphabetical order.
Table 3 show call history media Field Descriptions
Field
Description
CallDuration
Length of the call, in hours, minutes, and seconds, hh:mm:ss.
CallOrigin
Call origin: not answer or originate.
ChargedUnits
Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.
CodecBytes
Payload size, in bytes, for the codec used.
CoderTypeRate
Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.
ConnectionId
Global call identifier for this gateway call.
ConnectTime
Time, in ms, during which the call was connected.
GapFillWithInterpolation
Duration, in ms, of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithRedundancy
Duration, in ms, of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithPrediction
Duration, in ms, of the voice signal played out with a signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1 compression algorithms.
GapFillWithSilence
Duration, in ms, of a voice signal replaced with silence because voice data was lost or not received in time for this call.
GENERIC
Generic or common parameters; that is, parameters that are common for VoIP and telephony call legs.
H323 call-legs
Total H.323 call legs for which call records are available.
HiWaterPlayoutDelay
High-water-mark voice playout first in first out (FIFO) Delay during this call, in ms.
Index
Dial peer identification number.
InfoType
Information type for this call; for example, voice, speech, or fax.
LogicalIfIndex
Index number of the logical interface for this call.
LoWaterPlayoutDelay
Low-water-mark voice playout FIFO delay during this call, in ms.
OnTimeRvPlayout
Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.
PeerAddress
Destination pattern or number associated with this peer.
PeerId
ID value of the peer table entry to which this call was made.
PeerIfIndex
Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.
PeerSubAddress
Subaddress when this call is connected.
ReceiveBytes
Number of bytes received by the peer during this call.
ReceiveDelay
Average playout FIFO delay plus the decoder delay during this voice call, in ms.
ReceivePackets
Number of packets received by this peer during this call.
ReleaseSource
Number value of the release source.
RemoteIPAddress
Remote system IP address for the VoIP call.
RemoteUDPPort
Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.
RoundTripDelay
Voice packet round-trip delay, in ms, between the local and remote systems on the IP backbone for this call.
SelectedQoS
Selected Resource Reservation Protocol (RSVP) quality of service (QoS) for this call.
SessionProtocol
Session protocol used for an Internet call between the local and remote routers through the IP backbone.
SessionTarget
Session target of the peer used for this call.
SetupTime
Value of the system UpTime, in ms, when the call associated with this entry was started.
SIP call-legs
Total Session Initiation Protocol (SIP) call legs for which call records are available.
Telephony call-legs
Total telephony call legs for which call records are available.
TransmitBytes
Number of bytes sent by this peer during this call.
TransmitPackets
Number of packets sent by this peer during this call.
VAD
Whether voice activation detection (VAD) was enabled for this call.
Related Commands
Command
Description
dial-control-mib
Sets the maximum number of calls contained in the table.
showcallactivemedia
Displays call information for media calls in progress.
show call history video
To display call history information for signaling connection control protocol (SCCP) video calls, use the
showcallhistoryvideo command in user EXEC or privileged EXEC mode.
showcallhistoryvideo
[ [brief]
[ ididentifier ] | compact
[ duration
{ less | more }
seconds ] | lastnumber ]
Syntax Description
brief
(Optional) Displays a truncated version of video call history information.
ididentifier
(Optional) Displays only the video call history with the specified identifier. Range is a hexadecimal value from 1 to FFFF.
compact
(Optional) Displays a compact version of video call history information.
duration
(Optional) Displays the call history for the specified time duration.
less
Displays the call history for shorter duration calls.
more
Displays the call history for longer duration calls.
seconds
Time, in seconds. The range is from 1 to 2147483647.
lastnumber
(Optional) Displays the last calls connected, where the number of calls that appear is defined by the
numberargument. The range is from 1 to100 .
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Cisco IOS Release
Cisco Product
Modification
12.4(4)XC
Cisco Unified CME 4.0
This command was introduced.
12.4(9)T
Cisco Unified CME 4.0
This command was integrated into Cisco IOS Release 12.4(9)T.
12.4(16); 12.4(15)T
Cisco Unified CME 4.0
This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.
Examples
The following is sample output from theshowcallhistoryvideo command with the
compact option:
Router# show call history video compact
<callID> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
Total call-legs: 2
241 ANS T17 g729r8 VOIP P555-0100 192.0.2.0:16926
242 ORG T17 g729r8 TELE-VIDEO P555-0101
The table below describes the significant fields shown in the display.
Table 4 show call history video Field Descriptions
Field
Description
callID
Unique identifier for the call leg.
A/O
Call leg was an answer (ANS) or an originator (ORG).
FAX
Fax number for the call leg.
T<sec>
Duration in seconds.
Codec
Codec used for this call leg.
type
Call type for this call leg.
Peer Address
Called or calling number of the remote peer.
IP R<ip>:<udp>
IP address and port number
Total call-legs
Total number of call legs for this call.
Related Commands
Command
Description
showcallactivevideo
Displays call information for SCCP video calls in progress.
show call history video record
To display information about incoming and outgoing video calls, use the showcallhistoryvideorecordcommand in privileged EXEC mode.
showcallhistoryvideorecord
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.0(5)XK
This command was introduced on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
The following example displays information about two video calls:
(Optional) Displays a truncated version of the call history table.
ididentifier
(Optional) Displays only the call with the specified identifier. Range is from 1 to FFFF.
compact
(Optional) Displays a compact version of the call history table.
dest-route-stringtag
(Optional) Displays only the call with the specified destination route tag value. The range is from 1 to 10000.
durationseconds
(Optional) Displays history information for calls that are longer or shorter than the value of the specified
seconds
argument. The arguments and keywords are as follows:
less--Displays calls shorter than the
seconds value.
more--Displays calls longer than the
seconds value.
seconds--Elapsed time, in seconds. Range is from 1 to 2147483647.
lastnumber
(Optional) Displays the last calls connected, where the number of calls that appear is defined by the
numberargument. Range is from 1 to100 .
redirect
(Optional) Displays information about calls that were redirected using Release-to-Pivot (RTPvt) or Two B-Channel Transfer (TBCT). The keywords are as follows:
rtpvt--Displays information about RTPvt calls.
tbct--Displays information about TBCT calls.
stats
(Optional) Displays information about digital signal processing (DSP) voice quality metrics.
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series.
12.0(3)XG
Support was added for Voice over Frame Relay (VoFR) on the Cisco 2600 series and Cisco 3600 series.
12.0(4)XJ
This command was modified for store-and-forward fax.
12.0(4)T
The
brief keyword was added, and the command was implemented on the Cisco 7200 series.
12.0(5)XK
This command was implemented on the Cisco MC3810.
12.0(7)XK
The
brief keyword was implemented on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(5)XM
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XA
The output of this command was modified to indicate whether a specified call has been established using Annex E.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support was not included for the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
12.2(11)T
Support was added for Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
12.2(13)T
The ReleaseSource field was added to the Field Description table, and the
record keyword was deleted from the command name.
12.3(1)
The
redirect keyword was added.
12.4(2)T
The LocalHostname display field was added to the VoIP call leg record.
12.4(11)XW
The
stats keyword was added.
12.4(15)T
The Port and BearerChannel display fields were added to the TELE call leg record of the command output.
12.4(16)
The Port and BearerChannel display fields were added to the TELE call leg record of the command output.
12.4(22)T
Command output was updated to show IPv6 information.
15.3(3)M
This command was modified. Thedest-route-string keyword was added.
Cisco IOS XE Release 3.10S
This command was integrated into Cisco IOS XE Release 3.10S.
Usage Guidelines
This command displays a call-history table that contains a list of voice calls connected through the router in descending time order. The maximum number of calls contained in the table can be set to a number from 0 to 500 using thedial-control-mibcommand in global configuration mode. The default maximum number of table entries is 50. Each call record is aged out of the table after a configurable number of minutes has elapsed. The timer value is also specified by the
dial-control-mib command. The default timer value is 15 minutes.
You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the
last keyword, and define the number of calls to be displayed with the number argument.
To display a truncated version of the call history table, use the
brief keyword.
Use the
showcallactivevoiceredirectcommand to review records for calls that implemented RTPvt or TBCT.
When a call is active, you can display its statistics by using the
showcallactivevoice command.
Use the show call active voicedest-route-string command to display only the active voice calls with call routing configured using specified destination-route-string globally and at the dial-peer level.
Examples
The following is sample output from theshowcallhistoryvoice command:
Router# show call history voice
GENERIC:
SetupTime=104648 ms
Index=1
PeerAddress=55240
PeerSubAddress=
PeerId=2
PeerIfIndex=105
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing.
ConnectTime=104964
DisconectTime=143329
CallDuration=00:06:23
CallOrigin=1
ChargedUnits=0
InfoType=speech
TransmitPackets=37668
TransmitBytes=6157536
ReceivePackets=37717
ReceiveBytes=6158452
VOIP:
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
CallID=2
RemoteIPAddress=10.14.82.14
RemoteUDPPort=18202
RoundTripDelay=2 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=TRUE
SessionProtocol=cisco
SessionTarget=ipv4:10.14.82.14
OnTimeRvPlayout=40
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
ReceiveDelay=67 ms
LostPackets=0 ms
EarlyPackets=0 ms
LatePackets=0 ms
VAD = enabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
SignalingType=cas
Modem passthrough signaling method is nse
Buffer Fill Events = 0
Buffer Drain Events = 0
Percent Packet Loss = 0
Consecutive-packets-lost Events = 0
Corrected packet-loss Events = 0
Last Buffer Drain/Fill Event = 373sec
Time between Buffer Drain/Fills = Min 0sec Max 0sec
GENERIC:
SetupTime=104443 ms
Index=2
PeerAddress=50110
PeerSubAddress=
PeerId=100
PeerIfIndex=104
LogicalIfIndex=10
DisconnectCause=10
DisconnectText=normal call clearing.
ConnectTime=104964
DisconectTime=143330
CallDuration=00:06:23
CallOrigin=2
ChargedUnits=0
InfoType=speech
TransmitPackets=37717
TransmitBytes=5706436
ReceivePackets=37668
ReceiveBytes=6609552
TELE:
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
CallID=3
Port=3/0/0 (3)
BearerChannel=3/0/0.1
TxDuration=375300 ms
VoiceTxDuration=375300 ms
FaxTxDuration=0 ms
CoderTypeRate=g711ulaw
NoiseLevel=-75
ACOMLevel=11
SessionTarget=
ImgPages=0
The following example from a Cisco AS5350 router displays a sample of voice call history records showing release source information:
The following is sample output from the
showcallhistoryvoiceredirect command:
Router# show call history voice redirect tbct
index=2, xfr=tbct-notify, status=redirect_success, start_time=*00:12:25.981 UTC Mon Mar 1 1993, ctrl name=T1-2/0, tag=13
index=3, xfr=tbct-notify, status=redirect_success, start_time=*00:12:25.981 UTC Mon Mar 1 1993, ctrl name=T1-2/0, tag=13
index=4, xfr=tbct-notify, status=redirect_success, start_time=*00:13:07.091 UTC Mon Mar 1 1993, ctrl name=T1-2/0, tag=12
index=5, xfr=tbct-notify, status=redirect_success, start_time=*00:13:07.091 UTC Mon Mar 1 1993, ctrl name=T1-2/0, tag=12
Number of call-legs redirected using tbct with notify:4
The table below describes the significant fields shown in the
showcallhistoryvoiceredirecttbct display.
Table 5 show call history voice redirect Field Descriptions
Field
Description
index
Index number of the record in the history file.
xfr
Whether TBCT or TBCT with notify has been invoked.
status
Status of the redirect request.
start_time
Time, in hours, minutes, and seconds when the redirected call began.
ctrl name
Name of the T1 controller where the call originated.
tag
Call tag number that identifies the call.
Number of call-legs redirected using tbct with notify
Total number of call legs that were redirected using TBCT with notify.
Related Commands
Command
Description
dial-control-mib
Set the maximum number of calls contained in the table.
showcallactivefax
Displays call information for fax transmissions that are in progress.
showcallactivevoice
Displays call information for voice calls that are in progress.
showcallhistoryfax
Displays the call history table for fax transmissions.
showdial-peervoice
Displays configuration information for dial peers.
shownum-exp
Displays how the number expansions are configured in VoIP.
showvoiceport
Displays configuration information about a specific voice port.
show call language voice
To display a summary of languages configured and the URLs of the corresponding Tool Command Language (TCL) modules for the languages that are not built-in languages, use the showcalllanguagevoicecommandinEXEC mode.
showcalllanguagevoice
[ language | summary ]
Syntax Description
language
(Optional) Two-character prefix configured with the calllanguagevoice command in global configuration mode, either for a prefix for a built-in language or one that you have defined; for example, "en" for English or "ru" for Russian.
summary
(Optional) Summary of all the languages configured and the URLs for the TCL modules other than built-in languages.
Command Modes
EXEC (#)
Command History
Release
Modification
12.2(2)T
This command was introduced.
Usage Guidelines
This command is similar to theshowcallapplicationvoice command. If a language is built in, the URL listed reads "fixed." If you decide to overwrite the built-in language with your own language, the word "fixed" in the URL column changes to the actual URL where your new application lives.
Examples
The following command displays a summary of the configured languages:
Router# show call language voice summary
name url
sp fixed
ch fixed
en fixed
ru tftp://dirt/fwarlau/scripts/multilag/ru_translate.tcl
The following command displays information about Russian-language configuration:
Router# show call language voice ru
ru_translate.tcl
ru_translate.tcl~
singapore.cfg
test.tcl
people% more ru_translate.tcl
# Script Locked by: farmerj
# Script Version: 1.1.0.0
# Script Lock Date: Sept 24 2000
# ca_translate.tcl
#------------------------------------------------------------------
# Sept 24, 2000 Farmer Joe
#
# Copyright (c) 2000 by Cisco Systems, Inc.
# All rights reserved.
#------------------------------------------------------------------
#<snip>...
...set prefix ""
#puts "argc"
#foreach arg $argv {
#puts "$arg"
# translates $arg
# puts "\t\t**** $prompt RETURNED"
#}
Field descriptions should be self-explanatory.
Related Commands
Command
Description
calllanguagevoice
Configures a TCL module.
calllanguagevoiceload
Loads or reloads a TCL module from the configured URL location.
debugvoipivr
Specifies the type of VoIP IVR debug output that you want to view.
showcallapplicationvoice
Shows and describes applications.
show call leg
To display event logs and statistics for voice call legs, use the showcalllegcommand in privileged EXEC mode.
showcallleg
{ active | history }
[ summary | [ lastnumber | leg-idleg-id ]
[ event-log | info ] ]
Syntax Description
active
Statistics or event logs for active call legs.
history
Statistics or event logs for terminated call legs.
summary
(Optional) A summary of each call leg.
lastnumber
(Optional) Selected number of most recent call legs. Not available with active keyword.
leg-idleg-id
(Optional) A specific call leg. Output displays event logs or statistics for that call leg.
event-log
(Optional) Event logs for call legs.
info
(Optional) Statistics for call legs.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.3(8)T
This command was introduced.
Usage Guidelines
If you use the leg-id keyword, only statistics or event logs for that call leg display. To display event logs with this command, you must enable event logging with the calllegevent-log command.
Examples
The following is sample output from the showcallleg command using different keywords:
Router# show call leg active summary
G<id> L<id> Elog A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
G11DC L A Y ANS T2 None TELE P4085550198
Total call-legs: 1
Router# show call leg active event-log
Event log for call leg ID: A Connection ID: 11DC
buf_size=4K, log_lvl=INFO
<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>
A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing
A:1057277701:72:INFO: Dialpeer = 1
A:1057277701:77:INFO: Digit collection
A:1057277701:78:INFO: Call connected using codec None
Total call-legs: 1
Router# show call leg active info
Information for call leg ID: A Connection ID: 11DC
GENERIC:
SetupTime=3012940 ms
Index=1
PeerAddress=4085550198
PeerSubAddress=
PeerId=1
PeerIfIndex=329
LogicalIfIndex=253
ConnectTime=301295
CallDuration=00:00:20
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=2
TransmitPackets=412
TransmitBytes=98880
ReceivePackets=0
ReceiveBytes=0
TELE:
ConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]
IncomingConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]
TxDuration=20685 ms
VoiceTxDuration=0 ms
FaxTxDuration=0 ms
CoderTypeRate=None
NoiseLevel=-120
ACOMLevel=90
OutSignalLevel=-50
InSignalLevel=-41
InfoActivity=0
ERLLevel=38
EchoCancellerMaxReflector=16685
SessionTarget=
ImgPages=0
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=4085550198
OriginalCallingOctet=0x0
OriginalCalledNumber=52927
OriginalCalledOctet=0xE9
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=4085550198
TranslatedCallingOctet=0x0
TranslatedCalledNumber=52927
TranslatedCalledOctet=0xE9
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=52927
GwReceivedCalledOctet3=0xE9
GwReceivedCallingNumber=4085550198
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x81
Total call-legs: 1
For a description of the call leg statistics, see the description for the showcallactivevoice command.
Router# show call leg active leg-id A
Call Information - Connection ID: 11DC , Call Leg ID: A
GENERIC:
SetupTime=3012940 ms
Index=1
PeerAddress=4085550198
PeerSubAddress=
PeerId=1
PeerIfIndex=329
LogicalIfIndex=253
ConnectTime=301295
CallDuration=00:00:40
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=2
TransmitPackets=824
TransmitBytes=197760
ReceivePackets=0
ReceiveBytes=0
TELE:
ConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]
IncomingConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]
TxDuration=20685 ms
VoiceTxDuration=0 ms
FaxTxDuration=0 ms
CoderTypeRate=None
NoiseLevel=-120
ACOMLevel=90
OutSignalLevel=-50
InSignalLevel=-41
InfoActivity=0
ERLLevel=38
EchoCancellerMaxReflector=16685
SessionTarget=
ImgPages=0
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=4085550198
OriginalCallingOctet=0x0
OriginalCalledNumber=52927
OriginalCalledOctet=0xE9
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=4085550198
TranslatedCallingOctet=0x0
TranslatedCalledNumber=52927
TranslatedCalledOctet=0xE9
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=52927
GwReceivedCalledOctet3=0xE9
GwReceivedCallingNumber=4085550198
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x81
Call Event Log - Connection ID: 11DC , Call Leg ID: A
buf_size=4K, log_lvl=INFO
<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>
A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing
A:1057277701:72:INFO: Dialpeer = 1
A:1057277701:77:INFO: Digit collection
A:1057277701:78:INFO: Call connected using codec None
Call-leg found: 1
Router# show call leg active leg-id A event-log
Call Event Log - Connection ID: 11DC , Call Leg ID: A
buf_size=4K, log_lvl=INFO
<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>
A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing
A:1057277701:72:INFO: Dialpeer = 1
A:1057277701:77:INFO: Digit collection
A:1057277701:78:INFO: Call connected using codec None
Call-leg found: 1
Router# show call leg history summary
G<id> L<id> Elog A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp> disc-cause
G11DB L 7 Y ANS T24 None TELE P4085550198 D10
G11DC L A Y ANS T159 None TELE P4085550198 D10
Total call-legs: 2
Router# show call leg history last 1
Call Information - Connection ID: 11DC , Call Leg ID: A
GENERIC:
SetupTime=3012940 ms
Index=4
PeerAddress=4085550198
PeerSubAddress=
PeerId=1
PeerIfIndex=329
LogicalIfIndex=253
DisconnectCause=10
DisconnectText=normal call clearing (16)
ConnectTime=301295
DisconnectTime=317235
CallDuration=00:02:39
CallOrigin=2
ReleaseSource=1
ChargedUnits=0
InfoType=speech
TransmitPackets=2940
TransmitBytes=705600
ReceivePackets=0
ReceiveBytes=0
TELE:
ConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]
IncomingConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]
TxDuration=20685 ms
VoiceTxDuration=0 ms
FaxTxDuration=0 ms
CoderTypeRate=None
NoiseLevel=-120
ACOMLevel=90
SessionTarget=
ImgPages=0
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=4085550198
OriginalCallingOctet=0x0
OriginalCalledNumber=52927
OriginalCalledOctet=0xE9
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=4085550198
TranslatedCallingOctet=0x0
TranslatedCalledNumber=52927
TranslatedCalledOctet=0xE9
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=52927
GwReceivedCalledOctet3=0xE9
GwReceivedCallingNumber=4085550198
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x81
Call Event Log - Connection ID: 11DC , Call Leg ID: A
buf_size=4K, log_lvl=INFO
<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>
A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing
A:1057277701:72:INFO: Dialpeer = 1
A:1057277701:77:INFO: Digit collection
A:1057277701:78:INFO: Call connected using codec None
A:1057277860:150:INFO: Inform application call disconnected (cause = normal call clearing (16))
A:1057277860:154:INFO: Call disconnected (cause = normal call clearing (16))
A:1057277860:155:INFO: Call released
Total call-legs: 1
Total call-legs with event log: 1
Router# show call leg history leg-id A event-log
Call Event Log - Connection ID: 11DC , Call Leg ID: A
buf_size=4K, log_lvl=INFO
<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>
A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing
A:1057277701:72:INFO: Dialpeer = 1
A:1057277701:77:INFO: Digit collection
A:1057277701:78:INFO: Call connected using codec None
A:1057277860:150:INFO: Inform application call disconnected (cause = normal call clearing (16))
A:1057277860:154:INFO: Call disconnected (cause = normal call clearing (16))
A:1057277860:155:INFO: Call released
Call-leg matched ID found: 1
Call-legs matched ID with event log: 1
Field descriptions should be self-explanatory.
Related Commands
Command
Description
calllegevent-log
Enables event logging for voice, fax, and modem call legs.
calllegevent-logdumpftp
Enables the voice gateway to write the contents of the call-leg event log buffer to an external file.
calllegevent-logerror-only
Restricts event logging to error events only for voice call legs.
calllegevent-logmax-buffer-size
Sets the maximum size of the event log buffer for each call leg.
callleghistoryevent-logsave-exception-only
Saves to history only event logs for call legs that had at least one error.
monitorcalllegevent-log
Displays the event log for an active call leg in real-time.
show call media forking
To display currently active media forking sessions, use the show call
media forking command in user EXEC or privileged EXEC mode.
showcallmediaforking
Syntax Description
This command has no arguments or keywords.
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Release
Modification
15.2(2)T
This command was introduced.
Usage Guidelines
Use this command to verify that media forking was successful for relevant anchor
legs.
Examples
The following example is a sample output from the show call media
forking command..
Router# show call media forking
Warning: Output may be truncated if sessions are added/removed concurrently!
Session Call n/f Destination (port address)
7 6 far 1234 1.5.35.254
8 6 near 5678 1.5.35.254
The table below describes the fields that are displayed in the output.
Field
Description
Session
Session Identifier.
Call
Call Leg identifier in hexadecimal. It must
match the Call ID from the show call leg active command.
n/f
Direction (Near End or Far End) of the voice
stream that was forked.
Destination (port address)
Destination for the forked packets. It consists of the
folllowing:
RTP Port
IP Address
show callmon
To display call monitor information, use theshowcallmon command in user EXEC or privileged EXEC mode.
showcallmon
{ call | gcid | subscription | trace
{ all | event
{ all | call | connection } | exec | server | subscription | trigger } }
Syntax Description
call
Displays the active call monitor calls.
gcid
Displays the active global call ID information.
subscription
Displays the subscription information.
trace
Displays the trace information.
all
Displays all types of traces based on time.
event
Displays the event trace information.
all--Displays all event traces.
call--Displays event traces related to a call.
connection--Displays the event traces related to a connection.
exec
Displays all critical execution traces.
server
Displays all session server up or down traces.
subscription
Displays all subscription traces.
trigger
Displays the entire trigger structure by index.
Command Modes
User EXEC (>)
Privileged EXEC (#))
Command History
Release
Modification
12.4(22)T
This command was introduced.
Examples
The following sample output from the
showcallmoncall command shows active call monitor calls:
Router# show callmon call
line dn sub_id number of call instance
6401, 1
callID 2038(19D7), *cg = 6401, cd = 6601
6601, 1
callID 2039(19D7), cg = 6401, *cd = 6601
The table below describes the significant fields shown in the display.
Table 6 show callmon call Field Descriptions
Field
Description
dn
Directory number.
number of call
Number of call instances.
instance
Contents of the call instance.
The following sample output from the
showcallmongcid command shows the active global call ID information:
The table below describes the significant fields shown in the display.
Table 7 show callmon gcid Field Descriptions
Field
Description
GCID
Global call ID.
CallIDs
Active call IDs.
Related Commands
Command
Description
callmonitor
Enables call monitoring messaging functionality on a SIP endpoint in a VoIP network.
show call prompt-mem-usage
To display the amount of memory used by prompts, use the
showcallprompt-mem-usagecommand in privileged EXEC mode.
showcallprompt-mem-usage [detail]
Syntax Description
detail
(Optional) Displays details about memory usage and names of tones used.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.2(15)T
This command was introduced.
12.3(7)T
The
detail keyword was added.
Usage Guidelines
Use this command to display the number of prompts loaded into the gateway, the amount of memory used by the prompts, the number of prompts currently being played, and the status of prompt loads.
For calls transferred by a Cisco CallManager Express (Cisco CME) system, the ringback tone generation for commit-at-alerting uses an interactive voice response (IVR) prompt playback mechanism. Ringback tone is played to the transferred party by the Cisco CME system associated with the transferring party.
The system automatically generates tone prompts as needed on the basis of the network-locale setting made in the Cisco CME system.
Examples
The following sample output shows details about the memory usage of the prompts that are used.
Router# show call prompt-mem-usage
Prompt memory usage:
config’d wait active free mc total ms total
file(s) 0200 0010 0001 00189 00011 00002
memory 02097152 00081259 00055536 01960357 00136795
Prompt load counts: (counters reset 0)
success 11(1st try) 0(2nd try), failure 0
Other mem block usage:
mcDynamic mcReader
gauge 00001 00001
Number of prompts playing: 1
Number of start delays : 0
MCs in the ivr MC sharing table
===============================
Media Content: NoPrompt (0x83C64554)
URL:
cid=0, status=MC_READY size=24184 coding=g711ulaw refCount=0
Media Content: tone://GB_g729_tone_ringback (0x83266EC8)
URL: tone://GB_g729_tone_ringback
The table below describes the significant fields shown in the display.
Table 8 show call prompt-mem-usage Field Descriptions
Field
Description
file(s)
Number of prompts in different queues.
file(s) - config’d
Maximum number of configured prompts that can be simultaneously available in memory. In the sample output, the value of 200 in this field means that loading the 201st prompt results in the oldest prompts being removed.
file(s) -wait
Number of prompts in the wait queue that are not being used in any call and are ready to be deleted when there is no space for a new prompt. This field lists older prompts that can be deleted.
file(s) - active
Number of prompts that are being used in active calls. These prompts cannot be deleted.
file(s) - free
Number of prompts that can be loaded without deleting any prompt from the wait queue. This is the number of configured prompts (listed under config’d) minus the total number of prompts in the wait and active states.
file(s) - mc total
Total number of prompts in the wait and active states.
ms total
Number of media streams that are currently active. One media stream is used for playing INBOX prompts. A prompt is considered an INBOX prompt if its URL is either flash:, http:, ram:, or tftp:.
memory
Displays the memory used by prompts, in bytes.
memory - config’d
Maximum amount of memory configured to be available for prompts.
memory - wait
Total amount of memory used by prompts in the wait list.
memory - active
Total amount of memory used by prompts in the active list.
memory - free
Amount of available memory. This is the amount of configured prompts (listed under config’d) memory minus the total amount of memory used by the prompts in the wait and active lists.
memory - mc total
Total amount of memory used by prompts in the wait and active lists.
Prompt load counts
Number of successful attempts to load a prompt on the first try and on the second try, and the number of attempts to load a prompt that failed.
mcDynamic
Number of dynamic element queues that are active. A dynamic element queue is a list of prompts that are played together.
mcReader
Number of mcReaders that are active. An mcReader is used for playing one mcDynamic queue of prompts. An mcReader is used only if the mcDynamic contains prompts that are associated with one of the following types of URL: flash:, http:, ram:, or tftp:.
Number of prompts playing
Number of prompts that are currently playing.
Number of start delays
Number of times that prompts failed to start and have subsequently restarted.
MCs in the ivr MC sharing table
The fields below this line of text refer to each media content (prompt) currently cached in memory. In the sample output, the only cached prompt is the built-in default prompt named "NoPrompt."
Media Content
Name of the prompt, which is derived from the audio file URL (the characters after the last "/" in the URL). The address in parentheses is the memory location of the prompt.
URL
Location of the file for the prompt that is playing. In the case of the default prompt, NoPrompt, no URL is given.
cid
Call identification number of the call that initiated the loading of the prompt.
status
Status of the media content. The following values are possible:
MC_NOT_READY--Initial status for media content. When the media content is successfully loaded, the status will change to MC_READY.
MC_READY--Media content is loaded into memory and ready for use.
MC_LOAD_FAIL--Media content failed to load.
size
Size of the media content, in bytes.
coding
Type of encoding used by the media content.
refCount=0
Number of calls to which this media content is currently being streamed.
show call resource voice stats
To display resource statistics for an H.323 gateway, use the show call resource voice stats command in privileged EXEC mode.
showcallresourcevoicestats
[ ds0 | dsp ]
Syntax Description
ds0
(Optional) Specifies the voice digital signal level zero (DS0) resource statistics information.
dsp
(Optional) Specifies the voice digital signal processor (DSP) resource statistics information.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.0(5)T
This command was introduced.
12.1(5)XM2
This command was integrated into Cisco IOS Release 12.1(5)XM2
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB1
This command was integrated into Cisco IOS Release into 12.2(2)XB1.
12.2(8)T
This command was modified. Support for the Cisco AS5300,Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 series routers is not included in this release.
12.4(22)T
This command was modified. The
ds0 and
dsp keywords were added.
Usage Guidelines
The
showcallresourcevoicestats command displays the H.323 resources that are monitored when theresourcethreshold command is used to configure resource threshold reporting.
Examples
The following is sample output from the show call resource voice stats command, which shows the resource statistics for an H.323 gateway:
The table below describes significant fields shown in this output.
Table 9 show call resource voice stats Field Descriptions
Statistic
Definition
Total channels
Number of channels physically configured for the resource.
Inuse channels
Number of addressable channels that are in use. This value includes all channels that either have active calls or have been reserved for testing.
Disabled channels
Number of addressable channels that are physically down or that have been disabled administratively with the
shutdown or
busyout command.
Pending channels
Number of addressable channels that are pending in loadware download.
Free channels
Number of addressable channels that are free.
Addressable channels
Number of channels that can be used for a specific type of dialup service, such as H.323, which includes all the DS0 resources that have been associated with a voice plain old telephone service (POTS) dial plan profile.
Related Commands
Command
Description
resourcethreshold
Configures a gateway to report H.323 resource availability to the gatekeeper of the gateway.
showcallresourcevoicethreshold
Displays the threshold configuration settings and status for an H.323 gateway.
show call resource voice threshold
To display the threshold configuration settings and status for an H.323 gateway, use the
showcallresourcevoicethreshold command in privileged EXEC mode.
showcallresourcevoicethreshold
[ ds0 | dsp ]
Syntax Description
ds0
(Optional) Specifies the voice digital signal level zero (DS0) resource statistics information.
dsp
(Optional) Specifies the voice digital signal processor (DSP) resource statistics information.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.0(5)T
This command was introduced.
12.1(5)XM2
This command was integrated into Cisco IOS Release 12.1(5)XM2
12.2(2)XB1
This command was integrated into Cisco IOS Release into 12.2(2)XB1.
12.4(22)T
This command was modified. The
ds0 and
dsp keywords were added.
Usage Guidelines
The
showcallresourcevoicethreshold command displays the H.323 resource thresholds that are configured with theresourcethreshold command.
Examples
The following is sample output from the show call resource voice threshold command, which shows the resource threshold settings and status for an H.323 gateway:
Router# show call resource voice threshold
Resource Monitor - Dial-up Resource Threshold Information:
DS0 Threshold:
Client Type: h323
High Water Mark: 70
Low Water Mark: 60
Threshold State: init
DSP Threshold:
Client Type: h323
High Water Mark: 70
Low Water Mark: 60
Threshold State: low_threshold_hit
The table below describes the significant fields shown in the display.
Table 10 show call resource voice threshold Field Descriptions
Field
Description
High Water Mark
Resource-utilization level that triggers a message indicating that H.323 resource use is high. The range is 1 to 100. A value of 100 indicates that the resource is unavailable.The default is 90.
Low Water Mark
Resource-utilization level that triggers a message indicating that H.323 resource use has dropped below the high-usage level. The range is 1 to 100. The default is 90.
Related Commands
Command
Description
resourcethreshold
Configures a gateway to report H.323 resource availability to the gatekeeper of the gateway.
showcallresourcevoicestats
Displays resource statistics for an H.323 gateway.
show call rsvp-sync conf
To display the configuration settings for Resource Reservation Protocol (RSVP) synchronization, use the
showcallrsvp-syncconfcommand in privileged EXEC mode.
showcallrsvp-syncconf
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.1(3)XI1
This command was introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200, Cisco MC3810, Cisco AS5300, and Cisco AS5800.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
Support for the Cisco AS5300,Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
Examples
The following example shows sample output from this command:
Router# show call rsvp-sync conf
VoIP QoS: RSVP/Voice Signaling Synchronization config:
Overture Synchronization is ON
Reservation Timer is set to 10 seconds
The table below describes significant fields shown in this output.
Table 11 show call rsvp-sync conf Field Descriptions
Field
Description
Overture Synchronization is ON
Indicates whether RSVP synchronization is enabled.
Reservation Timer is set to xx seconds
Number of seconds for which the RSVP reservation timer is configured.
Related Commands
Command
Description
callrsvp-sync
Enables synchronization between RSVP and the H.323 voice signaling protocol.
callrsvp-syncresv-timer
Sets the timer for RSVP reservation setup.
debugcallrsvp-syncevents
Displays the events that occur during RSVP synchronization.
showcallrsvp-syncstats
Displays statistics for calls that attempted RSVP reservation.
show call rsvp-sync stats
To display statistics for calls that attempted Resource Reservation Protocol (RSVP) reservation, use the
showcallrsvp-syncstats command in privileged EXEC mode.
showcallrsvp-syncstats
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.1(3)XI1
This command was introduced.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Examples
The following example shows sample output from this command:
Router# show call rsvp-sync stats
VoIP QoS:Statistics Information:
Number of calls for which QoS was initiated : 18478
Number of calls for which QoS was torn down : 18478
Number of calls for which Reservation Success was notified : 0
Total Number of PATH Errors encountered : 0
Total Number of RESV Errors encountered : 0
Total Number of Reservation Timeouts encountered : 0
The table below describes significant fields shown in this output.
Table 12 show call rsvp-sync stats Field Descriptions
Field
Description
Number of calls for which QoS was initiated
Number of calls for which RSVP setup was attempted.
Number of calls for which QoS was torn down
Number of calls for which an established RSVP reservation was released.
Number of calls for which Reservation Success was notified
Number of calls for which an RSVP reservation was successfully established.
Total Number of PATH Errors encountered
Number of path errors that occurred.
Total Number of RESV Errors encountered
Number of reservation errors that occurred.
Total Number of Reservation Timeouts encountered
Number of calls in which the reservation setup was not complete before the reservation timer expired.
Related Commands
Command
Description
callrsvp-sync
Enables synchronization between RSVP and the H.323 voice signaling protocol.
callrsvp-syncresv-timer
Sets the timer for RSVP reservation setup.
debugcallrsvp-syncevents
Displays the events that occur during RSVP synchronization.
showcallrsvp-syncconf
Displays the RSVP synchronization configuration.
show call spike status
To display the configured call spike threshold and statistics for incoming calls, use the
showcallspikestatus command in privileged EXEC mode.
showcallspikestatus
[ dial-peertag ]
Syntax Description
dial-peer
(Optional) Displays configuration information for a dial peer.
tag
(Optional) Specifies the dial peer identifying number. Range is from 1 to 2147483647.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. This command was not supported on any other platforms in this release.
12.2(8)T
This command was implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 was not included in this release.
12.2(11)T
Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 was added in this release.
15.1(3)T
This command was modified. The output fields of the command were modified to include the output at the dial peer level.
Examples
The following is sample output from this command:
Router# show call spike status
Call Spiking:Configured
Call spiking :NOT TRIGGERED
total call count in sliding window ::20
The table below describes the significant fields shown in the display.
Table 13 show call spike status Field Descriptions
Field
Description
Call Spiking
Current enabled state of call spiking.
Call Spiking
Details if the call spiking limit has been triggered.
total call count in sliding window
Number of calls during the spiking interval.
Router# show call spike status dial-peer 400
TAG CONFIG SPIKED TOTAL REJECTED CALLS REJECTED CALLS
400 YES NO 4 0
The table below describes the significant fields shown in the display.
Table 14 show call spike status (dial peer) Field Descriptions
Field
Description
TAG
Dial peer tag.
CONFIG
Displays if the
callspike command has been configured.
SPIKED
Details if the call spiking limit has been triggered.
TOTAL REJECTED CALLS
Displays the number of calls rejected due to a call spike in the dial peer.
REJECTED CALLS
Displays the number of calls rejected when the call spike was triggered until the call spike control was released.
Related Commands
Command
Description
callspike
Configures the limit for the number of incoming calls in a short period of time.
show call threshold
To display enabled triggers, current values for configured triggers, and the number of application programming interface (API) calls that were made to global and interface resources, use the
showcallthreshold command in privileged EXEC mode.
showcallthreshold
{ config | status [unavailable] | stats }
Syntax Description
config
Displays the current threshold configuration.
status
Displays the status of all configured triggers and whether or not the CPU is available.
unavailable
(Optional) Displays the status for all unavailable resources.
stats
Displays statistics for API calls; that is, the resource-based measurement.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. This command is not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. This command is not supported on any other platforms in this release.
12.2(8)T
This command was implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was implemented on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850.
15.2(2)T
This command was modified. The output was modified to display the configured bandwidth threshold, bandwidth availability, and call admission control statistics.
Examples
The following is sample output from theshowcallthresholdconfig command:
Router# show call threshold config
Some resource polling interval:
CPU_AVG interval: 60
Memory interval: 5
IF Type Value Low High Enable
----- ---- ----- ---- ---- ------
Serial3/1:23 int-calls 0 107 107 N/A
N/A cpu-avg 0 70 90 busy&treat
The following is sample output from the
showcallthresholdstatus command:
Router# show call threshold status
Status IF Type Value Low High Enable
------ --- ------ ---- ---- ---- -----
Avail N/A total-calls 0 5 5000 busyout
Avail N/A cpu-avg 0 5 65 busyout
The following is sample output from the
showcallthresholdstatusunavailable command:
Router# show call threshold status unavailable
Unavailable configured resources at the current time:
IF Type Value Low High Enable
---- ----- ----- ---- ---- -----
The following is sample output from the
showcallthresholdstats command:
Router# show call threshold stats
Total resource check: 0
successful: 0
failed: 0
The table below describes significant fields shown in this output.
Table 15 show call threshold Field Descriptions
Field
Description
CPU_AVG interval
Interval of configured trigger CPU_AVG.
Memory interval
Interval of configured trigger Memory.
IF
Interface type and number.
Type
Type of resource.
Value
Value of a call that is to be matched against low and high thresholds.
Low
Low threshold.
High
High threshold.
Enable
Shows if busyout and the
calltreatment command are enabled.
Related Commands
Command
Description
callthreshold
Enables a resource and defines associated parameters.
callthresholdpoll-interval
Enables a polling interval threshold for CPU or memory.
clearcallthreshold
Clears enabled triggers and their associated parameters.
show call treatment
To display the call-treatment configuration and statistics for handling the calls on the basis of resource availability, use the
showcalltreatment command in privileged EXEC mode.
showcalltreatment
{ config | stats }
Syntax Description
config
Displays the call treatment configuration.
stats
Displays statistics for handling the calls on the basis of resource availability.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. This command was not supported on any other platforms in this release.
12.2(8)T
This command was implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Examples
The following is sample output from thiscommand:
Router# show call treatment config
Call Treatment Config
---------------------
Call treatment is OFF.
Call treatment action is: Reject
Call treatment disconnect cause is: no-resource
Call treatment ISDN reject cause-code is: 41
The table below describes significant fields shown in this output.
Table 16 show call treatment config Field Descriptions
Field
Description
Call treatment is:
State of call treatment, either ON or OFF.
Call treatment action is:
Action trigger assigned for call treatment.
Call treatment disconnect cause is:
Reason for disconnect.
Call treatment ISDN reject cause-code is:
Reject code number assigned.
The following is sample output from the
showcalltreatmentcommand:
Router# show call treatment stats
Call Treatment Statistics
-------------------------
Total Calls by call treatment: 0
Calls accepted by call treatment: 0
Calls rejected by call treatment: 0
Reason Num. of calls rejected
------ ----------------------
cpu-5sec: 0
cpu-avg: 0
total-mem: 0
io-mem: 0
proc-mem: 0
total-calls: 0
The table below describes significant fields shown in this output.
Table 17 show call treatment stats Field Descriptions
Field
Description
Total Calls by call treatment:
Number of calls received and treated.
Calls accepted by call treatment:
Calls that passed treatment parameters.
Calls rejected by call treatment:
Calls that failed treatment parameters.
cpu-5sec
Number of calls rejected for failing the cpu-5sec parameter.
cpu-avg
Number of calls rejected for failing the cpu-avg parameter.
total-mem
Number of calls rejected for failing the total-mem parameter.
io-mem
Number of calls rejected for failing the io-mem parameter.
proc-mem
Number of calls rejected for failing the proc-mem parameter.
total-calls
Number of calls rejected for failing the total-calls parameter.
Related Commands
Command
Description
calltreatmenton
Enables call treatment to process calls when local resources are unavailable.
calltreatmentaction
Configures the action that the router takes when local resources are unavailable.
calltreatmentcause-code
Specifies the reason for the disconnection to the caller when local resources are unavailable.
calltreatmentisdn-reject
Specifies the rejection cause-code for ISDN calls when local resources are unavailable.
clearcalltreatmentstats
Clears the call-treatment statistics.
show call-router routes
To display the routes cached in the current border element (BE), use the show call-router routes in EXEC mode.
showcall-routerroutes
[ static | dynamic | all ]
Syntax Description
static
Descriptors provisioned on the border element.
dynamic
Dynamically learned descriptors.
all
Both static and dynamic descriptors.
Command Default
All
Command Modes
EXEC (#)
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Examples
The following example is sample output from this command.
Displays active call information for a voice call in progress.
showcall-routerhistory
Displays the VoIP call-history table.
showcall-routerstatus
Displays the Annex G BE status.
showdial-peervoice
Displays configuration information for dial peers.
shownum-exp
Displays how the number expansions are configured in VoIP.
showvoiceport
Displays configuration information about a specific voice port.
show call-router status
To display the Annex G border element status, use the
showcall-routerstatuscommand in user EXEC mode.
showcall-routerstatus [neighbors]
Syntax Description
neighbors
(Optional) Displays the neighbor border element status.
Command Modes
User EXEC (#)
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T and modified to add the
neighborskeyword.
Examples
The following example displays the Annex G border element status. Note that the example shows the status for two neighbors.:
Router# show call-router status neighbors
ANNEX-G CALL ROUTER STATUS:
===========================
Border Element ID Tag : Celine
Domain Name : Celine-Domain
Border Element State : UP
Border Element Local IP : 172.18.193.31:2099
Advertise Policy : STATIC descriptors
Hopcount Value : 7
Descriptor TTL : 3180
Access Policy : Neighbors only
Current Active Calls : 0
Current Calls in Cache : 0
Cumulative Active Calls : 0
Usage Ind Messages Sent : 0
Usage Ind Cfm Rcvd : 0
IRRs Received : 0
DRQs Received : 0
Usage Ind Send Retrys : 0
NEIGHBOR INFORMATION:
=====================
Local Neighbor ID : (none)
Remote Element ID : (unknown)
Remote Domain ID : (unknown)
IP Addr : 1.2.3.4:2099
Status : DOWN
Caching : OFF
Query Interval : 30 MIN (querying disabled)
Usage Indications :
Current Active Calls : 0
Retry Period : 600 SEC
Retry Window : 3600 MIN
Service Relationship Status: ACTIVE
Inbound Service Relationship : DOWN
Service ID : (none)
TTL : 1200 SEC
Outbound Service Relationship : DOWN
Service ID : (none)
TTL : (none)
Retry interval : 120 SEC (0 until next attempt)
The table below describes significant fields shown in this output.
Table 18 show call-router status Field Descriptions
Field
Description
Border Element ID Tag
Identifier for the border element.
Border Element State
Indicates if the border element is running.
Border Element Local IP
Local IP address of the border element.
Advertise Policy
Type of descriptors that the border element advertises to its neighbors. Default is
static. Other values are
dynamic and
all.
Hopcount Value
Maximum number of border element hops through which an address resolution request can be forwarded. Default is 7.
Descriptor TTL
Time-to-live value, or the amount of time, in seconds, for which a route from a neighbor is considered valid. Range is from 1 to 2147483647. Default is 1800 (30 minutes).
Access Policy
Requires that a neighbor be explicitly configured for requests to be accepted.
Local Neighbor ID
Domain name reported in service relationships.
Service Relationship Status
Service relationship between two border elements is active.
Inbound Service Relationship
Inbound time-to-Live (TTL) value in number of seconds. Range is from 1 to 4294967295.
Outbound Service Relationship
Specifies the amount of time, in seconds, to establish the outbound relationship. Range is from 1 to 65535.
Retry interval
Retry value between delivery attempts, in number of seconds. Range is from 1 to 3600.
Related Commands
Command
Description
advertise
Controls the type of descriptors that the border element advertises to its neighbors.
call-router
Enables the Annex G border element configuration commands.
hopcount
Specifies the maximum number of border element hops through which an address resolution request can be forwarded.
local
Defines the local domain, including the IP address and port border elements that the border element should use for interacting with remote border elements.
shutdown
Shuts down the Annex G border element.
ttl
Sets the expiration timer for advertisements.
show ccm-manager
To display a list of Cisco CallManager servers and their current status and availability, use the
showccm-manager command in privileged EXEC mode.
(Optional) Displays information about the backhaul link.
config-download
(Optional) Displays information about the status of Media Gateway Control Protocol (MGCP) and Skinny Client Control Protocol (SCCP) configuration download.
fallback-mgcp
(Optional) Displays the status of the MGCP gateway fallback feature.
hosts
(Optional) Displays a list of each configured Cisco CallManager server in the network, together with its operational status and host IP address.
music-on-hold
(Optional) Displays information about all the multicast music-on-hold (MOH) sessions in the gateway at any given point in time.
redundancy
(Optional) Displays failover mode and status information for hosts, including the redundant link port, failover interval, keepalive interval, MGCP traffic time, switchover time, and switchback mode.
download-tonesc1|
c2
(Optional) Displays custom tones downloaded to the gateway. The custom tone value of c1 or c2 specifies which tone information to display.
Command Default
If none of the optional keywords is specified, information related to all keywords is displayed.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco CallManager Version 3.0 and Cisco VG200.
12.2(2)XA
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.2(2)XN
This command was modified to provide enhanced MGCP voice gateway interoperability to Cisco CallManager Version 3.1 for the Cisco 2600 series, Cisco 3600 series, and Cisco VG200.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11) and the Cisco CallManager Version 3.2. It was implemented on the Cisco IAD2420 series.
12.2(15)ZJ
The download-tones [ c1 | c2 ] keywords were added for the following platforms: Cisco 2610XM, Cisco 2611XM, Cisco 2620XM, Cisco 2621XM, Cisco 2650XM, Cisco 2651XM, Cisco 2691, Cisco 3640A, Cisco 3660, Cisco 3725, and Cisco 3745.
12.3(4)T
The keywords were integrated into Cisco IOS Release 12.3(4)T.
12.3(14)T
New output was added relating to SCCP autoconfiguration.
12.4(15)XY
The display output was modified to include the number of TFTP download failures allowed.
Usage Guidelines
Use the
showccm-managerconfig-download command to determine the status of Cisco Unified Communications Manager servers and the automatic download information and statistics.
Examples
The following sample output shows the configured amplitudes, frequencies, and cadences of custom tone 1, Hong Kong:
The three tables below and give descriptions of significant fields once the tones are automatically downloaded to the gateway.
Table 19 show ccm-manager download-tones Significant Output Fields
Field
Description
Percent make
Pulse ratio in percentage of make.
DTMF low Amp.
Low frequency level.
high Amp.
High frequency level.
Pcm
Pulse Code Modulation (mu-law or a-law).
Table 20 show ccm-manager download-tones Output Fields for Dual Tones
Field of Dual Tone
Description
DR
Direction to PSTN (0) or Packet Network (1).
NF
Number of Frequency (from 1 to 4).
FOF
Frequency of First component (in Hz).
FXS AOF
Amplitude of First component (from 1 to 65535 = +3 dBm0) for the foreign exchange station (FXS).
FXO AOF
Amplitude of First component (from 1 to 65535 = +3 dBm0) for the foreign exchange office (FXO).
E&M AOF
Amplitude of First component (from 1 to 65535 = +3 dBm0) for the recEive and transMit (E&M).
FXS AOS
Amplitude of Second component (from 1 to 65535 = +3 dBm0) for the FXS.
FXO AOS
Amplitude of Second component (from 1 to 65535 = +3 dBm0) for the FXO.
E&M AOS
Amplitude of Second component (from 1 to 65535 = +3 dBm0) for the E&M.
ONTF
On time; time the tone is generated (milliseconds) for the first frequency.
OFTF
Off time; silence time (milliseconds) for the first frequency.
ONTS
On time; time the tone is generated (milliseconds) for the second frequency.
OFTS
Off time; silence time (milliseconds) for the second frequency.
ONTT
On time; time the tone is generated (milliseconds) for the third frequency.
OFTT
Off time; silence time (milliseconds) for the third frequency.
ONT4
On time; time the tone is generated (milliseconds) for the fourth frequency.
OFT4
Off time; silence time (milliseconds) for the fourth frequency.
FOF2
Frequency of First component for the second cadence.
FOS2
Frequency of Second component for the second cadence.
FOF3
Frequency of First component for the third cadence.
FOS3
Frequency of Second component for the third cadence.
FOF4
Frequency of First component for the fourth cadence.
FOS4
Frequency of Second component for the fourth cadence.
FOT
Frequency of Third component (in Hertz).
FO4
Frequency of Fourth component (in Hertz).
AOT
Amplitude of Third component (from 1 to 65535 = +3 dBm0).
AO4
Amplitude of Fourth component (from 1 to 65535 = +3 dBm0).
RCT1
Number of repeat for the first cadence.
RCT2
Number of repeat for the second cadence.
RCT3
Number of repeat for the third cadence.
RCT4
Number of repeat for the fourth cadence.
Table 21 show ccm-manager download-tones Output Fields for Sequence Tones
Field of Sequence Tone
Description
DR
Direction to PSTN (0) or Packet Network (1).
NF
Number of Frequency (from 1 to 4).
F1C1
Frequency 1 of Cadence 1.
F2C1
Frequency 2 of Cadence 1.
AOF
Amplitude of First component (from 1 to 65535).
AOS
Amplitude of Second component (from 1 to 65535).
C1ONT
Cadence 1 On Time.
C1OFT
Cadence 1 Off Time.
C2ONT
Cadence 2 On Time.
C2OFT
Cadence 2 Off Time.
C3ONT
Cadence 3 On Time.
C3OFT
Cadence 3 Off Time.
C4ONT
Cadence 4 On Time.
C4OFT
Cadence 4 Off Time.
F1C2
Frequency 1 of Cadence 2.
F2C2
Frequency 2 of Cadence 2.
F1C3
Frequency 1 of Cadence 3.
F2C3
Frequency 2 of Cadence 3.
F1C4
Frequency 1 of Cadence 4.
F2C4
Frequency 2 of Cadence 4.
The following is sample output from the
showccm-manager command for displaying the status and availability of both the primary and the backup Cisco Unified Communications Manager server:
Router# show ccm-manager
MGCP Domain Name: Router2821.cisco.com
Priority Status Host
============================================================
Primary Registered 10.78.236.222
First Backup None
Second Backup None
Current active Call Manager: 10.78.236.222
Backhaul/Redundant link port: 2428
Failover Interval: 30 seconds
Keepalive Interval: 15 seconds
Last keepalive sent: 21:48:37 UTC Nov 4 2007 (elapsed time: 00:00:15)
Last MGCP traffic time: 21:48:51 UTC Nov 4 2007 (elapsed time: 00:00:02)
Last failover time: None
Last switchback time: None
Switchback mode: Graceful
MGCP Fallback mode: Not Selected
Last MGCP Fallback start time: None
Last MGCP Fallback end time: None
MGCP Download Tones: Disabled
TFTP retry count to shut Ports: 3
PRI Backhaul Link info:
Link Protocol: TCP
Remote Port Number: 2428
Remote IP Address: 172.20.71.38
Current Link State: OPEN
Statistics:
Packets recvd: 1
Recv failures: 0
Packets xmitted: 3
Xmit failures: 0
PRI Ports being backhauled:
Slot 1, port 1
MGCP Download Tones: Enabled
Configuration Auto-Download Information
=======================================
Current version-id: {1645327B-F59A-4417-8E01-7312C61216AE}
Last config-downloaded:00:00:49
Current state: Waiting for commands
Configuration Download statistics:
Download Attempted : 6
Download Successful : 6
Download Failed : 0
Configuration Attempted : 1
Configuration Successful : 1
Configuration Failed(Parsing): 0
Configuration Failed(config) : 0
Last config download command: New Registration
Configuration Error History:
FAX mode: cisco
The table below describes the significant fields shown in the display.
Table 22 show ccm-manager Field Descriptions
Field
Description
MGCP Domain Name (system)
System used in the Internet for translating domain names of network nodes into IP addresses.
Priority
Priority of the Cisco CallManager servers present in the network. Possible priorities are primary, first backup, and second backup.
Status
Current usage of the Cisco Unified Communications Manager server. Values are Registered, Idle, Backup Polling, and Undefined.
Host
Host IP address of the Cisco CallManager server.
Current active Call Manager
IP address of the active Cisco Communications Manager server. This field can be the IP address of any one of the following Cisco Communications Manager servers: Primary, First Backup, and Second Backup.
Backhaul/Redundant link port
Port that the Cisco CallManager server is to use.
Failover Interval
Maximum amount of time that can elapse without the gateway receiving messages from the currently active Cisco Call Manager before the gateway switches to the backup Cisco Call Manager.
Keepalive Interval
Interval following which, if the gateway has not received any messages from the currently active Cisco Communications Manager server within the specified amount of time, the gateway sends a keepalive message to the Cisco Communications Manager server to determine if it is operational.
Last keepalive sent
Time in hours (military format), minutes and seconds at which the last keepalive message was sent.
Last MGCP traffic time
Time in hours (military format), minutes and seconds at which the last MGCP traffic message was sent.
Switchback mode
Displays the switchback mode configuration that determines when the primary Cisco CallManager server is used if it becomes available again while a backup Cisco CallManager server is being used.
Values that can appear in this field are Graceful, Immediate,
Schedule -time, and
Uptime-delay.
MGCP Fallback mode
Displays the MGCP fallback mode configuration. If "Not Selected" displays, then fallback is not configured. If "Enabled/OFF" displays, then fallback is configured but not in effect. If "Enabled/ON" displays, then fallback is configured and in effect.
Last MGCP Fallback start time
Start time stamp in hours (military format), minutes and seconds of the last fallback.
Lasts MGCP Fallback end time
End time stamp in hours (military format), minutes and seconds of the last fallback.
MGCP Download Tones
Displays if the customized tone download is enabled.
TFTP retry count to shut Ports
Number of TFTP download failures allowed before endpoints are shutdown.
The following is sample output from theshowccm-managerconfig-downloadcommand showing the status of the SCCP download:
Router# show ccm-manager config-download
Configuration Auto-Download Information
=======================================
Current version-id:{4171F93A-D8FC-49D8-B1C4-CE33FA8095BF}
Last config-downloaded:00:00:47
Current state:Waiting for commands
Configuration Download statistics:
Download Attempted :6
Download Successful :6
Download Failed :0
Configuration Attempted :1
Configuration Successful :1
Configuration Failed(Parsing):0
Configuration Failed(config) :0
Last config download command:New Registration
SCCP auto-configuration status
===============================================================
Registered with Call Manager: No
Local interface: FastEthernet0/0 (000c.8522.6910)
Current version-id: {D3A886A2-9BC9-41F8-9DB2-0E565CF51E5A}
Current config applied at: 04:44:45 EST Jan 9 2003
Gateway downloads succeeded: 1
Gateway download attempts: 1
Last gateway download attempt: 04:44:45 EST Jan 9 2003
Last successful gateway download: 04:44:45 EST Jan 9 2003
Current TFTP server: 10.2.6.101
Gateway resets: 0
Gateway restarts: 0
Managed endpoints: 6
Endpoint downloads succeeded: 6
Endpoint download attempts: 6
Last endpoint download attempt: 04:44:45 EST Jan 9 2003
Last successful endpoint download: 04:44:45 EST Jan 9 2003
Endpoint resets: 0
Endpoint restarts: 0
Configuration Error History:
sccp ccm CCM-PUB7 identifier 1
end
controller T1 2/0no shut
controller T1 2/0no shut
controller T1 2/0no shut
isdn switch-type primary-ni
end
The table below describes the significant fields shown in the display.
Table 23 show ccm-manager config-download Field Descriptions
Field
Description
Current state
Current configuration state.
Download Attempted
Number of times the gateway has tried to download the configuration file. The number of successes and failures is displayed.
Configuration Attempted
Number of times the gateway has tried to configure the gateway based on the configuration file. The number of successes and failures is displayed.
Managed endpoints
Number of SSCP-controlled endpoints (analog and BRI phones).
Endpoint downloads succeeded
Number of times the gateway has successfully downloaded the configuration files for SCCP-controlled endpoints.
Endpoint download attempts
Number of times the gateway has tried to download the configuration files for SCCP-controlled endpoints.
Endpoint resets
Number of SCCP gateway resets.
Endpoint restarts
Number of SCCP gateway restarts.
Configuration Error History
Displays SCCP autoconfiguration errors.
The following is sample output from the show ccm-manager fallback-mgcp command:
Router# show ccm-manager fallback-mgcp
Current active Call Manager: 172.20.71.38
MGCP Fallback mode: Enabled/OFF
Last MGCP Fallback start time: 00:14:35
Last MGCP Fallback end time: 00:17:25
The table below displays te mode. Modes are as follows:
Table 24 show ccm-manager fallback-mgcp modes
Field
Description
MGCP Fallback mode
The following are displayed:
Not Selected--Fallback is not configured.
Enabled/OFF--Fallback is configured but not in effect.
Enabled/ON--Fallback is configured and in effect.
Last MGCP Fallback start time
Start time stamp in hh:mm:ss of the last fallback.
Last MGCP Fallback end time
End time stamp in hh:mm:ss of the last fallback.
The following is sample output from the show ccm-manager music-on-hold command:
Router# show ccm-manager music-on-hold
Current active multicast sessions :1
Multicast RTP port Packets Call Codec Incoming
Address number in/out id Interface
===================================================================
172.20.71.38 2428 5/5 99 g711
The table below describes the significant fields shown in the display.
Table 25 show ccm-manager music-on-hold Field Descriptions
Field
Description
Current active multicast sessions
Number of active calls on hold.
Multicast Address
Valid class D address from which the gateway is getting the RTP streams.
RTP port number
Valid RTP port number on which the gateway receives the RTP packets.
Packets in/out
Number of RTP packets received and sent to the digital signal processor (DSP).
Call id
Call ID of the call that is on hold.
Codec
Codec number.
Incoming Interface
Interface through which the gateway is receiving the RTP stream.
Related Commands
Command
Description
ccm-managerconfig
Supplies the local MGCP voice gateway with the IP address or logical name of the TFTP server from which to download XML configuration files and enable the download of the configuration.
debugccm-manager
Displays debugging information about the Cisco CallManager.
showccm-manager
Displays a list of Cisco CallManager servers, their current status, and their availability.
showccm-managerfallback-mgcp
Displays the status of the MGCP gateway fallback feature.
showisdnstatus
Displays the Cisco IOS gateway ISDN interface status.
showmgcp
Displays the MGCP configuration information.
show cdapi
To display the Call Distributor Application Programming Interface (CDAPI), use the showcdapicommand in privileged EXEC mode.
showcdapi
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco AS5300.
12.3(4)T
This command was enhanced to display V.120 call types registering with the modem.
Usage Guidelines
CDAPI is the internal application programming interface (API) that provides an interface between signaling stacks and applications.
Examples
The following is sample output from the showcdapi command. The output displays the following information:
Field descriptions should be self-explanatory. However, the following information may be of help:
Enbloc is the mode where all call-establishment information is sent in the setup message (opposite of overlap mode, where additional messages are needed to establish the call).
Cot is the Continuity Test (COT) subsystem that supports the continuity test required by the Signaling System 7 (SS7) network to conduct loopback and tone check testing on the path before a circuit is established.
Related Commands
Command
Description
debugcdapi
Displays information about the CDAPI.
show ces clock-select
To display the setting of the network clock for the specified port,
use the
showcesclock-selectcommand in privileged EXEC mode.
show cesslot/portclock-select
Syntax Description
slot
Backplane slot number.
/port
Interface port number. The slash must be entered.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 3600 series.
Examples
The following is sample output from this command for slot 1, port 0:
Router# show ces 1/0 clock-select
Priority 1 clock source:not configured
Priority 2 clock source:not configured
Priority 3 clock source:ATM1/0 UP
Priority 4 clock source:Local oscillator
Current clock source:ATM1/0, priority:3
Field descriptions should be self-explanatory.
Related Commands
Command
Description
clock-select
Establishes the sources and priorities of the requisite
clocking signals for the OC-3/STM-1 ATM Circuit Emulation Service network
module.
show connect
To display configuration information about drop-and-insert
connections that have been configured on a router, use the
showconnectcommand in privileged EXEC mode.
show connect
{ all | elements | name | id | port
{ T1 | E1 } slot/port }
Syntax Description
all
Information for all configured connections.
elements
Information for registered hardware or software
interworking elements.
name
Information for a connection that has been named by using
the
connect global configuration
command. The name you enter is case sensitive and must match the configured
name exactly.
id
Information for a connection that you specify by an
identification number or range of identification numbers. The router assigns
these IDs automatically in the order in which they were created, beginning with
1. The
showconnectallcommand displays these IDs.
port
Information for a connection that you specify by indicating
the type of controller (T1 or E1) and location of the interface.
T1
T1 controller.
E1
E1 controller.
slot/port
Location of the T1 or E1 controller port whose connection
status you want to see. Valid values for
slot andport are
0and
1. The slash must be entered.
Command Default
No default behavior or values
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.0(5)XK
This command was introduced on the Cisco 2600 series and
Cisco 3600 series.
12.0(7)T
This command was integrated into Cisco IOS Release
12.0(7)T.
Usage Guidelines
This command shows drop-and-insert connections on modular access
routers that support drop-and-insert. It displays different information in
different formats, depending on the keyword that you use.
Examples
The following examples show how the same tabular information appears
when you enter different keywords:
Router# show connect all
ID Name Segment 1 Segment 2 State
========================================================================
1 Test -T1 1/0 01 -T1 1/1 02 ADMIN UP
2 Test2 -T1 1/0 03 -T1 1/1 04 ADMIN UP
Router# show connect id 1-2
ID Name Segment 1 Segment 2 State
========================================================================
1 Test -T1 1/0 01 -T1 1/1 02 ADMIN UP
2 Test2 -T1 1/0 03 -T1 1/1 04 ADMIN UP
Router# show connect port t1 1/1
ID Name Segment 1 Segment 2 State
========================================================================
1 Test -T1 1/0 01 -T1 1/1 02 ADMIN UP
2 Test2 -T1 1/0 03 -T1 1/1 04 ADMIN UP
The following examples show details about specific connections,
including the number of time slots in use and the switching elements:
Router# show connect id 2
Connection: 2 - Test2
Current State: ADMIN UP
Segment 1: -T1 1/0 03
TDM timeslots in use: 14-18 (5 total)
Segment 2: -T1 1/1 04
TDM timeslots in use: 14-18
Internal Switching Elements: VIC TDM Switch
Router# show connect name Test
Connection: 1 - Test
Current State: ADMIN UP
Segment 1: -T1 1/0 01
TDM timeslots in use: 1-13 (13 total)
Segment 2: -T1 1/1 02
TDM timeslots in use: 1-13
Internal Switching Elements: VIC TDM Switch
Field descriptions should be self-explanatory.
Related Commands
Command
Description
connect
Defines connections between T1 or E1 controller ports for
Drop and Insert.
tdm-group
Configures a list of time slots for creating clear channel
groups (pass-through) for TDM cross-connect.
show controllers rs366
To display information about the RS-366 video interface on the video dialing module (VDM), use the
showcontrollersrs366command in privileged EXEC mode.
showcontrollersrs366slotport
Syntax Description
slot
Slot location of the VDM module. Valid entries are 1 or 2.
port
Port location of the EIA/TIA-366 interface in the VDM module.
Command Default
No default behavior or values
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.0(5)XK
This command was introduced on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
The following example displays information about the RS-366 controller:
Router# show controllers rs366 0 1
RS366:driver is initialized in slot 1, port 0:
STATUS STATE LSR LCR ICSR EXT T1 T2 T3 T4 T5
0x02 0x01 0x00 0x50 0xE0 0x00 5000 5000 5000 20000 10000
Dial string:
121C
The table below describes significant fields shown in this output.
Table 26 show controllers rs366 Field Descriptions
Field
Description
STATUS
Last interrupt status.
STATE
Current state of the state machine.
LSR
Line status register of the VDM.
LCR
Line control register of the VDM.
ICSR
Interrupt control and status register of the VDM.
EXT
Extended register of the VDM.
T1 through T5
Timeouts 1 through 5 of the watchdog timer, in milliseconds.
Dial string
Most recently dialed number collected by the driver. 0xC at the end of the string indicates the EON (end of number) character.
show controllers timeslots
To display the channel-associated signaling (CAS) and ISDN PRI state
in detail, use the show controllers timeslots command in privileged EXEC mode.
show controllers t1/e1controller-numbertimeslotstimeslot-range
Syntax Description
tl/e1controller-number
Controller number of CAS or ISDN PRI time slot. Range is
from 0 to 7.
timeslotstimeslot-range
Timeslot. E1 range is from 1 to 31. T1 range is from 1 to
24.
Command Default
No default behavior or values
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
10.0
This command was introduced.
12.1(3)T
The
timeslots keyword was added.
12.1(5)T
This command was implemented on the Cisco AS5400.
Usage Guidelines
Use this command to display the CAS and ISDN PRI channel state in
detail. The command shows whether the DS0 channels of a controller are in idle,
in-service, maintenance, or busyout states. Use the
showcontrollerse1 or
showcontrollerst1 command to display statistics about the E1 or
T1 links.
Examples
The following example shows that the CAS state is enabled on the
Cisco AS5300 with a T1 PRI card:
Router# show controllers timeslots
T1 1 is up:
Loopback: NONE
DS0 Type Modem <-> Service Channel Rx Tx
State State A B C D A B C D
-----------------------------------------------------------------------------------------
1 cas-modem 1 in insvc connected 1 1 1 1 1 1 1 1
2 cas - - insvc idle 0 0 0 0 0 0 0 0
3 cas - - insvc idle 0 0 0 0 0 0 0 0
4 cas - - insvc idle 0 0 0 0 0 0 0 0
5 cas - - insvc idle 0 0 0 0 0 0 0 0
6 cas - - insvc idle 0 0 0 0 0 0 0 0
7 cas - - insvc idle 0 0 0 0 0 0 0 0
8 cas - - insvc idle 0 0 0 0 0 0 0 0
9 cas - - insvc idle 0 0 0 0 0 0 0 0
10 cas - - maint static-bo 0 0 0 0 1 1 1 1
11 cas - - maint static-bo 0 0 0 0 1 1 1 1
12 cas - - maint static-bo 0 0 0 0 1 1 1 1
13 cas - - maint static-bo 0 0 0 0 1 1 1 1
14 cas - - maint static-bo 0 0 0 0 1 1 1 1
15 cas - - maint static-bo 0 0 0 0 1 1 1 1
16 cas - - maint static-bo 0 0 0 0 1 1 1 1
17 cas - - maint static-bo 0 0 0 0 1 1 1 1
18 cas - - maint static-bo 0 0 0 0 1 1 1 1
19 cas - - maint dynamic-bo 0 0 0 0 1 1 1 1
20 cas - - maint dynamic-bo 0 0 0 0 1 1 1 1
21 cas - - maint dynamic-bo 0 0 0 0 1 1 1 1
22 unused
23 unused
24 unused
The following example shows that the ISDN PRI state is enabled on the
Cisco AS5300 with a T1 PRI card:
T1 2 is up:
Loopback: NONE
DS0 Type Modem <-> Service Channel Rx Tx
State State A B C D A B C D
---------------------------------------------------------------------------
1 pri - - insvc idle
2 pri - - insvc idle
3 pri - - insvc idle
4 pri - - insvc idle
5 pri - - insvc idle
6 pri - - insvc idle
7 pri - - insvc idle
8 pri - - insvc idle
9 pri - - insvc idle
10 pri - - insvc idle
11 pri - - insvc idle
12 pri - - insvc idle
13 pri - - insvc idle
14 pri - - insvc idle
15 pri - - insvc idle
16 pri - - insvc idle
17 pri - - insvc idle
18 pri - - insvc idle
19 pri - - insvc idle
20 pri - - insvc idle
21 pri-modem 2 in insvc busy
22 pri-modem 1 out insvc busy
23 pri-digi - in insvc busy
24 pri-sig - - outofsvc reserved
Field descriptions should be self-explanatory.
Related Commands
Command
Description
showcontrollerse1
Displays information about E1 links.
showcontrollerst1
Displays information about T1 links.
show controllers voice
To display information about voice-related hardware, use the showcontrollersvoice command inprivileged EXEC mode.
showcontrollersvoice
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.0(5)XQ
This command was introduced on the Cisco 1750.
Usage Guidelines
This command displays interface status information that is specific to voice-related hardware, such as the registers of the TDM switch, the host port interface of the digital signal processor (DSP), and the DSP firmware versions. The information displayed is generally useful only for diagnostic tasks performed by technical support.
Field descriptions are hardware-dependent and are meant for use by trained technical support.
Related Commands
Command
Description
showdial-peervoice
Displays configuration information and call statistics for dial peers.
showinterfacedspfarm
Displays hardware information including DRAM, SRAM, and the revision-level information on the line card.
showvoicedsp
Displays the current status of all DSP voice channels.
showvoiceport
Displays configuration information about a specific voice port.
show crm
To display the carrier call capacities statistics, use the
showcrm command in privileged EXEC mode.
showcrm
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.2(11)T
This command was introduced.
Usage Guidelines
Both the
showtrunkgroupcommandand the
showcrm command display values for the maximum number of calls. These values originate from different configuration procedures:
In the
showtrunkgroup command, the Max Calls value originates from the
max-calls command in the trunk group configuration.
In the
showcrm command, Max calls indicates the maximum number of available channels after the carrier ID or trunk group label is assigned to an interface using the
trunk-group (interface) command.
Examples
The following example illustrates the carrier call capacities statistics:
Router# show crm
Carrier:1411
Max calls:4
Max Voice (in) : 4 Cur Voice (in) : 0
Max Voice (out): 4 Cur Voice (out): 0
Max Data (in) : 4 Cur Data (in) : 0
Max Data (out) : 4 Cur Data (out) : 0
Trunk Group Label: 100
Max calls:6
Max Voice (in) : 6 Cur Voice (in) : 0
Max Voice (out): 6 Cur Voice (out): 0
Max Data (in) : 6 Cur Data (in) : 0
Max Data (out) : 6 Cur Data (out) : 0
The table below describes the fields shown in this output, in alphabetical order.
Table 27 show crm Field Descriptions
Field
Description
Carrier
ID of the carrier that handles the calls.
Cur Data (in)
Current number of incoming data calls that are handled by the carrier or trunk group.
Cur Data (out)
Current number of outgoing data calls that are handled by the carrier or trunk group.
Cur Voice (in)
Current number of incoming voice calls that are handled by the carrier or trunk group.
Cur Voice (out)
Current number of outgoing voice calls that are handled by the carrier or trunk group.
Max Calls
Maximum number of calls that are handled by the carrier or trunk group.
Max Data (in)
Maximum number of incoming data calls that are handled by the carrier or trunk group.
Max Data (out)
Maximum number of outgoing data calls that are handled by the carrier or trunk group.
Max Voice (in)
Maximum number of incoming voice calls that are handled by the carrier or trunk group.
Max Voice (out)
Maximum number of outgoing voice calls that are handled by the carrier or trunk group.
Trunk Group Label
Label of the trunk group that handles the calls.
Related Commands
Command
Description
carrier-id (dial-peer)
Specifies the carrier associated with VoIP calls.
max-calls
Specifies the maximum number of calls handled by a trunk group.
show trunk group
Displays the configuration parameters for one or more trunk groups.
trunk-group (interface)
Assigns an interface to a trunk group.
trunk-group-label (dial-peer)
Specifies the trunk group associated with VoIP calls.
show csm
To display the call switching module (CSM) statistics for a particular digital signal processor (DSP) channel, all DSP channels, or a specific modem or DSP channel, use the
showcsmcommand in privileged EXEC mode.
Displays the incoming and outgoing call switching rate.
table
(Optional) Displays the incoming and outgoing call switching rate in the form of numerical table.
call-resource
Displays statistics about the CSM call resource.
modem
Displays CSM call statistics for modems.
slot/port
(Optional) Location (and thereby identity) of a specific modem.
group
(Optional) Displays modem group information.
modem-group-number
(Optional) Location of a particular dial peer. Range: 1 to 32767.
signaling-channel
Displays CSM signaling channel Information.
voice
Displays CSM call statistics for DSP channels.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
11.3 NA
This command was introduced.
12.0(3)T
This command was modified. Port-specific values for the Cisco AS5300 were added.
12.0(7)T
This command was modified. Port-specific values for the Cisco AS5800 were added.
Usage Guidelines
This command shows the information related to CSM, which includes the DSP channel, the start time of the call, the end time of the call, and the channel on the controller used by the call.
Use the
showcsmmodem command to display the CSM call statistics for a specific modem, for a group of modems, or for all modems. If a
slot/port argument is specified, then CSM call statistics are displayed for the specified modem. If the
modem-group-numberargument is specified, the CSM call statistics for all of the modems associated with that modem group are displayed. If no keyword is specified, CSM call statistics for all modems on the Cisco AS5300 universal access server are displayed.
Use the
showcsmvoice command to display CSM statistics for a particular DSP channel. If the
slot/dspm/dsp/dsp-channel or
shelf/slot/portargumentis specified, the CSM call statistics for calls using the identified DSP channel are displayed. If no argument is specified, all CSM call statistics for all DSP channels are displayed.
Examples
The following is sample output from the
showcsm command for the Cisco AS5300 universal access server:
Router# show csm voice 2/4/4/0
slot 2, dspm 4, dsp 4, dsp channel 0,
slot 2, port 56, tone, device_status(0x0002): VDEV_STATUS_ACTIVE_CALL.
csm_state(0x0406)=CSM_OC6_CONNECTED, csm_event_proc=0x600E2678, current call thru PRI line
invalid_event_count=0, wdt_timeout_count=0
wdt_timestamp_started is not activated
wait_for_dialing:False, wait_for_bchan:False
pri_chnl=TDM_PRI_STREAM(s0, u0, c22), tdm_chnl=TDM_DSP_STREAM(s2, c27)
dchan_idb_start_index=0, dchan_idb_index=0, call_id=0xA003, bchan_num=22
csm_event=CSM_EVENT_ISDN_CONNECTED, cause=0x0000
ring_no_answer=0, ic_failure=0, ic_complete=0
dial_failure=0, oc_failure=0, oc_complete=3
oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0
remote_link_disc=0, stat_busyout=0
oobp_failure=0
call_duration_started=00:06:53, call_duration_ended=00:00:00, total_call_duration=00:00:44
The calling party phone number = 408
The called party phone number = 5271086
total_free_rbs_timeslot = 0, total_busy_rbs_timeslot = 0, total_dynamic_busy_rbs_timeslot = 0, total_static_busy_rbs_timeslot = 0,
total_sw56_rbs_timeslot = 0, total_sw56_rbs_static_bo_ts = 0,
total_free_isdn_channels = 21, total_busy_isdn_channels = 0,total_auto_busy_isdn_channels = 0,
min_free_device_threshold = 0
The table below describes the significant fields shown in the display.
Table 28 show csm voice Field Descriptions
Field
Description
slot
Slot where the VFC resides.
dsp
DSP through which this call is established.
slot/port
Logical port number for the device. This is equivalent to the DSP channel number. The port number is derived as follows:
(max_number_of_dsp_channels per dspm=12) * the dspm # (0-based) +
(max_number_of_dsp_channels per dsp=2) * the dsp # (0-based) + the dsp channel number (0-based).
tone
Which signaling tone is being used (DTMF, MF, R2). This only applies to CAS calls. Possible values are as follows:
mf
dtmf
r2-compelled
r2-semi-compelled
r2-non-compelled
device_status
Status of the device. Possible values are as follows:
VDEV_STATUS_UNLOCKED--Device is unlocked (meaning that it is available for new calls).
VDEV_STATUS_ACTIVE_WDT--Device is allocated for a call and the watchdog timer is set to time the connection response from the central office.
VDEV_STATUS_ACTIVE_CALL--Device is engaged in an active, connected call.
VDEV_STATUS_BUSYOUT_REQ--Device is requested to busyout; does not apply to voice devices.
VDEV_STATUS_BAD--Device is marked as bad and not usable for processing calls.
VDEV_STATUS_BACK2BACK_TEST--Modem is performing back-to-back testing (for modem calls only).
VDEV_STATUS_RESET--Modem needs to be reset (for modem only).
VDEV_STATUS_DOWNLOAD_FILE--Modem is downloading a file (for modem only).
VDEV_STATUS_DOWNLOAD_FAIL--Modem has failed during downloading a file (for modem only).
VDEV_STATUS_SHUTDOWN--Modem is not powered up (for modem only).
VDEV_STATUS_BUSY--Modem is busy (for modem only).
VDEV_STATUS_DOWNLOAD_REQ--Modem is requesting connection (for modem only).
csm_state
CSM call state of the current call (PRI line) associated with this device. Possible values are as follows:
CSM_IDLE_STATE--Device is idle.
CSM_IC_STATE--A device has been assigned to an incoming call.
CSM_IC1_COLLECT_ADDR_INFO--A device has been selected to perform ANI/DNIS address collection for this call. ANI/DNIS address information collection is in progress. The ANI/DNIS is used to decide whether the call should be processed by a modem or a voice DSP.
CSM_IC2_RINGING--The device assigned to this incoming call has been told to get ready for the call.
CSM_IC3_WAIT_FOR_SWITCH_OVER--A new device is selected to take over this incoming call from the device collecting the ANI/DNIS address information.
CSM_IC4_WAIT_FOR_CARRIER--This call is waiting for the CONNECT message from the carrier.
CSM_IC5_CONNECTED--This incoming call is connected to the central office.
CSM_IC6_DISCONNECTING--This incoming call is waiting for a DISCONNECT message from the VTSP module to complete the disconnect process.
CSM_OC_STATE --An outgoing call is initiated.
CSM_OC1_REQUEST_DIGIT--The device is requesting the first digit for the dial-out number.
CSM_OC2_COLLECT_1ST_DIGIT--The first digit for the dial-out number has been collected.
CSM_OC3_COLLECT_ALL_DIGIT--All the digits for the dial-out number have been collected.
CSM_OC4_DIALING--This call is waiting for a dsx0 (B channel) to be available for dialing out.
CSM_OC5_WAIT_FOR_CARRIER--This (outgoing) call is waiting for the central office to connect.
CSM_OC6_CONNECTED--This (outgoing) call is connected.
CSM_OC7_BUSY_ERROR--A busy tone has been sent to the device (for VoIP call, no busy tone is sent; just a DISCONNECT INDICATION message is sent to the VTSP module), and this call is waiting for a DISCONNECT message from the VTSP module (or ONHOOK message from the modem) to complete the disconnect process.
CSM_OC8_DISCONNECTING--The central office has disconnected this (outgoing) call, and the call is waiting for a DISCONNECT message from the VTSP module to complete the disconnect process.
csm_state: invalid_event_count
Number of invalid events received by the CSM state machine.
wdt_timeout_count
Number of times the watchdog timer is activated for this call.
wdt_timestamp_started
Whether the watchdog timer is activated for this call.
wait_for_dialing
Whether this (outgoing) call is waiting for a free digit collector to become available to dial out the outgoing digits.
wait_for_bchan
Whether this (outgoing) call is waiting for a B channel to send the call out on.
pri_chnl
Which type of TDM stream is used for the PRI connection. For PRI and CAS calls, it is always TDM_PRI_STREAM.
tdm_chnl
Which type of TDM stream is used for the connection to the device used to process this call. In the case of a VoIP call, this is always set to TDM_DSP_STREAM.
dchan_idb_start_index
First index to use when searching for the next IDB of a free D channel.
dchan_idb_index
Index of the currently available IDB of a free D channel.
csm_event
Event just passed to the CSM state machine.
cause
Event cause.
ring_no_answer
Number of times a call failed because there was no response.
ic_failure
Number of failed incoming calls.
ic_complete
Number of successful incoming calls.
dial_failure
Number of times a connection failed because there was no dial tone.
oc_failure
Number of failed outgoing calls.
oc_complete
Number of successful outgoing calls.
oc_busy
Number of outgoing calls whose connection failed because there was a busy signal.
oc_no_dial_tone
Number of outgoing calls whose connection failed because there was no dial tone.
oc_dial_timeout
Number of outgoing calls whose connection failed because the timeout value was exceeded.
call_duration_started
Start of this call.
call_duration_ended
End of this call.
total_call_duration
Duration of this call.
The calling party phone number
Calling party number as given to CSM by ISDN.
The called party phone number
Called party number as given to CSM by ISDN.
total_free_rbs_time slot
Total number of free RBS (CAS) time slots available for the whole system.
total_busy_rbs_time slot
Total number of RBS (CAS) time slots that have been busied-out. This includes both dynamically and statically busied out RBS time slots.
total_dynamic_busy_rbs_time slot
Total number of RBS (CAS) time slots that have been dynamically busied out.
total_static_busy_rbs_time slot
Total number of RBS (CAS) time slots that have been statically busied out (that is, they are busied out using the CLI command).
total_free_isdn_channels
Total number of free ISDN channels.
total_busy_isdn_channels
Total number of busy ISDN channels.
total_auto_busy_isdn_channels
Total number of ISDN channels that are automatically busied out.
Related Commands
Command
Description
showcallactivevoice
Displays the contents of the active call table.
showcallhistoryvoice
Displays the contents of the call history table.
shownum-exp
Displays how number expansions are configured.
showvoiceport
Displays configuration information about a specific voice port.
show csm call
To view the call switching module (CSM) call statistics, use the showcsmcall command in privileged EXEC mode
showcsmcall
{ failed | rate | total }
Syntax Description
failed
CSM call fail/reject rate for the last 60 seconds, 60 minutes, and 72 hours.
rate
CSM call rate for the last 60 seconds, 60 minutes, and 72 hours.
total
Total number of CSM calls for the last 60 seconds, 60 minutes, and 72 hours.
Command Default
No default behavior or values.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.3(2)T
This command was introduced on the Cisco AS5850.
Usage Guidelines
Use this command to understand CSM call volume.
Examples
The following examples show the CSM call statistics for the last 60 seconds:
Router# show csm call rate 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0....5....1....1....2....2....3....3....4....4....5....5.... 0 5 0 5 0 5 0 5 0 5 CSM call switching rate per second (last 60 seconds) # = calls entering the module per second
Router# show csm call failed 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0....5....1....1....2....2....3....3....4....4....5....5.... 0 5 0 5 0 5 0 5 0 5 CSM call fail/reject rate per second (last 60 seconds) # = calls failing per second
Router# sh csm call total1344 1244 1144 1044 944 844 744 644 544 444 344 244 144 44 0....5....1....1....2....2....3....3....4....4....5....5.... 0 5 0 5 0 5 0 5 0 5 CSM total calls (last 60 seconds)
#=numberofcalls
Field descriptions should be self-explanatory.
show cube status
To display the Cisco Unified Border Element (Cisco UBE) status, the software version, the license capacity, the image version, and the platform name of the device, use the
showcubestatus command in user EXEC or privileged EXEC mode.
showcubestatus
Syntax Description
This command has no arguments or keywords.
Command Default
Cisco UBE status is not displayed.
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Release
Modification
15.2(1)T
This command was introduced.
15.1(3)S1
This command was modified.
The output was modified to have only token characters (an alphanumeric character, hyphen [-], dot [.], exclamation mark [!], percent [%], asterisk [*], underscore [_], plus sign [+], grave [`], apostrophe ['], or a tilde [~]) in server and user-agent Session Initiation Protocol (SIP) headers. The nontoken characters present in the image name is replaced by a dot[.].
Usage Guidelines
The display of Cisco UBE status-related information is supported by the implementation of the CISCO-UBE-MIB. This MIB also provides Simple Network Management Protocol (SNMP) support for the Cisco UBE status:
The Cisco UBE status display is enabled only if the
modeborder-element command is configured with call license capacity. The
showcubestatus command displays the following message if the license capacity is not configured.
Cisco Unified Border Element (CUBE) application is not enabled
Examples
The following example configures the
modeborder-element command with call license capacity and enables the display of Cisco UBE status on the Cisco 3845 router:
Device(config)# voice service voip
Device(conf-voi-serv)# mode border-element license capacity 200
After saving the configuration and reloading the device:
Device> show cube status
CUBE-Version : 8.8
SW-Version : 15.2(1)T, Platform 3845
HA-Type : none
Licensed-Capacity : 200
In Cisco IOS Release 15.1(3)S1 and later releases, the output is as follows:
Device> show cube status
CUBE-Version : 8.8
SW-Version : 15.2.1.T, Platform 3845
HA-Type : none
Licensed-Capacity : 200
The table below describes the fields shown in the display.
Table 29 show cube status Field Descriptions
Field
Description
CUBE-Version
Version of the Cisco UBE application running on the device.
SW-Version
Image version and platform name of the device running the Cisco UBE application. This matches the image version and platform name returned by the
showversion command.
HA-Type
The type of High Availability (HA) feature configured and running on the device. The following HA types are supported:
hot-standby-chassis-to-chassis: Device-to-device hot standby support.
Licensed-Capacity
Number of SIP call legs that Cisco UBE is licensed to use. The range is from 0 to 999999. This number matches the number of licenses configured using the
modeborder-elementlicensecapacity command.
Note
The number of SIP call legs that Cisco UBE can use is platform-dependent and is not affected by the specified value for the
capacity keyword in Cisco IOS Release 15.2(1)T.
Related Commands
Command
Description
modeborder-element
Enables the set of commands used in the border-element configuration on the Cisco 2900 and Cisco 3900 series platforms.
show debug condition
To display the debugging filters that have been enabled for VoiceXML applications, ATM-enabled interfaces, or Frame Relay interfaces, use the
showdebugcondition command in privileged EXEC mode.
showdebugcondition
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC (#)
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3640, Cisco 3660, Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.0(28)S
This command was integrated into Cisco IOS Release 12.0(28)S and was enhanced to include debugging for ATM-enabled and Frame Relay-enabled interfaces.
12.2(25)S
This command was integrated into Cisco IOS Release 12.2(25)S.
12.2(27)SBC
This command was integrated into Cisco IOS Release 12.2(27)SBC.
12.2(28)SB
This command was integrated into Cisco IOS Release 12.2(28)SB.
12.4(9)T
This command was enhanced to include debugging for ATM-enabled and Frame Relay-enabled interfaces.
Usage Guidelines
This command displays the debugging filter conditions that have been set for VoiceXML applications by using thedebugconditionapplicationvoice command.
Examples
The following is sample output from this command when it is used with the VoiceXML application: